From achalkov at ya.ru  Mon Aug  1 11:25:44 2016
From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Mon, 01 Aug 2016 12:25:44 +0300
Subject: [OpenSIPS-Users] name of event in the event message's body
In-Reply-To: <89fa155e-5952-e839-2c48-344ffdd40bcd@opensips.org>
References: <1003271469625183@web21h.yandex.ru>
	<89fa155e-5952-e839-2c48-344ffdd40bcd@opensips.org>
Message-ID: <122221470043544@web12h.yandex.ru>

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From bogdan at opensips.org  Mon Aug  1 12:35:38 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 1 Aug 2016 13:35:38 +0300
Subject: [OpenSIPS-Users] Provisional response handling in case of
	forking
In-Reply-To: <ae62761d873f46c9b1229971c3a04a90@COPDCEX28.cable.comcast.com>
References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com>
	<575E8B4E.2000708@opensips.org>
	<6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>
	<57986130.2030505@opensips.org>
	<4bfa207a853f49eba59637064b77e650@COPDCEX28.cable.comcast.com>
	<579B4FF6.3040603@opensips.org>
	<ae62761d873f46c9b1229971c3a04a90@COPDCEX28.cable.comcast.com>
Message-ID: <fa0836f5-3102-8475-9f98-96b2a8e2a280@opensips.org>

Hi Agalya,

The onreply route is triggered when OpenSIPS receives a reply (from 
network level) for the given transaction. This route does not reflect at 
all the fact that the reply may or may not be sent out further to the 
caller (to A).

For the kind of logging you want, you may consider using the accounting 
module (acc) to log the early media and final reponses that are sent to A.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 29.07.2016 16:27, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> In onreply_route(), it logs all the incoming response to OpenSIPS.
>
> My intention is to log and print the outgoing response from OpenSIPS.
>
> A -------->OpenSIPS------->B
>
> Here when User A calling User B, via OpenSIPS proxy, I am logging 
> response am getting from User B.
>
> I also want to log the response sent from OpenSIPS to User A.
>
> As per my understanding , to log the response I have to do in source 
> code from where the response is sent.  Due to some formatting we need 
> for the project it is easy if I handle it in config file. So am 
> calling relay2(route) and printing the logs.
>
> Now please guide me what am doing is right or not, if not let me know 
> what could be the best way to handle this.?
>
> P.S : I want to log something like this.
>
> xlog("INFO: event=Reply_from_client; reason=$rr; status=$rs 
> [from=$fu;fromtag=$ft;to=$tu;totag=$tt;callId=$ci]\n");
>
> Regards,
>
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Friday, July 29, 2016 8:46 AM
> *To:* Ramachandran, Agalya (Contractor) 
> <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list 
> <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Provisional response handling in case 
> of forking
>
> Hi Agalya,
>
> In the onreply_route(), the logic seems ok:
>     * if flag set -> drop and log
>     * for all 180, set the flag.
> This should allow only the first 180 reply to pass.
>
> But I do not understand what is the purpose of the relay2 route? the 
> onreply route should go everything you need (in terms of 180 logic).
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 27.07.2016 16:27, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi Bogdan,
>
>     Yes you are correct. Am trying to implement that logic. (keep and
>     log the first 180, drop the others)
>
>     Am giving you snippet of onreply_route and relay2 attached here
>     with this mail.
>
>     My intention is just to log the response sent by OpenSIPS. For
>     other responses logic seems good and I could see that response
>     sent out are logged.
>
>     Only for 180 resposne, am not meeting my condition.
>
>     Correct me if am wrong.
>
>     Regards,
>
>     Agalya
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Wednesday, July 27, 2016 3:22 AM
>     *To:* Ramachandran, Agalya (Contractor)
>     <Agalya_Ramachandran at comcast.com>
>     <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling
>     list <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject:* Re: [OpenSIPS-Users] Provisional response handling in
>     case of forking
>
>     Hi Agalya,
>
>     First, take care that you can set and trigger only one reply route
>     per transaction !
>
>     Now, looking at your logic in the script, it does not reflect your
>     statement : keep and log the first 180, drop the others <- is this
>     correct ?
>
>     Best regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:
>
>         Hi Bogdan,
>
>         I have a question related to the below topic.
>
>         When am receiving more than  180 response from different
>         destinations , am sending 1^st 180 to the caller and dropping
>         consecutive  180 ?s.
>
>         And this logic is working fine.
>
>         Now I have a scenario, where I need to log what are the
>         responses sent out of Opensips.
>
>         My logic for this is, am checking the flag value. If flag is
>         set, log the 1^st 180 response, reset the flag and drop the
>         remaining 180 response.
>
>         I could see FLAG_180 is set, and when I try to reset flag *is
>         not getting reset*. Is it not possible to reset the flag
>         values when relay is done?
>
>         I have added a route(relay2) in onreply_route  Below is the
>         snippet of it.
>
>         Please guide me if am doing something wrong.
>
>         route[relay2] {
>
>                 if ($rs==100)
>
>         drop();
>
>                 if ($rs==180) {
>
>                   if (isflagset(FLAG_180)){
>
>         xlog("INFO:opensips: Flag is set \n"
>         <INFO:opensips:Flagisset%5Cn>);
>
>         resetflag(FLAG_180);
>
>                   }
>
>                   else{
>
>              drop();
>
>                  }
>
>         #setflag(FLAG_180);
>
>                 }
>
>                 xlog("INFO:opensips: Sending $rs resposne out \n"
>         <INFO:opensips:Sending$rsresposneout%5Cn>);
>
>         }
>
>         Regards,
>         Agalya
>
>         *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>         *Sent:* Monday, June 13, 2016 6:31 AM
>         *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>; Ramachandran, Agalya
>         (Contractor) <Agalya_Ramachandran at comcast.com>
>         <mailto:Agalya_Ramachandran at comcast.com>
>         *Subject:* Re: [OpenSIPS-Users] Provisional response handling
>         in case of forking
>
>         Hi Agalya,
>
>         Use the onreply route (be sure to onreply_avp_mode to be set
>         to 1 - see
>         http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290)
>         in combination with flags, to record when the an 180 reply was
>         set.
>
>         Like:
>
>         if ($rs==180) {
>             if (isflagset(FLAG_180))
>                 drop();
>             setflag(FALG_180);
>         }
>
>         The onreply_avp_mode 1 will ensure that the onreply route will
>         not overlap for 2 replies .
>
>         Regards,
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
>
>             Hi team,
>
>             We are using opensips for our project and we are currently
>             using opensips as proxy.
>
>             Am forking the incoming sip call, to two destinations. It
>             Rings in both Dest A and Dest B, as a result I get two 180
>             Ringing response from A and B.
>
>             I want to filter only the first incoming 180 Ringing
>             response and send to the actual caller. Is there a way to
>             do this in opensips config file?
>
>             I have seen drop() function which drops the complete
>             provisional response. But in my case I have to forward one
>             180 Ringing to the caller.
>
>             Can it be achieved by the changes in config file? Please
>             guide me.
>
>             Regards,
>             Agalya
>
>
>
>
>
>
>             _______________________________________________
>
>             Users mailing list
>
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Mon Aug  1 13:13:09 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 1 Aug 2016 14:13:09 +0300
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
	<579B4EC1.4010809@opensips.org>
	<418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>
Message-ID: <0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>

Hi Ben,

I see your problem here. So, let's explore this:
1) for sending the call to carrier, on OpenSIPS, you do TH (with 
advertise) resulting in a Contact with the public IP of the SBC.
2) also, manually add a RR header with the private IP of OpenSIPS.
3) send call to SBC, which will add its own RR stuff.

Now, on the sequential request from Carrier, the RURI will contain the 
Contact of OpenSIPS (the pub IP of SBC), some Route hdrs due the SBC and 
the Route we added on OpenSIPS.
- whenrequestgetsto SBC, the SBC will do loose route, consume its Route 
headers, and it will use the next available Route which points to the 
priv IP of OpenSIPS(and it will not use the public IP in RURI for routing)
- requests gets to OpenSIPS, simply remove_hf() and Route headers (do 
not do any loose_route() as it is useless) and hit th_matching-> this 
will refactor the request (RURI, Contact, Route) for the leg on the 
other side -> this should be fine.

For the other direction(still sequential), you do th_matchingon OpenSIPS 
and nothing more. This will send arequest holding the Routes due the 
SBC, a Contact with the public IP and and RURI pointing probably to the 
carrier.

Shortly you do standard TH, but on outbound scenario, add a fake RR 
header to trick the SBC to route the sequential to your OpenSIPS.

Does it make sense ?


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 29.07.2016 18:05, Newlin, Ben wrote:
>
> Here is the scenario:
>
> My servers are only listening on a private IP address. There is a 
> public address on our SBC. I have a carrier that requires that the 
> Contact IP address matches the public address we provided to them. So 
> when I do TH on my server I have to also do set_advertised_address to 
> advertise the public address in the Contact header. Sequential 
> requests use the Contact as the Request URI and the SBC is doing RR so 
> all requests will come back through it. When the SBC receives a 
> sequential request it strips its Route headers and forwards to the 
> Request URI (previous Contact URI). But that URI now points back to 
> the SBC, so it cannot deliver the request. If my server could 
> Record-Route the initial request with its private address, the SBC 
> would be able to route the request back to it.
>
> Here is a trace of my scenario: http://pastebin.com/x927mFtq. I 
> created it with SIPp so some endpoints are on the same IPs but with 
> different ports. The public IP is 192.168.99.100, with port 7060 
> representing the PSTN carrier and port 5060 the SBC. The private IP is 
> 10.0.2.15, with port 5060 being the SBC again and port 6060 being my 
> server. You can see that the ACK cannot be delivered correctly to my 
> server with this configuration.
>
> The problem here stems from the fact that Topology Hiding should 
> really be done at the edge of the network in order to be most 
> effective. But my OpenSIPS server doesn?t sit on the Edge of the 
> network, it is behind an SBC. So TH is complicated by the fact that my 
> OpenSIPS has no public IP of its own and must advertise the public IP 
> of the SBC instead, but future requests must still be routable into 
> the private network. I am really using TH to hide the two ends of the 
> call from each other, not to hide my internal network topology.
>
> Ben Newlin
>
> *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date: *Friday, July 29, 2016 at 8:40 AM
> *To: *"Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling 
> list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
>
> Hi Ben,
>
> Sorry to disagree, but IMHO they do exclude one each other.
>
> Adding RR to TH should not be seen as a way of fixing some broken TH 
> scenarios (with advertise).
> So, let me try to understand what is not working for you. You do TH 
> and advertise. In this case, normally, in the Contact headers 
> generated by OpenSIPS (as a result of TH), it should be the TH 
> interface, right ? What exactly seems to be the problem ? do you have 
> a trace to show the issues ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 27.07.2016 16:05, Newlin, Ben wrote:
>
>     I understand that normally you would not need RR with TH, but the
>     two concepts are not mutually exclusive in SIP. As I said, I have
>     a need to Record-Route the call on my server as I am advertising a
>     different address than I am listening on. This means that TH will
>     populate the Contact header with the advertised address and if I
>     cannot Record-Route with the actual address then I will not
>     receive sequential requests.
>
>     Ben Newlin
>
>     *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>     <mailto:bogdan at opensips.org>
>     *Date: *Wednesday, July 27, 2016 at 3:59 AM
>     *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>     <mailto:users at lists.opensips.org>, "Newlin, Ben"
>     <Ben.Newlin at inin.com> <mailto:Ben.Newlin at inin.com>
>     *Subject: *Re: [OpenSIPS-Users] Record-Route and Dialog
>     topology_hiding()
>
>     Hi Ben,
>
>     As I mentioned in different thread, TH is not compatible with the
>     RR mechanism. If you do TH, your OpenSIPS will act as and end
>     point (from SIP perspective), so there will be no Route/RR headers
>     at all. So no need to do loose_route or so. You just do TH
>     matching for the sequential requests and nothing more.
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 22.07.2016 16:48, Newlin, Ben wrote:
>
>         Hi,
>
>         I am using the Dialog module with topology_hiding() in my
>         server and I have a need to Record-Route the call on my server
>         as I am advertising a different address than I am listening
>         on. I have found what I believe is an inconsistency in the
>         handling of Record-Route within the Dialog topology_hiding
>         functionality. The topology_hiding isn?t a true B2BUA, but it
>         does set up different parameters for the incoming UAC and
>         outgoing UAS sides of the call for the Via headers,
>         Record-Route and Route headers, and the Contact header(s).
>
>         The problem is that the record_route() and loose_route()
>         functions operate on different sides of the call. The
>         record_route() function will only add a Record-Route header to
>         the outgoing UAS side of the call. And since the
>         record_route() function cannot be called from onreply_route,
>         but is no way to add a Record-Route header to the UAC side of
>         the call.
>
>         On the other hand, the loose_route() function only operates on
>         the incoming UAC side of the call and there is no way to
>         perform loose_route() on the UAS side of the call.
>
>         So there is a situation where Record-Route headers can only be
>         added on the outgoing UAS side, but the associated Route
>         headers can only be removed on the incoming UAC side (where
>         they won?t exist since they can?t be added) and any added
>         headers on the UAS side cannot be processed properly due to
>         the lack of loose_route.
>
>         I can provide further information if this is unclear. It
>         should be easily reproducible by attempting to use
>         record_route in a topology_hiding scenario. The route is added
>         to the outbound leg, but is not removed by loose_route so the
>         message is looped back every time.
>
>         *Ben Newlin***| Sr Voice Network Engineer, PureCloud
>
>         phone & fax +1.317.957.1009 | ben.newlin at inin.com
>         <mailto:ben.newlin at inin.com>
>
>         age removed by sender.
>
>         www.inin.com <http://www.inin.com>
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Mon Aug  1 13:17:10 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 1 Aug 2016 14:17:10 +0300
Subject: [OpenSIPS-Users] ACK method keeps looping
In-Reply-To: <11d8f9a85e284162910da715f29a2326@COPDCEX28.cable.comcast.com>
References: <11d8f9a85e284162910da715f29a2326@COPDCEX28.cable.comcast.com>
Message-ID: <9cbac1f0-18dc-998b-24b9-71b2e78754b9@opensips.org>

Hi, Agalya,

Do you do Record Routing in OpenSIPS and SIPP ? This is a must.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 29.07.2016 20:28, Ramachandran, Agalya (Contractor) wrote:
>
> Hi team,
>
> We are trying to run load test on OpenSIPS for our project.
>
> We are using sipp tool to generate load test. We are facing some 
> issues when we try to run a successful call.
>
> Source SIPp ----------Opensips --------------->Dest SIPp
>
> There are two scenarios.
>
> 1.(Works) Failure code, send Invite from source SIPp and receive 480 .
>
> 2.(Fails) Answering call, Send Invite from source SIPp  and receive 
> 200 OK.
>
> In first case, after receiving 480, OpenSIPS could able to send ACK to 
> the destination SIPp
>
> Where as in second case, there are two issues
>
> i.After receiving 200 OK, OpenSIPS doesn?t send ACK to destination 
> SIPp. When it tries to send ACK out, it fails stating, 
> usd_send_failed. Or
>
> ii.Also the same ACK keeps looping somehow for huge number of times.
>
> For the 2^nd case, we are running only single call and it is failing.
>
> Am attaching the log of SIPp messages and some snippet of OpenSIPS logs.
>
> Have you come across this scenario? Why ACK is been looped so many times?
>
> Also we see ACK message is so small, but it says, usd_send is failed 
> due to message is long.
>
> Any clue here would help us  a lot.
>
> Regards,
>
> Agalya
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Mon Aug  1 13:19:42 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 1 Aug 2016 14:19:42 +0300
Subject: [OpenSIPS-Users] Get Error while send multiple value from 302
 reply
In-Reply-To: <CANDZiCRhw5Rte774_zFD_MtVDex1c9211YUVgdqFgyyNWN3p-Q@mail.gmail.com>
References: <CANDZiCRhw5Rte774_zFD_MtVDex1c9211YUVgdqFgyyNWN3p-Q@mail.gmail.com>
Message-ID: <ddf6d93a-0579-e576-8605-4ce6555265ab@opensips.org>

Hi Shahid,

This mailing list is for the OpenSIPS project, while your question is on 
Kamailio -> you are on the wrong mailing hist.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 29.07.2016 15:58, Shahid Mehmood wrote:
> I am getting  errors while returning multipule value from response of 302
> Jul 28 21:58:01 Kamailio-2 /usr/sbin/kamailio[3939]: ERROR: pv 
> [pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
> Jul 28 21:58:01 Kamailio-2 /usr/sbin/kamailio[3939]: ERROR: <core> 
> [sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
> Jul 28 21:58:03 Kamailio-2 /usr/sbin/kamailio[3940]: ERROR: pv 
> [pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
> Jul 28 21:58:03 Kamailio-2 /usr/sbin/kamailio[3940]: ERROR: <core> 
> [sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
> Jul 28 21:58:07 Kamailio-2 /usr/sbin/kamailio[3956]: ERROR: pv 
> [pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
> Jul 28 21:58:07 Kamailio-2 /usr/sbin/kamailio[3956]: ERROR: <core> 
> [sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
> Jul 28 21:58:15 Kamailio-2 /usr/sbin/kamailio[3955]: ERROR: pv 
> [pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
> Jul 28 21:58:15 Kamailio-2 /usr/sbin/kamailio[3955]: ERROR: <core> 
> [sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
> Jul 28 21:58:31 Kamailio-2 /usr/sbin/kamailio[3954]: ERROR: pv 
> [pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
> Jul 28 21:58:31 Kamailio-2 /usr/sbin/kamailio[3954]: ERROR: <core> 
> [sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
>
>
>
> *and code is *
>
>
> $var(x)= "sip:112233#923224210048 at 192.168.12.13 
> <mailto:923224210048 at 192.168.12.13>";
> $var(y)=$var(x) + ", "+ "sip:1122#923224210048 at 192.168.12.13 
> <mailto:923224210048 at 192.168.12.13>";
> $ruri=$var(y);
>  sl_send_reply("302", "$ruri");
>
>
>
> how can i remove this error?
>
> -- 
> Shahid Mehmood
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From spanda at 3clogic.com  Mon Aug  1 13:45:42 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 1 Aug 2016 17:15:42 +0530
Subject: [OpenSIPS-Users] Can I count the number of active calls in a
 particular time interval through opensips.
Message-ID: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>

Hi All ,

        We are facing some network bandwidth when there is high load in the
opensips server . And that time some of the calls getting dropped .

      Is there any module in opensips which  can give me the active call
count in a perticualr interval of time and it may get updated automatically
. So that I may set some alert when the load become maximum .

      If I can get the call count basing on a particular user agent then
also I will be helpful .

     I have gone through "*statistics module*" . I am not sure but I think
this is possible using this module . If there is any other way then let me
know .

      Thanks in advance for any type of help .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From surabhi.garg at vvdntech.in  Mon Aug  1 14:18:41 2016
From: surabhi.garg at vvdntech.in (Surabhi Garg)
Date: Mon, 1 Aug 2016 17:48:41 +0530
Subject: [OpenSIPS-Users] Can I count the number of active calls in a
 particular time interval through opensips.
In-Reply-To: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>
References: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>
Message-ID: <CAAJ=DzcMZ2+yRV3bpyd2i85g4ZYc7HW=G=FeBdgWNTzA_6i5qA@mail.gmail.com>

Hi Sasmita,

OpenSIPS dialog module provide exported statistics which could give you
'Number of Current Active Dialogs' (at the instant command is fired).


*Below is the opensipsctl command to get active dialog statistics:*
<opensips_install_dir>/opensipsctl fifo get_statistics active_dialogs

In case you require active dialogs in a particular interval of time I would
suggest that you could make either some script or some program which
calculates the data fetched using above command.

For instance; will give you required statistics
(active dialog at timestamp X) - (active dialog at timestamp Y); where X >
Y.

Hope that helps.





*With Best Regards,Surabhi GargCell : +91
<%2B91%209910100290>8130321886| Skype : surabhi.garg9*



On Mon, Aug 1, 2016 at 5:15 PM, Sasmita Panda <spanda at 3clogic.com> wrote:

> Hi All ,
>
>         We are facing some network bandwidth when there is high load in
> the opensips server . And that time some of the calls getting dropped .
>
>       Is there any module in opensips which  can give me the active call
> count in a perticualr interval of time and it may get updated automatically
> . So that I may set some alert when the load become maximum .
>
>       If I can get the call count basing on a particular user agent then
> also I will be helpful .
>
>      I have gone through "*statistics module*" . I am not sure but I
> think this is possible using this module . If there is any other way then
> let me know .
>
>       Thanks in advance for any type of help .
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From spanda at 3clogic.com  Mon Aug  1 14:24:05 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 1 Aug 2016 17:54:05 +0530
Subject: [OpenSIPS-Users] Can I count the number of active calls in a
 particular time interval through opensips.
In-Reply-To: <CAAJ=DzcMZ2+yRV3bpyd2i85g4ZYc7HW=G=FeBdgWNTzA_6i5qA@mail.gmail.com>
References: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>
	<CAAJ=DzcMZ2+yRV3bpyd2i85g4ZYc7HW=G=FeBdgWNTzA_6i5qA@mail.gmail.com>
Message-ID: <CALf-Q45XyvbM=ATg5eWki0XnP29v+t33h=rh2n6Ky95xhYGViw@mail.gmail.com>

I am aware about the command . But I wanted to do something inside opensips
script .

        My aim is to do something like this . I wanted to count the total
number of active call and ll compare that with a fixed value, if will
become greater than this then I may drop the next call by sending some
error code . Is this possible any way with dialog module ?

      By the way , thanks for your quick reply .


*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Mon, Aug 1, 2016 at 5:48 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
wrote:

> Hi Sasmita,
>
> OpenSIPS dialog module provide exported statistics which could give you
> 'Number of Current Active Dialogs' (at the instant command is fired).
>
>
> *Below is the opensipsctl command to get active dialog statistics:*
> <opensips_install_dir>/opensipsctl fifo get_statistics active_dialogs
>
> In case you require active dialogs in a particular interval of time I
> would suggest that you could make either some script or some program which
> calculates the data fetched using above command.
>
> For instance; will give you required statistics
> (active dialog at timestamp X) - (active dialog at timestamp Y); where X >
> Y.
>
> Hope that helps.
>
>
>
>
>
> *With Best Regards,Surabhi GargCell : +91
> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>
>
>
> On Mon, Aug 1, 2016 at 5:15 PM, Sasmita Panda <spanda at 3clogic.com> wrote:
>
>> Hi All ,
>>
>>         We are facing some network bandwidth when there is high load in
>> the opensips server . And that time some of the calls getting dropped .
>>
>>       Is there any module in opensips which  can give me the active call
>> count in a perticualr interval of time and it may get updated automatically
>> . So that I may set some alert when the load become maximum .
>>
>>       If I can get the call count basing on a particular user agent then
>> also I will be helpful .
>>
>>      I have gone through "*statistics module*" . I am not sure but I
>> think this is possible using this module . If there is any other way then
>> let me know .
>>
>>       Thanks in advance for any type of help .
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From surabhi.garg at vvdntech.in  Mon Aug  1 14:45:37 2016
From: surabhi.garg at vvdntech.in (Surabhi Garg)
Date: Mon, 1 Aug 2016 18:15:37 +0530
Subject: [OpenSIPS-Users] Can I count the number of active calls in a
 particular time interval through opensips.
In-Reply-To: <CALf-Q45XyvbM=ATg5eWki0XnP29v+t33h=rh2n6Ky95xhYGViw@mail.gmail.com>
References: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>
	<CAAJ=DzcMZ2+yRV3bpyd2i85g4ZYc7HW=G=FeBdgWNTzA_6i5qA@mail.gmail.com>
	<CALf-Q45XyvbM=ATg5eWki0XnP29v+t33h=rh2n6Ky95xhYGViw@mail.gmail.com>
Message-ID: <CAAJ=DzcQyCU=Jh7K9U6Fd2XOq876tvd5sz6DjjFD1pjtyNjC3A@mail.gmail.com>

Hi Sasmita,

For this you can use Exported Pseudo Variable *"$DLG_Count"* of Dialog
Module in your OpenSIPS configuration file.



*Below is the sample snippet:*






*if (!has_totag() && is_method("INVITE")                         &&
$DLG_count > <max_threashold>) {                 send_reply("503", "Service
Unavailable");                 exit;}*
In above snippet, *max_threashold* you can replace with your desired limit.






*With Best Regards,Surabhi GargVVDN Technologies Pvt LtdCell : +91
<%2B91%209910100290>8130321886| Skype : surabhi.garg9*



On Mon, Aug 1, 2016 at 5:54 PM, Sasmita Panda <spanda at 3clogic.com> wrote:

> I am aware about the command . But I wanted to do something inside
> opensips script .
>
>         My aim is to do something like this . I wanted to count the total
> number of active call and ll compare that with a fixed value, if will
> become greater than this then I may drop the next call by sending some
> error code . Is this possible any way with dialog module ?
>
>       By the way , thanks for your quick reply .
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> On Mon, Aug 1, 2016 at 5:48 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
> wrote:
>
>> Hi Sasmita,
>>
>> OpenSIPS dialog module provide exported statistics which could give you
>> 'Number of Current Active Dialogs' (at the instant command is fired).
>>
>>
>> *Below is the opensipsctl command to get active dialog statistics:*
>> <opensips_install_dir>/opensipsctl fifo get_statistics active_dialogs
>>
>> In case you require active dialogs in a particular interval of time I
>> would suggest that you could make either some script or some program which
>> calculates the data fetched using above command.
>>
>> For instance; will give you required statistics
>> (active dialog at timestamp X) - (active dialog at timestamp Y); where X
>> > Y.
>>
>> Hope that helps.
>>
>>
>>
>>
>>
>> *With Best Regards,Surabhi GargCell : +91
>> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>>
>>
>>
>> On Mon, Aug 1, 2016 at 5:15 PM, Sasmita Panda <spanda at 3clogic.com> wrote:
>>
>>> Hi All ,
>>>
>>>         We are facing some network bandwidth when there is high load in
>>> the opensips server . And that time some of the calls getting dropped .
>>>
>>>       Is there any module in opensips which  can give me the active call
>>> count in a perticualr interval of time and it may get updated automatically
>>> . So that I may set some alert when the load become maximum .
>>>
>>>       If I can get the call count basing on a particular user agent then
>>> also I will be helpful .
>>>
>>>      I have gone through "*statistics module*" . I am not sure but I
>>> think this is possible using this module . If there is any other way then
>>> let me know .
>>>
>>>       Thanks in advance for any type of help .
>>>
>>> *Thanks & Regards*
>>> *Sasmita Panda*
>>> *Network Testing and Software Engineer*
>>> *3CLogic , ph:07827611765*
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From spanda at 3clogic.com  Mon Aug  1 14:48:23 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 1 Aug 2016 18:18:23 +0530
Subject: [OpenSIPS-Users] Can I count the number of active calls in a
 particular time interval through opensips.
In-Reply-To: <CAAJ=DzcQyCU=Jh7K9U6Fd2XOq876tvd5sz6DjjFD1pjtyNjC3A@mail.gmail.com>
References: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>
	<CAAJ=DzcMZ2+yRV3bpyd2i85g4ZYc7HW=G=FeBdgWNTzA_6i5qA@mail.gmail.com>
	<CALf-Q45XyvbM=ATg5eWki0XnP29v+t33h=rh2n6Ky95xhYGViw@mail.gmail.com>
	<CAAJ=DzcQyCU=Jh7K9U6Fd2XOq876tvd5sz6DjjFD1pjtyNjC3A@mail.gmail.com>
Message-ID: <CALf-Q450NfTEQVQGDt8M0XNqbTyOevDggJf6QEngZvSu6orYWg@mail.gmail.com>

Thank you so much . This may help me .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Mon, Aug 1, 2016 at 6:15 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
wrote:

> Hi Sasmita,
>
> For this you can use Exported Pseudo Variable *"$DLG_Count"* of Dialog
> Module in your OpenSIPS configuration file.
>
>
>
> *Below is the sample snippet:*
>
>
>
>
>
>
> *if (!has_totag() && is_method("INVITE")                         &&
> $DLG_count > <max_threashold>) {                 send_reply("503", "Service
> Unavailable");                 exit;}*
> In above snippet, *max_threashold* you can replace with your desired
> limit.
>
>
>
>
>
>
> *With Best Regards,Surabhi GargVVDN Technologies Pvt LtdCell : +91
> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>
>
>
> On Mon, Aug 1, 2016 at 5:54 PM, Sasmita Panda <spanda at 3clogic.com> wrote:
>
>> I am aware about the command . But I wanted to do something inside
>> opensips script .
>>
>>         My aim is to do something like this . I wanted to count the total
>> number of active call and ll compare that with a fixed value, if will
>> become greater than this then I may drop the next call by sending some
>> error code . Is this possible any way with dialog module ?
>>
>>       By the way , thanks for your quick reply .
>>
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>> On Mon, Aug 1, 2016 at 5:48 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
>> wrote:
>>
>>> Hi Sasmita,
>>>
>>> OpenSIPS dialog module provide exported statistics which could give you
>>> 'Number of Current Active Dialogs' (at the instant command is fired).
>>>
>>>
>>> *Below is the opensipsctl command to get active dialog statistics:*
>>> <opensips_install_dir>/opensipsctl fifo get_statistics active_dialogs
>>>
>>> In case you require active dialogs in a particular interval of time I
>>> would suggest that you could make either some script or some program which
>>> calculates the data fetched using above command.
>>>
>>> For instance; will give you required statistics
>>> (active dialog at timestamp X) - (active dialog at timestamp Y); where X
>>> > Y.
>>>
>>> Hope that helps.
>>>
>>>
>>>
>>>
>>>
>>> *With Best Regards,Surabhi GargCell : +91
>>> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>>>
>>>
>>>
>>> On Mon, Aug 1, 2016 at 5:15 PM, Sasmita Panda <spanda at 3clogic.com>
>>> wrote:
>>>
>>>> Hi All ,
>>>>
>>>>         We are facing some network bandwidth when there is high load in
>>>> the opensips server . And that time some of the calls getting dropped .
>>>>
>>>>       Is there any module in opensips which  can give me the active
>>>> call count in a perticualr interval of time and it may get updated
>>>> automatically . So that I may set some alert when the load become maximum .
>>>>
>>>>       If I can get the call count basing on a particular user agent
>>>> then also I will be helpful .
>>>>
>>>>      I have gone through "*statistics module*" . I am not sure but I
>>>> think this is possible using this module . If there is any other way then
>>>> let me know .
>>>>
>>>>       Thanks in advance for any type of help .
>>>>
>>>> *Thanks & Regards*
>>>> *Sasmita Panda*
>>>> *Network Testing and Software Engineer*
>>>> *3CLogic , ph:07827611765*
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From Ben.Newlin at inin.com  Mon Aug  1 15:42:05 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Mon, 1 Aug 2016 13:42:05 +0000
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
	<579B4EC1.4010809@opensips.org>
	<418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>
	<0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>
Message-ID: <568094C4-36E6-476E-A470-5AB11A937B2A@inin.com>

Bogdan,

You are recommending using RR on my server with TH, which is exactly what I want to do. Your solution is to do it manually because the RR module can?t be used with the TH module. This is not faking or tricking just because it is done manually; it is a real RR header and the SBC would be following correct SIP routing processes.

So we agree that the SIP concepts of RR and TH are not mutually exclusive, it is only the OpenSIPS modules that are incompatible. This confirms what I saw in my testing and reported in the initial email.

I think it would be a useful enhancement if the two modules did work together and the RR module supported doing everything you suggest doing manually below, just as it does today when TH is not used. But if that is not feasible or practical within the OpenSIPS architecture I understand. In that case, I believe a note in the documentation of the incompatibility between these modules would be beneficial as well, as it would have saved me many hours attempting to make this work and may save others that time as well.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Monday, August 1, 2016 at 7:13 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

I see your problem here. So, let's explore this:
1) for sending the call to carrier, on OpenSIPS, you do TH (with advertise) resulting in a Contact with the public IP of the SBC.
2) also, manually add a RR header with the private IP of OpenSIPS.
3) send call to SBC, which will add its own RR stuff.

Now, on the sequential request from Carrier, the RURI will contain the Contact of OpenSIPS (the pub IP of SBC), some Route hdrs due the SBC and the Route we added on OpenSIPS.
- when request gets to SBC, the SBC will do loose route, consume its Route headers, and it will use the next available Route which points to the priv IP of OpenSIPS (and it will not use the public IP in RURI for routing)
- requests gets to OpenSIPS, simply remove_hf() and Route headers (do not do any loose_route() as it is useless) and hit th_matching -> this will refactor the request (RURI, Contact, Route) for the leg on the other side -> this should be fine.

For the other direction (still sequential), you do th_matching on OpenSIPS and nothing more. This will send a request holding the Routes due the SBC, a Contact with the public IP and and RURI pointing probably to the carrier.

Shortly you do standard TH, but on outbound scenario, add a fake RR header to trick the SBC to route the sequential to your OpenSIPS.

Does it make sense ?


Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 29.07.2016 18:05, Newlin, Ben wrote:
Here is the scenario:

My servers are only listening on a private IP address. There is a public address on our SBC. I have a carrier that requires that the Contact IP address matches the public address we provided to them. So when I do TH on my server I have to also do set_advertised_address to advertise the public address in the Contact header. Sequential requests use the Contact as the Request URI and the SBC is doing RR so all requests will come back through it. When the SBC receives a sequential request it strips its Route headers and forwards to the Request URI (previous Contact URI). But that URI now points back to the SBC, so it cannot deliver the request. If my server could Record-Route the initial request with its private address, the SBC would be able to route the request back to it.

Here is a trace of my scenario: http://pastebin.com/x927mFtq. I created it with SIPp so some endpoints are on the same IPs but with different ports. The public IP is 192.168.99.100, with port 7060 representing the PSTN carrier and port 5060 the SBC. The private IP is 10.0.2.15, with port 5060 being the SBC again and port 6060 being my server. You can see that the ACK cannot be delivered correctly to my server with this configuration.

The problem here stems from the fact that Topology Hiding should really be done at the edge of the network in order to be most effective. But my OpenSIPS server doesn?t sit on the Edge of the network, it is behind an SBC. So TH is complicated by the fact that my OpenSIPS has no public IP of its own and must advertise the public IP of the SBC instead, but future requests must still be routable into the private network. I am really using TH to hide the two ends of the call from each other, not to hide my internal network topology.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Friday, July 29, 2016 at 8:40 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

Sorry to disagree, but IMHO they do exclude one each other.

Adding RR to TH should not be seen as a way of fixing some broken TH scenarios (with advertise).
So, let me try to understand what is not working for you. You do TH and advertise. In this case, normally, in the Contact headers generated by OpenSIPS (as a result of TH), it should be the TH interface, right ? What exactly seems to be the problem ? do you have a trace to show the issues ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 16:05, Newlin, Ben wrote:
I understand that normally you would not need RR with TH, but the two concepts are not mutually exclusive in SIP. As I said, I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. This means that TH will populate the Contact header with the advertised address and if I cannot Record-Route with the actual address then I will not receive sequential requests.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:59 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR mechanism. If you do TH, your OpenSIPS will act as and end point (from SIP perspective), so there will be no Route/RR headers at all. So no need to do loose_route or so. You just do TH matching for the sequential requests and nothing more.

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,

I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog topology_hiding functionality. The topology_hiding isn?t a true B2BUA, but it does set up different parameters for the incoming UAC and outgoing UAS sides of the call for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on different sides of the call. The record_route() function will only add a Record-Route header to the outgoing UAS side of the call. And since the record_route() function cannot be called from onreply_route, but is no way to add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC side of the call and there is no way to perform loose_route() on the UAS side of the call.

So there is a situation where Record-Route headers can only be added on the outgoing UAS side, but the associated Route headers can only be removed on the incoming UAC side (where they won?t exist since they can?t be added) and any added headers on the UAS side cannot be processed properly due to the lack of loose_route.

I can provide further information if this is unclear. It should be easily reproducible by attempting to use record_route in a topology_hiding scenario. The route is added to the outbound leg, but is not removed by loose_route so the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.newlin at inin.com<mailto:ben.newlin at inin.com>
[ge removed by sender.]
www.inin.com<http://www.inin.com>







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From bogdan at opensips.org  Mon Aug  1 16:01:46 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 1 Aug 2016 17:01:46 +0300
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <568094C4-36E6-476E-A470-5AB11A937B2A@inin.com>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
	<579B4EC1.4010809@opensips.org>
	<418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>
	<0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>
	<568094C4-36E6-476E-A470-5AB11A937B2A@inin.com>
Message-ID: <f61521fb-3dc6-b3de-e436-3210e678bc44@opensips.org>

Hi Ben,

1) I'm not recommending, I'm giving a workaround to your problem, 
considering the limited SBC that cannot do TH or advertise the desired 
contact IP (and abusing OpenSIPS to deal with this limitationof the SBC).

2) I'm saying "fake" RR as you do it just for the SBC sake - from 
OpenSIPS perspective has 0 valueand usage. You insert the RR headerwith 
the single purpose of forcing the SBC "notto look" into the RURI and to 
provide a workaround for routing to OpenSIPS. And it is fake as you do 
not have to do the other half, the loose_route() part.

4) No, I do not agree- the RR and TH are excluding one each other IMO. 
Once again, from the logical perspective of the SIP routing, this is is 
a non-sense. I'm suggesting this forced "marriage" just to solve the SBC 
problemand nothing more.

5) what you are attempting is not in the SIP normals - you are forcing 
the boundaries just to overcome the limitations of the SBC.

6) I agree there should be nocrash, whatever crazy combination is used 
from the script, and if ok with you, I would like to push forward here.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 01.08.2016 16:42, Newlin, Ben wrote:
>
> Bogdan,
>
> You are recommending using RR on my server with TH, which is exactly 
> what I want to do. Your solution is to do it manually because the RR 
> module can?t be used with the TH module. This is not faking or 
> tricking just because it is done manually; it is a real RR header and 
> the SBC would be following correct SIP routing processes.
>
> So we agree that the SIP concepts of RR and TH are not mutually 
> exclusive, it is only the OpenSIPS modules that are incompatible. This 
> confirms what I saw in my testing and reported in the initial email.
>
> I think it would be a useful enhancement if the two modules did work 
> together and the RR module supported doing everything you suggest 
> doing manually below, just as it does today when TH is not used. But 
> if that is not feasible or practical within the OpenSIPS architecture 
> I understand. In that case, I believe a note in the documentation of 
> the incompatibility between these modules would be beneficial as well, 
> as it would have saved me many hours attempting to make this work and 
> may save others that time as well.
>
> Ben Newlin
>
> *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date: *Monday, August 1, 2016 at 7:13 AM
> *To: *"Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling 
> list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
>
> Hi Ben,
>
> I see your problem here. So, let's explore this:
> 1) for sending the call to carrier, on OpenSIPS, you do TH (with 
> advertise) resulting in a Contact with the public IP of the SBC.
> 2) also, manually add a RR header with the private IP of OpenSIPS.
> 3) send call to SBC, which will add its own RR stuff.
>
> Now, on the sequential request from Carrier, the RURI will contain the 
> Contact of OpenSIPS (the pub IP of SBC), some Route hdrs due the SBC 
> and the Route we added on OpenSIPS.
> - when request gets to SBC, the SBC will do loose route, consume its 
> Route headers, and it will use the next available Route which points 
> to the priv IP of OpenSIPS (and it will not use the public IP in RURI 
> for routing)
> - requests gets to OpenSIPS, simply remove_hf() and Route headers (do 
> not do any loose_route() as it is useless) and hit th_matching -> this 
> will refactor the request (RURI, Contact, Route) for the leg on the 
> other side -> this should be fine.
>
> For the other direction (still sequential), you do th_matching on 
> OpenSIPS and nothing more. This will send a request holding the Routes 
> due the SBC, a Contact with the public IP and and RURI pointing 
> probably to the carrier.
>
> Shortly you do standard TH, but on outbound scenario, add a fake RR 
> header to trick the SBC to route the sequential to your OpenSIPS.
>
> Does it make sense ?
>
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 29.07.2016 18:05, Newlin, Ben wrote:
>
>     Here is the scenario:
>
>     My servers are only listening on a private IP address. There is a
>     public address on our SBC. I have a carrier that requires that the
>     Contact IP address matches the public address we provided to them.
>     So when I do TH on my server I have to also do
>     set_advertised_address to advertise the public address in the
>     Contact header. Sequential requests use the Contact as the Request
>     URI and the SBC is doing RR so all requests will come back through
>     it. When the SBC receives a sequential request it strips its Route
>     headers and forwards to the Request URI (previous Contact URI).
>     But that URI now points back to the SBC, so it cannot deliver the
>     request. If my server could Record-Route the initial request with
>     its private address, the SBC would be able to route the request
>     back to it.
>
>     Here is a trace of my scenario: http://pastebin.com/x927mFtq. I
>     created it with SIPp so some endpoints are on the same IPs but
>     with different ports. The public IP is 192.168.99.100, with port
>     7060 representing the PSTN carrier and port 5060 the SBC. The
>     private IP is 10.0.2.15, with port 5060 being the SBC again and
>     port 6060 being my server. You can see that the ACK cannot be
>     delivered correctly to my server with this configuration.
>
>     The problem here stems from the fact that Topology Hiding should
>     really be done at the edge of the network in order to be most
>     effective. But my OpenSIPS server doesn?t sit on the Edge of the
>     network, it is behind an SBC. So TH is complicated by the fact
>     that my OpenSIPS has no public IP of its own and must advertise
>     the public IP of the SBC instead, but future requests must still
>     be routable into the private network. I am really using TH to hide
>     the two ends of the call from each other, not to hide my internal
>     network topology.
>
>     Ben Newlin
>
>     *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>     <mailto:bogdan at opensips.org>
>     *Date: *Friday, July 29, 2016 at 8:40 AM
>     *To: *"Newlin, Ben" <Ben.Newlin at inin.com>
>     <mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list
>     <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject: *Re: [OpenSIPS-Users] Record-Route and Dialog
>     topology_hiding()
>
>     Hi Ben,
>
>     Sorry to disagree, but IMHO they do exclude one each other.
>
>     Adding RR to TH should not be seen as a way of fixing some broken
>     TH scenarios (with advertise).
>     So, let me try to understand what is not working for you. You do
>     TH and advertise. In this case, normally, in the Contact headers
>     generated by OpenSIPS (as a result of TH), it should be the TH
>     interface, right ? What exactly seems to be the problem ? do you
>     have a trace to show the issues ?
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 27.07.2016 16:05, Newlin, Ben wrote:
>
>         I understand that normally you would not need RR with TH, but
>         the two concepts are not mutually exclusive in SIP. As I said,
>         I have a need to Record-Route the call on my server as I am
>         advertising a different address than I am listening on. This
>         means that TH will populate the Contact header with the
>         advertised address and if I cannot Record-Route with the
>         actual address then I will not receive sequential requests.
>
>         Ben Newlin
>
>         *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>         <mailto:bogdan at opensips.org>
>         *Date: *Wednesday, July 27, 2016 at 3:59 AM
>         *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>, "Newlin, Ben"
>         <Ben.Newlin at inin.com> <mailto:Ben.Newlin at inin.com>
>         *Subject: *Re: [OpenSIPS-Users] Record-Route and Dialog
>         topology_hiding()
>
>         Hi Ben,
>
>         As I mentioned in different thread, TH is not compatible with
>         the RR mechanism. If you do TH, your OpenSIPS will act as and
>         end point (from SIP perspective), so there will be no Route/RR
>         headers at all. So no need to do loose_route or so. You just
>         do TH matching for the sequential requests and nothing more.
>
>         Regards,
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 22.07.2016 16:48, Newlin, Ben wrote:
>
>             Hi,
>
>             I am using the Dialog module with topology_hiding() in my
>             server and I have a need to Record-Route the call on my
>             server as I am advertising a different address than I am
>             listening on. I have found what I believe is an
>             inconsistency in the handling of Record-Route within the
>             Dialog topology_hiding functionality. The topology_hiding
>             isn?t a true B2BUA, but it does set up different
>             parameters for the incoming UAC and outgoing UAS sides of
>             the call for the Via headers, Record-Route and Route
>             headers, and the Contact header(s).
>
>             The problem is that the record_route() and loose_route()
>             functions operate on different sides of the call. The
>             record_route() function will only add a Record-Route
>             header to the outgoing UAS side of the call. And since the
>             record_route() function cannot be called from
>             onreply_route, but is no way to add a Record-Route header
>             to the UAC side of the call.
>
>             On the other hand, the loose_route() function only
>             operates on the incoming UAC side of the call and there is
>             no way to perform loose_route() on the UAS side of the call.
>
>             So there is a situation where Record-Route headers can
>             only be added on the outgoing UAS side, but the associated
>             Route headers can only be removed on the incoming UAC side
>             (where they won?t exist since they can?t be added) and any
>             added headers on the UAS side cannot be processed properly
>             due to the lack of loose_route.
>
>             I can provide further information if this is unclear. It
>             should be easily reproducible by attempting to use
>             record_route in a topology_hiding scenario. The route is
>             added to the outbound leg, but is not removed by
>             loose_route so the message is looped back every time.
>
>             *Ben Newlin***| Sr Voice Network Engineer, PureCloud
>
>             phone & fax +1.317.957.1009 | ben.newlin at inin.com
>             <mailto:ben.newlin at inin.com>
>
>             ge removed by sender.
>
>             www.inin.com <http://www.inin.com>
>
>
>
>
>
>
>             _______________________________________________
>
>             Users mailing list
>
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From daniel.zanutti at gmail.com  Mon Aug  1 16:30:18 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Mon, 1 Aug 2016 11:30:18 -0300
Subject: [OpenSIPS-Users] Can I count the number of active calls in a
 particular time interval through opensips.
In-Reply-To: <CALf-Q450NfTEQVQGDt8M0XNqbTyOevDggJf6QEngZvSu6orYWg@mail.gmail.com>
References: <CALf-Q45N9Q-S9zjegU7oUMbFBzG9rFVzXF1H4+PVHTB+d2Offw@mail.gmail.com>
	<CAAJ=DzcMZ2+yRV3bpyd2i85g4ZYc7HW=G=FeBdgWNTzA_6i5qA@mail.gmail.com>
	<CALf-Q45XyvbM=ATg5eWki0XnP29v+t33h=rh2n6Ky95xhYGViw@mail.gmail.com>
	<CAAJ=DzcQyCU=Jh7K9U6Fd2XOq876tvd5sz6DjjFD1pjtyNjC3A@mail.gmail.com>
	<CALf-Q450NfTEQVQGDt8M0XNqbTyOevDggJf6QEngZvSu6orYWg@mail.gmail.com>
Message-ID: <CAN0jgiNfgJ+EY=V-vq+WhtJ7rE=wBOcV4uZYUBqqZcHJqyKNMg@mail.gmail.com>

I use something different. I defined a profile with value:

modparam("dialog", "profiles_with_value", "customercps");

Then use this profile in a counter with customer_id + current timestamp:

set_dlg_profile("customercps", "$avp(customer_id)$Ts");

This way I can count current calls on current second and limit CPS per
customer.

The idea is the same, you can define your period and do something like this.

Regards


On Mon, Aug 1, 2016 at 9:48 AM, Sasmita Panda <spanda at 3clogic.com> wrote:

> Thank you so much . This may help me .
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> On Mon, Aug 1, 2016 at 6:15 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
> wrote:
>
>> Hi Sasmita,
>>
>> For this you can use Exported Pseudo Variable *"$DLG_Count"* of Dialog
>> Module in your OpenSIPS configuration file.
>>
>>
>>
>> *Below is the sample snippet:*
>>
>>
>>
>>
>>
>>
>> *if (!has_totag() && is_method("INVITE")                         &&
>> $DLG_count > <max_threashold>) {                 send_reply("503", "Service
>> Unavailable");                 exit;}*
>> In above snippet, *max_threashold* you can replace with your desired
>> limit.
>>
>>
>>
>>
>>
>>
>> *With Best Regards,Surabhi GargVVDN Technologies Pvt LtdCell : +91
>> <%2B91%209910100290>8130321886 <8130321886>| Skype : surabhi.garg9*
>>
>>
>>
>> On Mon, Aug 1, 2016 at 5:54 PM, Sasmita Panda <spanda at 3clogic.com> wrote:
>>
>>> I am aware about the command . But I wanted to do something inside
>>> opensips script .
>>>
>>>         My aim is to do something like this . I wanted to count the
>>> total number of active call and ll compare that with a fixed value, if will
>>> become greater than this then I may drop the next call by sending some
>>> error code . Is this possible any way with dialog module ?
>>>
>>>       By the way , thanks for your quick reply .
>>>
>>>
>>> *Thanks & Regards*
>>> *Sasmita Panda*
>>> *Network Testing and Software Engineer*
>>> *3CLogic , ph:07827611765*
>>>
>>> On Mon, Aug 1, 2016 at 5:48 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
>>> wrote:
>>>
>>>> Hi Sasmita,
>>>>
>>>> OpenSIPS dialog module provide exported statistics which could give you
>>>> 'Number of Current Active Dialogs' (at the instant command is fired).
>>>>
>>>>
>>>> *Below is the opensipsctl command to get active dialog statistics:*
>>>> <opensips_install_dir>/opensipsctl fifo get_statistics active_dialogs
>>>>
>>>> In case you require active dialogs in a particular interval of time I
>>>> would suggest that you could make either some script or some program which
>>>> calculates the data fetched using above command.
>>>>
>>>> For instance; will give you required statistics
>>>> (active dialog at timestamp X) - (active dialog at timestamp Y); where
>>>> X > Y.
>>>>
>>>> Hope that helps.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *With Best Regards,Surabhi GargCell : +91
>>>> <%2B91%209910100290>8130321886 <8130321886>| Skype : surabhi.garg9*
>>>>
>>>>
>>>>
>>>> On Mon, Aug 1, 2016 at 5:15 PM, Sasmita Panda <spanda at 3clogic.com>
>>>> wrote:
>>>>
>>>>> Hi All ,
>>>>>
>>>>>         We are facing some network bandwidth when there is high load
>>>>> in the opensips server . And that time some of the calls getting dropped .
>>>>>
>>>>>       Is there any module in opensips which  can give me the active
>>>>> call count in a perticualr interval of time and it may get updated
>>>>> automatically . So that I may set some alert when the load become maximum .
>>>>>
>>>>>       If I can get the call count basing on a particular user agent
>>>>> then also I will be helpful .
>>>>>
>>>>>      I have gone through "*statistics module*" . I am not sure but I
>>>>> think this is possible using this module . If there is any other way then
>>>>> let me know .
>>>>>
>>>>>       Thanks in advance for any type of help .
>>>>>
>>>>> *Thanks & Regards*
>>>>> *Sasmita Panda*
>>>>> *Network Testing and Software Engineer*
>>>>> *3CLogic , ph:07827611765*
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From Ben.Newlin at inin.com  Mon Aug  1 16:34:47 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Mon, 1 Aug 2016 14:34:47 +0000
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <f61521fb-3dc6-b3de-e436-3210e678bc44@opensips.org>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
	<579B4EC1.4010809@opensips.org>
	<418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>
	<0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>
	<568094C4-36E6-476E-A470-5AB11A937B2A@inin.com>
	<f61521fb-3dc6-b3de-e436-3210e678bc44@opensips.org>
Message-ID: <B68FD830-D259-4556-9D86-5C564F8B07CB@inin.com>

Bogdan,

2) All Record-Routing is done for the sake of other servers. The Route headers are always useless to the server that generated them. Record-Routes are always inserted to force another server ?not to look? at the RURI and instead route somewhere else. That is the entire purpose of Record-Routes! I can?t see why you feel this scenario somehow violates their normal usage.

4) Again, I don?t see why this would be considered nonsense. SIP allows any server to use Record-Routing not just proxies, but even if it didn?t OpenSIPS is a proxy unless using B2BUA. Even when using TH, it is still a proxy from SIP standpoint because it is not changing the fundamental identifiers of the call (From-tag, To-tag, Call-ID).

5) Again, Record-Routing is very much in the normal SIP process, no matter where it is being done.

6) I appreciate you looking into the crashes I have reported in a separate thread, but they have nothing to do with this scenario.

All that being said, I can agree to disagree. You have confirmed that OpenSIPS doesn?t and won?t support the functionality I?m looking for. I have several options for workarounds that I am investigating.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Monday, August 1, 2016 at 10:01 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

1) I'm not recommending, I'm giving a workaround to your problem, considering the limited SBC that cannot do TH or advertise the desired contact IP (and abusing OpenSIPS to deal with this limitation of the SBC).

2) I'm saying "fake" RR as you do it just for the SBC sake - from OpenSIPS perspective has 0 value and usage. You insert the RR header with the single purpose of forcing the SBC "not to look" into the RURI and to provide a workaround for routing to OpenSIPS. And it is fake as you do not have to do the other half, the loose_route() part.

4) No, I do not agree - the RR and TH are excluding one each other IMO. Once again, from the logical perspective of the SIP routing, this is is a non-sense. I'm suggesting this forced "marriage"  just to solve the SBC problem and nothing more.

5) what you are attempting is not in the SIP normals - you are forcing the boundaries just to overcome the limitations of the SBC.

6) I agree there should be no crash, whatever crazy combination is used from the script, and if ok with you, I would like to push forward here.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 01.08.2016 16:42, Newlin, Ben wrote:
Bogdan,

You are recommending using RR on my server with TH, which is exactly what I want to do. Your solution is to do it manually because the RR module can?t be used with the TH module. This is not faking or tricking just because it is done manually; it is a real RR header and the SBC would be following correct SIP routing processes.

So we agree that the SIP concepts of RR and TH are not mutually exclusive, it is only the OpenSIPS modules that are incompatible. This confirms what I saw in my testing and reported in the initial email.

I think it would be a useful enhancement if the two modules did work together and the RR module supported doing everything you suggest doing manually below, just as it does today when TH is not used. But if that is not feasible or practical within the OpenSIPS architecture I understand. In that case, I believe a note in the documentation of the incompatibility between these modules would be beneficial as well, as it would have saved me many hours attempting to make this work and may save others that time as well.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Monday, August 1, 2016 at 7:13 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

I see your problem here. So, let's explore this:
1) for sending the call to carrier, on OpenSIPS, you do TH (with advertise) resulting in a Contact with the public IP of the SBC.
2) also, manually add a RR header with the private IP of OpenSIPS.
3) send call to SBC, which will add its own RR stuff.

Now, on the sequential request from Carrier, the RURI will contain the Contact of OpenSIPS (the pub IP of SBC), some Route hdrs due the SBC and the Route we added on OpenSIPS.
- when request gets to SBC, the SBC will do loose route, consume its Route headers, and it will use the next available Route which points to the priv IP of OpenSIPS (and it will not use the public IP in RURI for routing)
- requests gets to OpenSIPS, simply remove_hf() and Route headers (do not do any loose_route() as it is useless) and hit th_matching -> this will refactor the request (RURI, Contact, Route) for the leg on the other side -> this should be fine.

For the other direction (still sequential), you do th_matching on OpenSIPS and nothing more. This will send a request holding the Routes due the SBC, a Contact with the public IP and and RURI pointing probably to the carrier.

Shortly you do standard TH, but on outbound scenario, add a fake RR header to trick the SBC to route the sequential to your OpenSIPS.

Does it make sense ?


Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 29.07.2016 18:05, Newlin, Ben wrote:
Here is the scenario:

My servers are only listening on a private IP address. There is a public address on our SBC. I have a carrier that requires that the Contact IP address matches the public address we provided to them. So when I do TH on my server I have to also do set_advertised_address to advertise the public address in the Contact header. Sequential requests use the Contact as the Request URI and the SBC is doing RR so all requests will come back through it. When the SBC receives a sequential request it strips its Route headers and forwards to the Request URI (previous Contact URI). But that URI now points back to the SBC, so it cannot deliver the request. If my server could Record-Route the initial request with its private address, the SBC would be able to route the request back to it.

Here is a trace of my scenario: http://pastebin.com/x927mFtq. I created it with SIPp so some endpoints are on the same IPs but with different ports. The public IP is 192.168.99.100, with port 7060 representing the PSTN carrier and port 5060 the SBC. The private IP is 10.0.2.15, with port 5060 being the SBC again and port 6060 being my server. You can see that the ACK cannot be delivered correctly to my server with this configuration.

The problem here stems from the fact that Topology Hiding should really be done at the edge of the network in order to be most effective. But my OpenSIPS server doesn?t sit on the Edge of the network, it is behind an SBC. So TH is complicated by the fact that my OpenSIPS has no public IP of its own and must advertise the public IP of the SBC instead, but future requests must still be routable into the private network. I am really using TH to hide the two ends of the call from each other, not to hide my internal network topology.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Friday, July 29, 2016 at 8:40 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

Sorry to disagree, but IMHO they do exclude one each other.

Adding RR to TH should not be seen as a way of fixing some broken TH scenarios (with advertise).
So, let me try to understand what is not working for you. You do TH and advertise. In this case, normally, in the Contact headers generated by OpenSIPS (as a result of TH), it should be the TH interface, right ? What exactly seems to be the problem ? do you have a trace to show the issues ?

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 16:05, Newlin, Ben wrote:
I understand that normally you would not need RR with TH, but the two concepts are not mutually exclusive in SIP. As I said, I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. This means that TH will populate the Contact header with the advertised address and if I cannot Record-Route with the actual address then I will not receive sequential requests.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:59 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR mechanism. If you do TH, your OpenSIPS will act as and end point (from SIP perspective), so there will be no Route/RR headers at all. So no need to do loose_route or so. You just do TH matching for the sequential requests and nothing more.

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,

I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog topology_hiding functionality. The topology_hiding isn?t a true B2BUA, but it does set up different parameters for the incoming UAC and outgoing UAS sides of the call for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on different sides of the call. The record_route() function will only add a Record-Route header to the outgoing UAS side of the call. And since the record_route() function cannot be called from onreply_route, but is no way to add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC side of the call and there is no way to perform loose_route() on the UAS side of the call.

So there is a situation where Record-Route headers can only be added on the outgoing UAS side, but the associated Route headers can only be removed on the incoming UAC side (where they won?t exist since they can?t be added) and any added headers on the UAS side cannot be processed properly due to the lack of loose_route.

I can provide further information if this is unclear. It should be easily reproducible by attempting to use record_route in a topology_hiding scenario. The route is added to the outbound leg, but is not removed by loose_route so the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.newlin at inin.com<mailto:ben.newlin at inin.com>
[e removed by sender.]
www.inin.com<http://www.inin.com>








_______________________________________________

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From bogdan at opensips.org  Mon Aug  1 16:57:32 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 1 Aug 2016 17:57:32 +0300
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <273E1DB2-DD7E-412E-B762-2E1826B2AC7D@inin.com>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
	<204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
	<579B5127.7080805@opensips.org>
	<273E1DB2-DD7E-412E-B762-2E1826B2AC7D@inin.com>
Message-ID: <8fa14a30-da35-3a54-257e-390ce097975e@opensips.org>

Hi Ben,

According to the BT,the crash is in a pkg_malloc() call:
                     route = pkg_malloc(size);
Please double check this with gdb info.

Ifso, this indicate a memory corruptionand we have 2 options here:
     - you compile with memory debugger (see my previous emails)
     - provide step-by-step indications on how to reproduce this crash.

Thanks and Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 29.07.2016 15:54, Newlin, Ben wrote:
>
> This is 1.11.6, running on CentOS 7.
>
> Ben Newlin
>
> *From: *<users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei 
> Iancu <bogdan at opensips.org>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Friday, July 29, 2016 at 8:50 AM
> *To: *"Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling 
> list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
> Ben,
>
> What OpenSIPS version is this (the crashing one) ? 1.11 or 2.1 ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 27.07.2016 19:02, Newlin, Ben wrote:
>
>     I have identified that these crashes are occurring when the far
>     end system is not returning the Record-Route headers in the 200 OK
>     response. The headers are present in the 180 response, but not the
>     200 OK. I have reproduced the scenario using SIPp and captured a
>     SIP trace: http://pastebin.com/ckKk3EhY <http://pastebin.com/ckKk3EhY>
>
>     The crash occurs on receipt of the ACK request and attempt to
>     match the dialog.
>
>     I also captured a BT for this scenario as well, in case anything
>     specific in the trace made the issue easier to find:
>     http://pastebin.com/cM3FhPiw
>
>     I am working with the other system to try to fix their behavior.
>
>     Ideally the Record-Route headers from previous replies could be
>     used in this case to allow the call to succeed, but I don?t know
>     if that is possible.
>
>     Thanks,
>
>     Ben Newlin
>
>     *From: *"Newlin, Ben" <Ben.Newlin at inin.com>
>     <mailto:Ben.Newlin at inin.com>
>     *Date: *Wednesday, July 27, 2016 at 9:44 AM
>     *To: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>     <mailto:bogdan at opensips.org>, OpenSIPS users mailling list
>     <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
>     Bogdan,
>
>     This is a different scenario than the other you responded to. As I
>     said, we have two types of servers that work together. One is a
>     load-balancer and runs as a proxy. It uses double Record-Route
>     because it sends messages between public and private networks.
>     Then we have our other servers using TH which receive those
>     requests. We are not using TH and RR on the same server (although
>     I would like to).
>
>     If validate_dialog() and fix_route_dialog() (and possibly
>     loose_route()) should not be called when using TH, I believe the
>     documentation should reference that. It states that match_dialog()
>     must be used with TH, but does not indicate that the other
>     functions should not be used or that the functionality won?t work.
>     There is also no documentation of the incompatibility between RR
>     and TH.
>
>     Either way, I ran a test where I removed all calls to
>     loose_route(), validate_dialog(), and fix_route_dialog() from my
>     script. The crash still occurred and the BT still pointed to
>     fix_route_dialog() function. So it must be getting called from
>     within Dialog module somewhere. That BT is here:
>     http://pastebin.com/wu2X2Hxh
>
>     I collected this BT with loose_route() being called from my
>     script, but not validate_dialog() or fix_route_dialog():
>     http://pastebin.com/6V7yPaHF
>
>     This BT was collected with all three functions being called from
>     my script: http://pastebin.com/fZYYdndn
>
>     Ben Newlin
>
>     *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>     <mailto:bogdan at opensips.org>
>     *Date: *Wednesday, July 27, 2016 at 3:57 AM
>     *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>     <mailto:users at lists.opensips.org>, "Newlin, Ben"
>     <Ben.Newlin at inin.com> <mailto:Ben.Newlin at inin.com>
>     *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
>     Hi Ben,
>
>     First, if you use TH, makes no sense to do Record-Routing - there
>     are 2 SIP concepts that overlaps. You either act as an end-point
>     (by doing TH), either as a proxy (doing RR).
>
>     If doing TH, makes no sense to use validate + fix as these
>     functions check and repair the routing information in the request
>     (like Route and Contact headers). if you do TH, this routing info
>     is actually hidden and added by OpenSIPS, so there is nothing to
>     fix and repair.
>
>     Nevertheless, this should not crash or corrupt OpenSIPS. HAve you
>     managed to get a corefile ?
>
>     Also if you suspect memory corruption, you can compile-in the
>     memory debugger - see
>     http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .
>
>     Regards,
>
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 26.07.2016 23:20, Newlin, Ben wrote:
>
>         I have had 3 OpenSIPS server crashes in the last week. All
>         were due to segmentation faults. I was not able to capture
>         core dumps; I am configuring that now to catch the next crash.
>
>         My logs leading up to the crash are full of errors from
>         fix_route_dialog() complaining about invalid URIs for
>         sequential requests:
>
>         Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to
>         parse SIP uri
>
>         Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0
>         parsed: <ip:1> (4) /
>         <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)
>
>         Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to
>         parse SIP uri
>
>         Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0
>         parsed: <b0i2> (4) /
>         <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on>
>         (65)
>
>         Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to
>         parse SIP uri
>
>         Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0
>         parsed: <ervi> (4) /
>         <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)
>
>         Many times the ?URI? displayed in the error message is
>         actually internal OpenSIPS variables, as in the last error
>         above. When they are from the SIP message, I have verified
>         that the messages themselves are correctly formatted. This
>         leads me to believe there is memory corruption occurring.
>
>         This all started when I updated my load-balancer servers to
>         use Record-Routing, specifically the ?double_rr? mechanism for
>         when multiple interfaces exist. The Record-Routing is
>         occurring on different servers which have not crashed. Only
>         the servers receiving the Record-Routed messages are
>         experiencing the errors.
>
>         Here is a piece of the code processing sequential requests. I
>         am using the topology_hiding() functionality of the Dialog
>         module. Are validate_dialog() and fix_route_dialog() still
>         valid in a topology_hiding scenario?
>
>         if (t_check_trans())
>
>         setflag(SEQ_REQUEST);
>
>           if (has_totag())
>
>           {
>
>             loose_route();
>
>             if (match_dialog())
>
>             {
>
>               if (!validate_dialog())
>
>         fix_route_dialog();
>
>               if (is_method("BYE"))
>
>         setflag(ACC_FLAG);
>
>         setflag(SEQ_REQUEST);
>
>             }
>
>             else if (!isflagset(SEQ_REQUEST))
>
>             {
>
>               if (!is_method("ACK")) {
>
>                 route(rlog, LV_ERROR, "check_sequential", "Sequential
>         request not matched");
>
>           route(reply_error, "481", "Call Does Not Exist");
>
>               }
>
>               return(EXIT);
>
>             }
>
>           }
>
>         I will attempt to get core dumps of future crashes.
>
>         Thanks,
>
>         Ben Newlin
>
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Agalya_Ramachandran at comcast.com  Mon Aug  1 20:34:31 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Mon, 1 Aug 2016 18:34:31 +0000
Subject: [OpenSIPS-Users] ACK method keeps looping
In-Reply-To: <9cbac1f0-18dc-998b-24b9-71b2e78754b9@opensips.org>
References: <11d8f9a85e284162910da715f29a2326@COPDCEX28.cable.comcast.com>
	<9cbac1f0-18dc-998b-24b9-71b2e78754b9@opensips.org>
Message-ID: <23cff545ac5146238446b8c1706c0963@COPDCEX28.cable.comcast.com>

Hi Bogdan,

Thank you for hint. I have enabled in Opensips. After enabling in SIPP too it worked.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Monday, August 01, 2016 7:17 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Cc: Nolan, Eugene <Eugene_Nolan at cable.comcast.com>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] ACK method keeps looping

Hi, Agalya,

Do you do Record Routing in OpenSIPS and SIPP ? This is a must.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 29.07.2016 20:28, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are trying to run load test on OpenSIPS for our project.
We are using sipp tool to generate load test. We are facing some issues when we try to run a successful call.

Source SIPp ----------Opensips --------------->Dest SIPp

There are two scenarios.

1.       (Works) Failure code, send Invite from source SIPp and receive 480 .

2.       (Fails) Answering call, Send Invite from source SIPp  and receive 200 OK.
In first case, after receiving 480, OpenSIPS could able to send ACK to the destination SIPp
Where as in second case, there are two issues

                                 i.            After receiving 200 OK, OpenSIPS doesn't send ACK to destination SIPp. When it tries to send ACK out, it fails stating, usd_send_failed. Or

                               ii.            Also the same ACK keeps looping somehow for huge number of times.

For the 2nd case, we are running only single call and it is failing.
Am attaching the log of SIPp messages and some snippet of OpenSIPS logs.

Have you come across this scenario? Why ACK is been looped so many times?
Also we see ACK message is so small, but it says, usd_send is failed due to message is long.
Any clue here would help us  a lot.

Regards,
Agalya







_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Ben.Newlin at inin.com  Mon Aug  1 19:17:31 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Mon, 1 Aug 2016 17:17:31 +0000
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <8fa14a30-da35-3a54-257e-390ce097975e@opensips.org>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
	<204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
	<579B5127.7080805@opensips.org>
	<273E1DB2-DD7E-412E-B762-2E1826B2AC7D@inin.com>
	<8fa14a30-da35-3a54-257e-390ce097975e@opensips.org>
Message-ID: <6CE77C34-7F1B-459F-8BDF-2C6ABDA56BC6@inin.com>

Bogdan,

I am not familiar with gdb, so I double check what you?ve assessed. If there are some other steps with gdb you would like me to perform, just let me know what to do.

Is there a way to compile the memory debugger without using the interactive `make menuconfig` command? Our build system is completely automated, so it impossible for me to do it this way. Can I pass the options as build parameters or alter the makefile in some way?

I can provide a SIPp scenario which should reproduce the issue on any basic script that uses Dialog with topology hiding, if that would be easier.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Monday, August 1, 2016 at 10:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

According to the BT, the crash is in a pkg_malloc() call:
                    route = pkg_malloc(size);
Please double check this with gdb info.

If so, this indicate a memory corruption and we have 2 options here:
    - you compile with memory debugger (see my previous emails)
    - provide step-by-step indications on how to reproduce this crash.

Thanks and Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 29.07.2016 15:54, Newlin, Ben wrote:
This is 1.11.6, running on CentOS 7.


Ben Newlin

From: <users-bounces at lists.opensips.org><mailto:users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Date: Friday, July 29, 2016 at 8:50 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Ben,

What OpenSIPS version is this (the crashing one) ? 1.11 or 2.1 ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 19:02, Newlin, Ben wrote:
I have identified that these crashes are occurring when the far end system is not returning the Record-Route headers in the 200 OK response. The headers are present in the 180 response, but not the 200 OK. I have reproduced the scenario using SIPp and captured a SIP trace: http://pastebin.com/ckKk3EhY

The crash occurs on receipt of the ACK request and attempt to match the dialog.

I also captured a BT for this scenario as well, in case anything specific in the trace made the issue easier to find: http://pastebin.com/cM3FhPiw

I am working with the other system to try to fix their behavior.

Ideally the Record-Route headers from previous replies could be used in this case to allow the call to succeed, but I don?t know if that is possible.

Thanks,

Ben Newlin

From: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Date: Wednesday, July 27, 2016 at 9:44 AM
To: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Bogdan,

This is a different scenario than the other you responded to. As I said, we have two types of servers that work together. One is a load-balancer and runs as a proxy. It uses double Record-Route because it sends messages between public and private networks. Then we have our other servers using TH which receive those requests. We are not using TH and RR on the same server (although I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should not be called when using TH, I believe the documentation should reference that. It states that match_dialog() must be used with TH, but does not indicate that the other functions should not be used or that the functionality won?t work. There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), validate_dialog(), and fix_route_dialog() from my script. The crash still occurred and the BT still pointed to fix_route_dialog() function. So it must be getting called from within Dialog module somewhere. That BT is here: http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP concepts that overlaps. You either act as an end-point (by doing TH), either as a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and repair the routing information in the request (like Route and Contact headers). if you do TH, this routing info is actually hidden and added by OpenSIPS, so there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: <b0i2> (4) / <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)

Many times the ?URI? displayed in the error message is actually internal OpenSIPS variables, as in the last error above. When they are from the SIP message, I have verified that the messages themselves are correctly formatted. This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, specifically the ?double_rr? mechanism for when multiple interfaces exist. The Record-Routing is occurring on different servers which have not crashed. Only the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the topology_hiding() functionality of the Dialog module. Are validate_dialog() and fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
    setflag(SEQ_REQUEST);

  if (has_totag())
  {
    loose_route();

    if (match_dialog())
    {
      if (!validate_dialog())
        fix_route_dialog();

      if (is_method("BYE"))
        setflag(ACC_FLAG);

      setflag(SEQ_REQUEST);
    }
    else if (!isflagset(SEQ_REQUEST))
    {
      if (!is_method("ACK")) {
        route(rlog, LV_ERROR, "check_sequential", "Sequential request not matched");
        route(reply_error, "481", "Call Does Not Exist");
      }

      return(EXIT);
    }
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin







_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users






_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Tue Aug  2 09:47:31 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 2 Aug 2016 10:47:31 +0300
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <6CE77C34-7F1B-459F-8BDF-2C6ABDA56BC6@inin.com>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
	<204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
	<579B5127.7080805@opensips.org>
	<273E1DB2-DD7E-412E-B762-2E1826B2AC7D@inin.com>
	<8fa14a30-da35-3a54-257e-390ce097975e@opensips.org>
	<6CE77C34-7F1B-459F-8BDF-2C6ABDA56BC6@inin.com>
Message-ID: <c929d518-9bb8-63c7-25d0-260b6989d9d1@opensips.org>

Ben,

To make it easier, please send me the instructions on how to reproduce 
the crash.

Thanks and Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 01.08.2016 20:17, Newlin, Ben wrote:
>
> Bogdan,
>
> I am not familiar with gdb, so I double check what you?ve assessed. If 
> there are some other steps with gdb you would like me to perform, just 
> let me know what to do.
>
> Is there a way to compile the memory debugger without using the 
> interactive `make menuconfig` command? Our build system is completely 
> automated, so it impossible for me to do it this way. Can I pass the 
> options as build parameters or alter the makefile in some way?
>
> I can provide a SIPp scenario which should reproduce the issue on any 
> basic script that uses Dialog with topology hiding, if that would be 
> easier.
>
> Ben Newlin
>
> *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date: *Monday, August 1, 2016 at 10:57 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>, 
> "Newlin, Ben" <Ben.Newlin at inin.com>
> *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
> Hi Ben,
>
> According to the BT, the crash is in a pkg_malloc() call:
>                     route = pkg_malloc(size);
> Please double check this with gdb info.
>
> If so, this indicate a memory corruption and we have 2 options here:
>     - you compile with memory debugger (see my previous emails)
>     - provide step-by-step indications on how to reproduce this crash.
>
> Thanks and Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 29.07.2016 15:54, Newlin, Ben wrote:
>
>     This is 1.11.6, running on CentOS 7.
>
>     Ben Newlin
>
>     *From: *<users-bounces at lists.opensips.org>
>     <mailto:users-bounces at lists.opensips.org> on behalf of
>     Bogdan-Andrei Iancu <bogdan at opensips.org> <mailto:bogdan at opensips.org>
>     *Reply-To: *OpenSIPS users mailling list
>     <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Date: *Friday, July 29, 2016 at 8:50 AM
>     *To: *"Newlin, Ben" <Ben.Newlin at inin.com>
>     <mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list
>     <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
>     Ben,
>
>     What OpenSIPS version is this (the crashing one) ? 1.11 or 2.1 ?
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 27.07.2016 19:02, Newlin, Ben wrote:
>
>         I have identified that these crashes are occurring when the
>         far end system is not returning the Record-Route headers in
>         the 200 OK response. The headers are present in the 180
>         response, but not the 200 OK. I have reproduced the scenario
>         using SIPp and captured a SIP trace:
>         http://pastebin.com/ckKk3EhY <http://pastebin.com/ckKk3EhY>
>
>         The crash occurs on receipt of the ACK request and attempt to
>         match the dialog.
>
>         I also captured a BT for this scenario as well, in case
>         anything specific in the trace made the issue easier to find:
>         http://pastebin.com/cM3FhPiw
>
>         I am working with the other system to try to fix their behavior.
>
>         Ideally the Record-Route headers from previous replies could
>         be used in this case to allow the call to succeed, but I don?t
>         know if that is possible.
>
>         Thanks,
>
>         Ben Newlin
>
>         *From: *"Newlin, Ben" <Ben.Newlin at inin.com>
>         <mailto:Ben.Newlin at inin.com>
>         *Date: *Wednesday, July 27, 2016 at 9:44 AM
>         *To: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>         <mailto:bogdan at opensips.org>, OpenSIPS users mailling list
>         <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>         *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
>         Bogdan,
>
>         This is a different scenario than the other you responded to.
>         As I said, we have two types of servers that work together.
>         One is a load-balancer and runs as a proxy. It uses double
>         Record-Route because it sends messages between public and
>         private networks. Then we have our other servers using TH
>         which receive those requests. We are not using TH and RR on
>         the same server (although I would like to).
>
>         If validate_dialog() and fix_route_dialog() (and possibly
>         loose_route()) should not be called when using TH, I believe
>         the documentation should reference that. It states that
>         match_dialog() must be used with TH, but does not indicate
>         that the other functions should not be used or that the
>         functionality won?t work. There is also no documentation of
>         the incompatibility between RR and TH.
>
>         Either way, I ran a test where I removed all calls to
>         loose_route(), validate_dialog(), and fix_route_dialog() from
>         my script. The crash still occurred and the BT still pointed
>         to fix_route_dialog() function. So it must be getting called
>         from within Dialog module somewhere. That BT is here:
>         http://pastebin.com/wu2X2Hxh
>
>         I collected this BT with loose_route() being called from my
>         script, but not validate_dialog() or fix_route_dialog():
>         http://pastebin.com/6V7yPaHF
>
>         This BT was collected with all three functions being called
>         from my script: http://pastebin.com/fZYYdndn
>
>         Ben Newlin
>
>         *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
>         <mailto:bogdan at opensips.org>
>         *Date: *Wednesday, July 27, 2016 at 3:57 AM
>         *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>, "Newlin, Ben"
>         <Ben.Newlin at inin.com> <mailto:Ben.Newlin at inin.com>
>         *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
>         Hi Ben,
>
>         First, if you use TH, makes no sense to do Record-Routing -
>         there are 2 SIP concepts that overlaps. You either act as an
>         end-point (by doing TH), either as a proxy (doing RR).
>
>         If doing TH, makes no sense to use validate + fix as these
>         functions check and repair the routing information in the
>         request (like Route and Contact headers). if you do TH, this
>         routing info is actually hidden and added by OpenSIPS, so
>         there is nothing to fix and repair.
>
>         Nevertheless, this should not crash or corrupt OpenSIPS. HAve
>         you managed to get a corefile ?
>
>         Also if you suspect memory corruption, you can compile-in the
>         memory debugger - see
>         http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .
>
>         Regards,
>
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 26.07.2016 23:20, Newlin, Ben wrote:
>
>             I have had 3 OpenSIPS server crashes in the last week. All
>             were due to segmentation faults. I was not able to capture
>             core dumps; I am configuring that now to catch the next crash.
>
>             My logs leading up to the crash are full of errors from
>             fix_route_dialog() complaining about invalid URIs for
>             sequential requests:
>
>             Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog:
>             Failed to parse SIP uri
>
>             Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state
>             0 parsed: <ip:1> (4) /
>             <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)
>
>             Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog:
>             Failed to parse SIP uri
>
>             Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state
>             0 parsed: <b0i2> (4) /
>             <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on>
>             (65)
>
>             Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog:
>             Failed to parse SIP uri
>
>             Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state
>             0 parsed: <ervi> (4) /
>             <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)
>
>             Many times the ?URI? displayed in the error message is
>             actually internal OpenSIPS variables, as in the last error
>             above. When they are from the SIP message, I have verified
>             that the messages themselves are correctly formatted. This
>             leads me to believe there is memory corruption occurring.
>
>             This all started when I updated my load-balancer servers
>             to use Record-Routing, specifically the ?double_rr?
>             mechanism for when multiple interfaces exist. The
>             Record-Routing is occurring on different servers which
>             have not crashed. Only the servers receiving the
>             Record-Routed messages are experiencing the errors.
>
>             Here is a piece of the code processing sequential
>             requests. I am using the topology_hiding() functionality
>             of the Dialog module. Are validate_dialog() and
>             fix_route_dialog() still valid in a topology_hiding scenario?
>
>             if (t_check_trans())
>
>             setflag(SEQ_REQUEST);
>
>               if (has_totag())
>
>               {
>
>             loose_route();
>
>                 if (match_dialog())
>
>                 {
>
>                   if (!validate_dialog())
>
>             fix_route_dialog();
>
>                   if (is_method("BYE"))
>
>             setflag(ACC_FLAG);
>
>             setflag(SEQ_REQUEST);
>
>                 }
>
>                 else if (!isflagset(SEQ_REQUEST))
>
>                 {
>
>                   if (!is_method("ACK")) {
>
>             route(rlog, LV_ERROR, "check_sequential", "Sequential
>             request not matched");
>
>               route(reply_error, "481", "Call Does Not Exist");
>
>                   }
>
>             return(EXIT);
>
>                 }
>
>               }
>
>             I will attempt to get core dumps of future crashes.
>
>             Thanks,
>
>             Ben Newlin
>
>
>
>
>
>
>
>             _______________________________________________
>
>             Users mailing list
>
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From john.quick at smartvox.co.uk  Wed Aug  3 15:18:30 2016
From: john.quick at smartvox.co.uk (John Quick)
Date: Wed, 3 Aug 2016 14:18:30 +0100
Subject: [OpenSIPS-Users] Memory leak v2.1.4
Message-ID: <000601d1ed89$874714a0$95d53de0$@smartvox.co.uk>

One of my OpenSIPS Proxy servers has a memory leak that is bad enough to
require the service to be restarted once a week.
It was running version 2.1.1 and I just upgraded it this morning to v.2.1.4,
but the memory leak is still happening.

What are the best next steps to help track down the problem?

John Quick
Smartvox Limited
Web: www.smartvox.co.uk




From pimenta at inatel.br  Wed Aug  3 16:55:21 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 3 Aug 2016 14:55:21 +0000
Subject: [OpenSIPS-Users] Function set_advertised_address() seems to have
 wrong decision. Workaround?
Message-ID: <BY2PR02MB165552FCF92069AF246044E2B5060@BY2PR02MB1655.namprd02.prod.outlook.com>



Hi.

The function set_advertised_address() is changing the wrong IP in record-routs from the SIP OK message.
I need to avoid this issue. The caller is in a remote network and the callee is in the local network.


What is my situation:
----------------------------
An UAS (the callee), running in the same hardware as OpenSIPS, is registered with IP 127.0.0.1. It could be 192.168.0.100 too. However, due to some specifics requirements of our project, such UAS must register itself using IP 127.0.0.1, not 192.168.0.100.


What is the problem:
---------------------------
When set_advertised_address("domain")  is called, for the SIP OK message, this function decides to change 127.0.0.1 to "domain".
For example, OpenSIPS receives:

SIP/2.0 200 OK
Via: SIP/2.0/TCP XXX.YYY.240.204:61871;rport=61871;received=131.221.240.204;branch=z9hG4bKPj8bd4d5988f4a4a0ba3599eba77f42600;alias
Record-Route: <sip:127.0.0.1;transport=tcp;lr;r2=on;did=862.47466066>
Record-Route: <sip:192.168.0.100;transport=tcp;lr;r2=on;did=862.47466066>

and change it to:

SIP/2.0 200 OK
Via: SIP/2.0/TCP XXX.YYY.240.204:61871;rport=61871;received=131.221.240.204;branch=z9hG4bKPj8bd4d5988f4a4a0ba3599eba77f42600;alias
Record-Route: <sip:domain;transport=tcp;lr;r2=on;did=862.47466066>
Record-Route: <sip:192.168.0.100;transport=tcp;lr;r2=on;did=862.47466066>

But, OpenSIPS should change the IP 192.168.0.100 to "domain", not the other Record-Route.
As I have this issue, the UAC can't send the SIP ACK confirming the SIP OK.


What I need to provide:
--------------------------------
I have to get/build a solution to make the set_advertised_address("domain") change the Record-Route that contains the IP 1921.168.0.100.

Maybe, the OpenSIPS always change the top most Record-Route. If it is true, I need a  workaround for it.

So, how can I fix the record-routs as I need? Does it make sense to do what I'm needing?
Any hint will be very helpful!
Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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From hunterj91 at hotmail.com  Wed Aug  3 18:15:06 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 3 Aug 2016 16:15:06 +0000
Subject: [OpenSIPS-Users] t_replicate and dispatcher module to forward
	REGISTER messages
Message-ID: <DUB112-W594BDBF6F3D87BC9D8D940BD060@phx.gbl>

Hi Guys,
We have an old platform we are looking to redesign but in the meantime we need to fix a problem we have.
Currently we have an opensips proxy/registrar with 2 x Asterisk boxes behind it.
Now currently we use t_replicate and the dispatcher module (ds_next_domain()) to send REGISTER requests in parallel to both Asterisk Servers so they can update their sip peers.
This works great but we sometimes (an issue we are investigating) dont always get a 200ok response back from one or both Asterisk servers for the REGISTER message.
Therefore I am looking to resend the REGISTER whenever this may happen.
This is fine using the fr timers should both Asterisk Servers not respond, however if only 1 Server doesnt respond, and the other does, this problem is lost, due to the parallel forking being applied to the t_replicate scenario.
I can capture 200 ok messages in onreply routes to confirm which Asterisk servers have responded, however I cannot t_relay again from this type of route.
Is there another way to do this? Whereby I can remain transaction stateful, and resend a REGISTER message if only one of the Asterisk Servers doesnt respond to a REGISTER message?
Thanks
Jon 		 	   		  
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From spanda at 3clogic.com  Thu Aug  4 12:43:00 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Thu, 4 Aug 2016 16:13:00 +0530
Subject: [OpenSIPS-Users] Fixing contact by opensips .
Message-ID: <CALf-Q44XBCfr3GS7u+WLV2qMJ0pvAFGy5mNKaFLJU90w_QgM5A@mail.gmail.com>

Hi All ,

      I two machine with opensips , One is peering server and another is
media proxy . Media proxy is changing the contact while forwarding the
request and response .

       As I know , Nat_traversal module was getting used for this
previously like bellow .
 * if (client_nat_test("8")) {*
*             xlog("L_NOTICE", "Contact changed  \n");*
*             fix_contact();*
*        };*
   This was getting used both is route and onreply_route . Depending on the
condition its changing the contact uri .

    Now I don't want my media proxy to change the contact in any condition
. I have disable nat_traversal module and the above portion in the config
file .

     Again its changing the contact . Is there any other module which do
the same this . Now its changing the contact in "200 Ok " .

    I am attaching my config file , please have a look into it and let me
know if this is possible and how can i do this .



*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From spanda at 3clogic.com  Thu Aug  4 13:15:44 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Thu, 4 Aug 2016 16:45:44 +0530
Subject: [OpenSIPS-Users] Fixing contact by opensips .
In-Reply-To: <CALf-Q44XBCfr3GS7u+WLV2qMJ0pvAFGy5mNKaFLJU90w_QgM5A@mail.gmail.com>
References: <CALf-Q44XBCfr3GS7u+WLV2qMJ0pvAFGy5mNKaFLJU90w_QgM5A@mail.gmail.com>
Message-ID: <CALf-Q47vYOQUCfT7GGpFM4OZOyJqotiYmandLNU1z4nEiKMwwA@mail.gmail.com>

I got the problem . Thanks

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Thu, Aug 4, 2016 at 4:13 PM, Sasmita Panda <spanda at 3clogic.com> wrote:

> Hi All ,
>
>       I two machine with opensips , One is peering server and another is
> media proxy . Media proxy is changing the contact while forwarding the
> request and response .
>
>        As I know , Nat_traversal module was getting used for this
> previously like bellow .
>  * if (client_nat_test("8")) {*
> *             xlog("L_NOTICE", "Contact changed  \n");*
> *             fix_contact();*
> *        };*
>    This was getting used both is route and onreply_route . Depending on
> the condition its changing the contact uri .
>
>     Now I don't want my media proxy to change the contact in any condition
> . I have disable nat_traversal module and the above portion in the config
> file .
>
>      Again its changing the contact . Is there any other module which do
> the same this . Now its changing the contact in "200 Ok " .
>
>     I am attaching my config file , please have a look into it and let me
> know if this is possible and how can i do this .
>
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
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From pimenta at inatel.br  Thu Aug  4 15:21:03 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Thu, 4 Aug 2016 13:21:03 +0000
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
Message-ID: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi.


How to discover if OpenSIPS is connected do Internet, from its configuration script?


Sometimes the Internet Link is down and then just local calls will work. If I can discover if OpenSIPS is "online" on Internet, I will use this information to implement some specific logic in my script.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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From liviu at opensips.org  Thu Aug  4 16:10:06 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Thu, 4 Aug 2016 17:10:06 +0300
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
In-Reply-To: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <b12cb71d-f380-98d0-b83c-5f2729f1bd28@opensips.org>

Hi, Rodrigo!


That's quite a fun question. Off the top of my head, here are 3 possible 
ways in which you can achieve this:


* relay over TCP: not sure if's relevant to your needs, but if you arm a 
failure route and t_relay("0x02") out to the internet, you will be able 
to properly tell if connectivity was down should you hit the failure route.

* ICMP test: you can do an exec("/bin/ping -w1 -c1 <ICMP-enabled public 
IP>") and decide from the return code

* HTTP GET: you can use the rest_client module, attempt to fetch some 
page, and decide from the return code. Be sure to set a proper TCP 
connect timeout!


Best regards,

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 04.08.2016 16:21, Rodrigo Pimenta Carvalho wrote:
>
> Hi.
>
>
> How to discover if OpenSIPS is connected do Internet, from its 
> configuration script?
>
>
> Sometimes the Internet Link is down and then just local calls will 
> work. If I can discover if OpenSIPS is "online" on Internet, I will 
> use this information to implement some specific logic in my script.
>
>
> Best regards.
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From pimenta at inatel.br  Thu Aug  4 16:35:01 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Thu, 4 Aug 2016 14:35:01 +0000
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
In-Reply-To: <b12cb71d-f380-98d0-b83c-5f2729f1bd28@opensips.org>
References: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>,
	<b12cb71d-f380-98d0-b83c-5f2729f1bd28@opensips.org>
Message-ID: <BY2PR02MB16558BF53E4469A31E3C38C2B5070@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi Liviu.


Thank you very much.

By the way, do you know if it is possible to discovery what is the IP address from where the OpenSIPS is running?

For example, the node can have an ethernet IP or a WLAN IP. It depends on if the node is connected to the DHCP server via wireless or cabe.


Can the OpenSIPS script tells me, by some way, what is the private IP address?


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Liviu Chircu <liviu at opensips.org>
Enviado: quinta-feira, 4 de agosto de 2016 11:10
Para: users at lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.


Hi, Rodrigo!


That's quite a fun question. Off the top of my head, here are 3 possible ways in which you can achieve this:


* relay over TCP: not sure if's relevant to your needs, but if you arm a failure route and t_relay("0x02") out to the internet, you will be able to properly tell if connectivity was down should you hit the failure route.

* ICMP test: you can do an exec("/bin/ping -w1 -c1 <ICMP-enabled public IP>") and decide from the return code

* HTTP GET: you can use the rest_client module, attempt to fetch some page, and decide from the return code. Be sure to set a proper TCP connect timeout!


Best regards,

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities.

On 04.08.2016 16:21, Rodrigo Pimenta Carvalho wrote:

Hi.


How to discover if OpenSIPS is connected do Internet, from its configuration script?


Sometimes the Internet Link is down and then just local calls will work. If I can discover if OpenSIPS is "online" on Internet, I will use this information to implement some specific logic in my script.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From pimenta at inatel.br  Thu Aug  4 16:47:50 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Thu, 4 Aug 2016 14:47:50 +0000
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
In-Reply-To: <BY2PR02MB16558BF53E4469A31E3C38C2B5070@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>,
	<b12cb71d-f380-98d0-b83c-5f2729f1bd28@opensips.org>,
	<BY2PR02MB16558BF53E4469A31E3C38C2B5070@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <BY2PR02MB16552A87EF1EFBEB77323F6DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi.


Forget the questions from my previous message, please.


Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Rodrigo Pimenta Carvalho <pimenta at inatel.br>
Enviado: quinta-feira, 4 de agosto de 2016 11:35
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.


Hi Liviu.


Thank you very much.

By the way, do you know if it is possible to discovery what is the IP address from where the OpenSIPS is running?

For example, the node can have an ethernet IP or a WLAN IP. It depends on if the node is connected to the DHCP server via wireless or cabe.


Can the OpenSIPS script tells me, by some way, what is the private IP address?


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Liviu Chircu <liviu at opensips.org>
Enviado: quinta-feira, 4 de agosto de 2016 11:10
Para: users at lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.


Hi, Rodrigo!


That's quite a fun question. Off the top of my head, here are 3 possible ways in which you can achieve this:


* relay over TCP: not sure if's relevant to your needs, but if you arm a failure route and t_relay("0x02") out to the internet, you will be able to properly tell if connectivity was down should you hit the failure route.

* ICMP test: you can do an exec("/bin/ping -w1 -c1 <ICMP-enabled public IP>") and decide from the return code

* HTTP GET: you can use the rest_client module, attempt to fetch some page, and decide from the return code. Be sure to set a proper TCP connect timeout!


Best regards,

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities.

On 04.08.2016 16:21, Rodrigo Pimenta Carvalho wrote:

Hi.


How to discover if OpenSIPS is connected do Internet, from its configuration script?


Sometimes the Internet Link is down and then just local calls will work. If I can discover if OpenSIPS is "online" on Internet, I will use this information to implement some specific logic in my script.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From Agalya_Ramachandran at comcast.com  Thu Aug  4 22:14:23 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 4 Aug 2016 20:14:23 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
Message-ID: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>

Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From liviu at opensips.org  Fri Aug  5 10:46:47 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Fri, 5 Aug 2016 11:46:47 +0300
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
In-Reply-To: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
Message-ID: <d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have 
to use 2.1+

Async rest_put() is currently not implemented, but its logic would be 
fairly similar to the GET and POST counterparts.

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
>
> Hi team,
>
> We are using opensips1.11.3 version for our project.
>
> I have a question here. Right now we are using a REST API call from 
> config file for our project requirement and it is in synchronous mode 
> currently.
>
> From the opensips document, I see asynchronous support at script level 
> can be done in 2.1 version.
>
> Is it possible to change in asynchronous mode in 1.11 version?
>
> If not, is there a way to achieve asynchronous REST calls for the 
> version of opensips what we are using?
>
> Also I see from documentation,
>
> The following functions may also be called asynchronously:
>
>   * *avp_db_query*
>     <http://www.opensips.org/html/docs/modules/2.1.x/avpops.html#id294986>
>   * *rest_get*
>     <http://www.opensips.org/html/docs/modules/2.1.x/rest_client.html#id293741>
>   * *rest_post*
>     <http://www.opensips.org/html/docs/modules/2.1.x/rest_client.html#id293886>
>   * *exec* <http://www.opensips.org/html/docs/modules/2.1.x/exec#id294052>
>
> Can rest_put can also be called asynchronously if we use 2.1 version ?
>
> Please clarify.
>
> Regards,
>
> Agalya
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Ben.Newlin at inin.com  Fri Aug  5 16:22:32 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Fri, 5 Aug 2016 14:22:32 +0000
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
	<579B4EC1.4010809@opensips.org>
	<418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>
	<0d0dbc56-cc82-c884-ee04-a81afb2b8711@opensips.org>
Message-ID: <25602746-D95E-49D2-8792-BE16F18CA642@inin.com>

Bogdan,

Just as an update, this does not work. The match_dialog function must do loose routing on its own and even though I call remove_hf(?Route?) before match_dialog(), it still processes the Route header on the incoming message. So match_dialog returns true, but the TH refactoring is not applied and $du is set to the IP from the incoming message?s Route header, which is my server.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Monday, August 1, 2016 at 7:13 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

I see your problem here. So, let's explore this:
1) for sending the call to carrier, on OpenSIPS, you do TH (with advertise) resulting in a Contact with the public IP of the SBC.
2) also, manually add a RR header with the private IP of OpenSIPS.
3) send call to SBC, which will add its own RR stuff.

Now, on the sequential request from Carrier, the RURI will contain the Contact of OpenSIPS (the pub IP of SBC), some Route hdrs due the SBC and the Route we added on OpenSIPS.
- when request gets to SBC, the SBC will do loose route, consume its Route headers, and it will use the next available Route which points to the priv IP of OpenSIPS (and it will not use the public IP in RURI for routing)
- requests gets to OpenSIPS, simply remove_hf() and Route headers (do not do any loose_route() as it is useless) and hit th_matching -> this will refactor the request (RURI, Contact, Route) for the leg on the other side -> this should be fine.

For the other direction (still sequential), you do th_matching on OpenSIPS and nothing more. This will send a request holding the Routes due the SBC, a Contact with the public IP and and RURI pointing probably to the carrier.

Shortly you do standard TH, but on outbound scenario, add a fake RR header to trick the SBC to route the sequential to your OpenSIPS.

Does it make sense ?


Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 29.07.2016 18:05, Newlin, Ben wrote:
Here is the scenario:

My servers are only listening on a private IP address. There is a public address on our SBC. I have a carrier that requires that the Contact IP address matches the public address we provided to them. So when I do TH on my server I have to also do set_advertised_address to advertise the public address in the Contact header. Sequential requests use the Contact as the Request URI and the SBC is doing RR so all requests will come back through it. When the SBC receives a sequential request it strips its Route headers and forwards to the Request URI (previous Contact URI). But that URI now points back to the SBC, so it cannot deliver the request. If my server could Record-Route the initial request with its private address, the SBC would be able to route the request back to it.

Here is a trace of my scenario: http://pastebin.com/x927mFtq. I created it with SIPp so some endpoints are on the same IPs but with different ports. The public IP is 192.168.99.100, with port 7060 representing the PSTN carrier and port 5060 the SBC. The private IP is 10.0.2.15, with port 5060 being the SBC again and port 6060 being my server. You can see that the ACK cannot be delivered correctly to my server with this configuration.

The problem here stems from the fact that Topology Hiding should really be done at the edge of the network in order to be most effective. But my OpenSIPS server doesn?t sit on the Edge of the network, it is behind an SBC. So TH is complicated by the fact that my OpenSIPS has no public IP of its own and must advertise the public IP of the SBC instead, but future requests must still be routable into the private network. I am really using TH to hide the two ends of the call from each other, not to hide my internal network topology.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Friday, July 29, 2016 at 8:40 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

Sorry to disagree, but IMHO they do exclude one each other.

Adding RR to TH should not be seen as a way of fixing some broken TH scenarios (with advertise).
So, let me try to understand what is not working for you. You do TH and advertise. In this case, normally, in the Contact headers generated by OpenSIPS (as a result of TH), it should be the TH interface, right ? What exactly seems to be the problem ? do you have a trace to show the issues ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 16:05, Newlin, Ben wrote:
I understand that normally you would not need RR with TH, but the two concepts are not mutually exclusive in SIP. As I said, I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. This means that TH will populate the Contact header with the advertised address and if I cannot Record-Route with the actual address then I will not receive sequential requests.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:59 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR mechanism. If you do TH, your OpenSIPS will act as and end point (from SIP perspective), so there will be no Route/RR headers at all. So no need to do loose_route or so. You just do TH matching for the sequential requests and nothing more.

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,

I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog topology_hiding functionality. The topology_hiding isn?t a true B2BUA, but it does set up different parameters for the incoming UAC and outgoing UAS sides of the call for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on different sides of the call. The record_route() function will only add a Record-Route header to the outgoing UAS side of the call. And since the record_route() function cannot be called from onreply_route, but is no way to add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC side of the call and there is no way to perform loose_route() on the UAS side of the call.

So there is a situation where Record-Route headers can only be added on the outgoing UAS side, but the associated Route headers can only be removed on the incoming UAC side (where they won?t exist since they can?t be added) and any added headers on the UAS side cannot be processed properly due to the lack of loose_route.

I can provide further information if this is unclear. It should be easily reproducible by attempting to use record_route in a topology_hiding scenario. The route is added to the outbound leg, but is not removed by loose_route so the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.newlin at inin.com<mailto:ben.newlin at inin.com>
[ge removed by sender.]
www.inin.com<http://www.inin.com>







_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

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From adrian.fretwell at topgreen.co.uk  Fri Aug  5 17:14:10 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Fri, 5 Aug 2016 16:14:10 +0100
Subject: [OpenSIPS-Users] Topology Hiding module Call_ID encryption
Message-ID: <6e5cd9b4-3b09-9bad-fcb0-a2775ac23afb@topgreen.co.uk>

Hello, may I ask for a little help?

Using Topology Hiding Module in Opensips 2.1.2. I'm finding that looking 
at a tcpdump the Call_ID looks like this:

LTXCH01_c1FNUVlRXkNvaA--

But when I log $TH_callee_callid "xlog("L_INFO", "sequential: Callee 
side callid is $TH_callee_callid");" the Call_ID looks like this:

LTXCH01_c1FNUVlRXkNvaA==

Notice the dashes are replaced with equals signs.

If I rewrite the Call-ID for NOTIFY messages with the value of 
$TH_callee_callid the tcpdump will show the Call-ID with the equals 
signs (==) not the dashes

         if (is_method("NOTIFY")) {
             if( !remove_hf("Call-ID")) {
                 sl_send_reply("503", "Service Unavailable");
                 xlog("L_INFO", "sequential: could not remove header");
                 exit;
             }
             append_hf("Call-ID: $TH_callee_callid\r\n", "To");
         }

Could this be a bug or am I doing something silly?

Kind regards,

Adrian Fretwell
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From Agalya_Ramachandran at comcast.com  Fri Aug  5 20:49:25 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Fri, 5 Aug 2016 18:49:25 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
In-Reply-To: <d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
Message-ID: <b3f92e559f4943759562f73b0cef626c@COPDCEX28.cable.comcast.com>

Hi Liviu,

Thanks for quick response and clarification.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.


Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From nabeelshikder at gmail.com  Sat Aug  6 01:56:26 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sat, 6 Aug 2016 00:56:26 +0100
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
Message-ID: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>

Hi,

OpenSIPS does not receive a '486 Busy' signal when a callee is using the
Asterisk voicemail service. If a user is currently listening to voicemail
or leaving a voice message, an attempt to call that user should result in a
'486 Busy' signal. From that response, I will be able to play a busy tone
on the caller's side. Please let me know how to fix this.

Nabeel
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From bogdan at opensips.org  Sun Aug  7 19:15:17 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Sun, 7 Aug 2016 20:15:17 +0300
Subject: [OpenSIPS-Users] Memory leak v2.1.4
In-Reply-To: <000601d1ed89$874714a0$95d53de0$@smartvox.co.uk>
References: <000601d1ed89$874714a0$95d53de0$@smartvox.co.uk>
Message-ID: <34d88fb6-bf3b-9f72-dc4b-b665eabb2c43@opensips.org>

Hi John,

Please see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem

Let me know if you additional help .

Best Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03.08.2016 16:18, John Quick wrote:
> One of my OpenSIPS Proxy servers has a memory leak that is bad enough to
> require the service to be restarted once a week.
> It was running version 2.1.1 and I just upgraded it this morning to v.2.1.4,
> but the memory leak is still happening.
>
> What are the best next steps to help track down the problem?
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



From bogdan at opensips.org  Sun Aug  7 19:31:01 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Sun, 7 Aug 2016 20:31:01 +0300
Subject: [OpenSIPS-Users] Function set_advertised_address() seems to
 have wrong decision. Workaround?
In-Reply-To: <BY2PR02MB165552FCF92069AF246044E2B5060@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB165552FCF92069AF246044E2B5060@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <441c29a8-259e-e3c1-9fd7-3ad9feec950c@opensips.org>

Hi Rodrigo,

The set_advertised_address() is not changing anything by itself, but 
setting (as the name says) what should be the IP when comes to 
advertising OpenSIPS in new headers like VIA or Record-Route - once 
again, when building and inserting such headers. This happens for 
requests, not for replies.

So, I do not understand your case with calling the function for a 200 OK 
:( .Do you have a trace at opensips level ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03.08.2016 17:55, Rodrigo Pimenta Carvalho wrote:
>
>
> Hi.
>
> The function set_advertised_address() is changing the wrong IP in 
> record-routs from the SIP OK message.
> I need to avoid this issue. The caller is in a remote network and the 
> callee is in the local network.
>
>
> What is my situation:
> ----------------------------
> An UAS (the callee), running in the same hardware as OpenSIPS, is 
> registered with IP 127.0.0.1. It could be 192.168.0.100 too. However, 
> due to some specifics requirements of our project, such UAS must 
> register itself using IP 127.0.0.1, not 192.168.0.100.
>
>
> What is the problem:
> ---------------------------
> When set_advertised_address("domain")  is called, for the SIP OK 
> message, this function decides to change 127.0.0.1 to "domain".
> For example, OpenSIPS receives:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 
> XXX.YYY.240.204:61871;rport=61871;received=131.221.240.204;branch=z9hG4bKPj8bd4d5988f4a4a0ba3599eba77f42600;alias
> Record-Route: <sip:127.0.0.1;transport=tcp;lr;r2=on;did=862.47466066>
> Record-Route: <sip:192.168.0.100;transport=tcp;lr;r2=on;did=862.47466066>
>
> and change it to:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 
> XXX.YYY.240.204:61871;rport=61871;received=131.221.240.204;branch=z9hG4bKPj8bd4d5988f4a4a0ba3599eba77f42600;alias
> Record-Route: <sip:domain;transport=tcp;lr;r2=on;did=862.47466066>
> Record-Route: <sip:192.168.0.100;transport=tcp;lr;r2=on;did=862.47466066>
>
> But, OpenSIPS should change the IP 192.168.0.100 to "domain", not the 
> other Record-Route.
> As I have this issue, the UAC can't send the SIP ACK confirming the 
> SIP OK.
>
>
> What I need to provide:
> --------------------------------
> I have to get/build a solution to make the 
> set_advertised_address("domain") change the Record-Route that contains 
> the IP 1921.168.0.100.
>
> Maybe, the OpenSIPS always change the top most Record-Route. If it is 
> true, I need a  workaround for it.
>
> So, how can I fix the record-routs as I need? Does it make sense to do 
> what I'm needing?
> Any hint will be very helpful!
> Best regards.
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Sun Aug  7 19:39:59 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Sun, 7 Aug 2016 20:39:59 +0300
Subject: [OpenSIPS-Users] t_replicate and dispatcher module to forward
 REGISTER messages
In-Reply-To: <DUB112-W594BDBF6F3D87BC9D8D940BD060@phx.gbl>
References: <DUB112-W594BDBF6F3D87BC9D8D940BD060@phx.gbl>
Message-ID: <fbd2fd9b-a6b6-921e-3705-26995d38c2c8@opensips.org>

Hi Jonathan,

Unfortunately this is a design problem - the t_replicate() is actually a 
t_reply() but without passing the received replies back to the caller. 
So t_replicate() behaves on parallel forking exactly as t_reply().

Some additional work may be required to change this and have 
t_replicate() properly replicating to all destination. Could you please 
open a ticket on github about this ?

Thanks and Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03.08.2016 19:15, Jonathan Hunter wrote:
> Hi Guys,
>
> We have an old platform we are looking to redesign but in the meantime 
> we need to fix a problem we have.
>
> Currently we have an opensips proxy/registrar with 2 x Asterisk boxes 
> behind it.
>
> Now currently we use t_replicate and the dispatcher module 
> (ds_next_domain()) to send REGISTER requests in parallel to both 
> Asterisk Servers so they can update their sip peers.
>
> This works great but we sometimes (an issue we are investigating) dont 
> always get a 200ok response back from one or both Asterisk servers for 
> the REGISTER message.
>
> Therefore I am looking to resend the REGISTER whenever this may happen.
>
> This is fine using the fr timers should both Asterisk Servers not 
> respond, however if only 1 Server doesnt respond, and the other does, 
> this problem is lost, due to the parallel forking being applied to the 
> t_replicate scenario.
>
> I can capture 200 ok messages in onreply routes to confirm which 
> Asterisk servers have responded, however I cannot t_relay again from 
> this type of route.
>
> Is there another way to do this? Whereby I can remain transaction 
> stateful, and resend a REGISTER message if only one of the Asterisk 
> Servers doesnt respond to a REGISTER message?
>
> Thanks
>
> Jon
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Sun Aug  7 19:50:24 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Sun, 7 Aug 2016 20:50:24 +0300
Subject: [OpenSIPS-Users] Topology Hiding module Call_ID encryption
In-Reply-To: <6e5cd9b4-3b09-9bad-fcb0-a2775ac23afb@topgreen.co.uk>
References: <6e5cd9b4-3b09-9bad-fcb0-a2775ac23afb@topgreen.co.uk>
Message-ID: <4362b1e1-180f-8ac3-71ee-7f7247b1c605@opensips.org>

Hi Adrian,

The code (for the $TH_callee_callid variable) should convert the 
internal '=' into '-', but it looks like it fails. Is this happening all 
the time for you ? (I'm just looking for an easy way to reproduce).

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.08.2016 18:14, Adrian Fretwell wrote:
>
> Hello, may I ask for a little help?
>
> Using Topology Hiding Module in Opensips 2.1.2.  I'm finding that 
> looking at a tcpdump the Call_ID looks like this:
>
> LTXCH01_c1FNUVlRXkNvaA--
>
> But when I log $TH_callee_callid "xlog("L_INFO", "sequential: Callee 
> side callid is $TH_callee_callid");" the Call_ID looks like this:
>
> LTXCH01_c1FNUVlRXkNvaA==
>
> Notice the dashes are replaced with equals signs.
>
> If I rewrite the Call-ID for NOTIFY messages with the value of 
> $TH_callee_callid the tcpdump will show the Call-ID with the equals 
> signs (==) not the dashes
>
>         if (is_method("NOTIFY")) {
>             if( !remove_hf("Call-ID")) {
>                 sl_send_reply("503", "Service Unavailable");
>                 xlog("L_INFO", "sequential: could not remove header");
>                 exit;
>             }
>             append_hf("Call-ID: $TH_callee_callid\r\n", "To");
>         }
>
> Could this be a bug or am I doing something silly?
>
> Kind regards,
>
> Adrian Fretwell
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Sun Aug  7 19:57:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Sun, 7 Aug 2016 20:57:08 +0300
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
In-Reply-To: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
References: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
Message-ID: <698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>

Hi Nabeel,

Who should generate the 486 Busy reply ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 06.08.2016 02:56, Nabeel wrote:
> Hi,
>
> OpenSIPS does not receive a '486 Busy' signal when a callee is using 
> the Asterisk voicemail service. If a user is currently listening to 
> voicemail or leaving a voice message, an attempt to call that user 
> should result in a '486 Busy' signal. From that response, I will be 
> able to play a busy tone on the caller's side. Please let me know how 
> to fix this.
>
> Nabeel
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From nabeelshikder at gmail.com  Mon Aug  8 00:44:54 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sun, 7 Aug 2016 23:44:54 +0100
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
In-Reply-To: <698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>
References: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
	<698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>
Message-ID: <CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>

Hi,

The busy reply should be generated in exactly the same way as when the
callee is busy on the phone with another SIP user (from the OpenSIPS
subscriber table). The only difference is that the callee is on the phone
with asterisk voicemail instead of a SIP user from the subscriber table.
I'm not sure who exactly should generate the reply but I think it depends
on how OpenSIPS is interacting with the Voicemail server during a call to
voicemail. Perhaps it's not treating the voicemail service as another
'user' so the caller didn't receive the busy signal.

Nabeel

On 7 Aug 2016 6:57 p.m., "Bogdan-Andrei Iancu" <bogdan at opensips.org> wrote:

> Hi Nabeel,
>
> Who should generate the 486 Busy reply ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 06.08.2016 02:56, Nabeel wrote:
>
> Hi,
>
> OpenSIPS does not receive a '486 Busy' signal when a callee is using the
> Asterisk voicemail service. If a user is currently listening to voicemail
> or leaving a voice message, an attempt to call that user should result in a
> '486 Busy' signal. From that response, I will be able to play a busy tone
> on the caller's side. Please let me know how to fix this.
>
> Nabeel
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From johan at democon.be  Mon Aug  8 08:58:57 2016
From: johan at democon.be (Johan De Clercq)
Date: Mon, 8 Aug 2016 08:58:57 +0200
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
In-Reply-To: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <CAEVWGm8EwO=6zr7GzNzuDK4EDBTt8e-1c6FyO_qRC=fUYgaZ5A@mail.gmail.com>

create a start up route startup_route, the use module exec to f.e. get your
pub ip with curl.

2016-08-04 15:21 GMT+02:00 Rodrigo Pimenta Carvalho <pimenta at inatel.br>:

> Hi.
>
>
> How to discover if OpenSIPS is connected do Internet, from its
> configuration script?
>
>
> Sometimes the Internet Link is down and then just local calls will work.
> If I can discover if OpenSIPS is "online" on Internet, I will use this
> information to implement some specific logic in my script.
>
>
> Best regards.
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From adrian.fretwell at topgreen.co.uk  Mon Aug  8 09:27:57 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Mon, 8 Aug 2016 08:27:57 +0100
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
In-Reply-To: <CAEVWGm8EwO=6zr7GzNzuDK4EDBTt8e-1c6FyO_qRC=fUYgaZ5A@mail.gmail.com>
References: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAEVWGm8EwO=6zr7GzNzuDK4EDBTt8e-1c6FyO_qRC=fUYgaZ5A@mail.gmail.com>
Message-ID: <bf742840-f151-712f-8fe6-1cf886d24174@topgreen.co.uk>

Johan,

If your internet connection is going up and down regularly, you may be 
better off executing your test from a timer route:

timer_route[internet_check, 300] {
     # -----------------------------------------------------------------
     # Timer Route every 5 minutes
     # -----------------------------------------------------------------

    exec("/usr/local/bin/some_check _script");
}

There are many different ways to check if you have an internet 
connection, the way you do it will depend on your environment and 
application, but here is a very simple shell script as an example:

#!/bin/bash
ping -c 2 8.8.8.8 > /dev/null
if [ $? -eq 0 ]; then
echo "Internet Alive $(date)";
else
echo "Internet Dead $(date)";
fi

Kind regards,

Adrian Fretwell

On 08/08/16 07:58, Johan De Clercq wrote:
> create a start up route startup_route, the use module exec to f.e. get 
> your pub ip with curl.
>
> 2016-08-04 15:21 GMT+02:00 Rodrigo Pimenta Carvalho <pimenta at inatel.br 
> <mailto:pimenta at inatel.br>>:
>
>     Hi.
>
>
>     How to discover if OpenSIPS is connected do Internet, from its
>     configuration script?
>
>
>     Sometimes the Internet Link is down and then just local calls will
>     work. If I can discover if OpenSIPS is "online" on Internet, I
>     will use this information to implement some specific logic in my
>     script.
>
>
>     Best regards.
>
>
>     RODRIGO PIMENTA CARVALHO
>     Inatel Competence Center
>     Software
>     Ph: +55 35 3471 9200 <tel:%2B55%2035%203471%209200> RAMAL 979
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From hunterj91 at hotmail.com  Mon Aug  8 11:43:26 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Mon, 8 Aug 2016 09:43:26 +0000
Subject: [OpenSIPS-Users] t_replicate and dispatcher module to forward
 REGISTER messages
In-Reply-To: <fbd2fd9b-a6b6-921e-3705-26995d38c2c8@opensips.org>
References: <DUB112-W594BDBF6F3D87BC9D8D940BD060@phx.gbl>,
	<fbd2fd9b-a6b6-921e-3705-26995d38c2c8@opensips.org>
Message-ID: <DUB130-W674D81D0B726B05A988D7CBD1B0@phx.gbl>

Issue opened on github, thanks Bogdan.
#943.
Jon

Subject: Re: [OpenSIPS-Users] t_replicate and dispatcher module to forward REGISTER messages
To: users at lists.opensips.org; hunterj91 at hotmail.com
From: bogdan at opensips.org
Date: Sun, 7 Aug 2016 20:39:59 +0300


  
    
  
  
    Hi Jonathan,

      

      Unfortunately this is a design problem - the t_replicate() is
      actually a t_reply() but without passing the received replies back
      to the caller. So t_replicate() behaves on parallel forking
      exactly as t_reply().

      

      Some additional work may be required to change this and have
      t_replicate() properly replicating to all destination. Could you
      please open a ticket on github about this ?

      

      Thanks and Regards,

    
    Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
    On 03.08.2016 19:15, Jonathan Hunter
      wrote:

    
    
      
      
        Hi Guys,
        

        
        We have an old platform we are looking to redesign but in
          the meantime we need to fix a problem we have.
        

        
        Currently we have an opensips proxy/registrar with 2 x
          Asterisk boxes behind it.
        

        
        Now currently we use t_replicate and the dispatcher module
          (ds_next_domain()) to send REGISTER requests in parallel to
          both Asterisk Servers so they can update their sip peers.
        

        
        This works great but we sometimes (an issue we are
          investigating) dont always get a 200ok response back from one
          or both Asterisk servers for the REGISTER message.
        

        
        Therefore I am looking to resend the REGISTER whenever this
          may happen.
        

        
        This is fine using the fr timers should both Asterisk
          Servers not respond, however if only 1 Server doesnt respond,
          and the other does, this problem is lost, due to the parallel
          forking being applied to the t_replicate scenario.
        

        
        I can capture 200 ok messages in onreply routes to confirm
          which Asterisk servers have responded, however I cannot
          t_relay again from this type of route.
        

        
        Is there another way to do this? Whereby I can remain
          transaction stateful, and resend a REGISTER message if only
          one of the Asterisk Servers doesnt respond to a REGISTER
          message?
        

        
        Thanks
        

        
        Jon
      
      

      
      

      _______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

    
    
 		 	   		  
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From ziv at communitake.com  Mon Aug  8 11:58:58 2016
From: ziv at communitake.com (Ziv Gabel)
Date: Mon, 8 Aug 2016 12:58:58 +0300
Subject: [OpenSIPS-Users] Monitoring end to end solution.
Message-ID: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>

Hi,

I'm running several opensips servers and I want to monitor the health of the
servers.

I can check for service status and an open port, but what I'm looking for is
a way to make an "end to end test" that will include a registration, and a
call.

The test should check that

1.       Registration is working.

2.       Call goes through 

3.       Rtp (I use rtpengine) is flowing currectly.

 

Is there any known way to do that or should I figure out I way myself.

 

Ziv Gabel l Professional services l CommuniTake Technologies Ltd.

 

M: +972535265553 l Skype: ziv_gabel l E:  <mailto:ziv at communitake.com>
ziv at communitake.com

T: +972.4.696.8908 l F: +972.4.959.1654 l  <http://www.communitake.com/>
www.communitake.com

 

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From govoiper at gmail.com  Mon Aug  8 12:52:13 2016
From: govoiper at gmail.com (SamyGo)
Date: Mon, 8 Aug 2016 06:52:13 -0400
Subject: [OpenSIPS-Users] Monitoring end to end solution.
In-Reply-To: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
References: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
Message-ID: <CAJUJwth-CHAgnUdjm2W_qFKHBf2Qpq10sNqOWwPW6H3CYbdRfQ@mail.gmail.com>

Hi Ziv,
You should create scripts based on SIPP to execute these scenarios, and
based on results evaluate status of the servers.

Regards,
Sammy

On Aug 8, 2016 05:59, "Ziv Gabel" <ziv at communitake.com> wrote:

> Hi,
>
> I?m running several opensips servers and I want to monitor the health of
> the servers.
>
> I can check for service status and an open port, but what I?m looking for
> is a way to make an ?end to end test? that will include a registration, and
> a call.
>
> The test should check that
>
> 1.       Registration is working.
>
> 2.       Call goes through
>
> 3.       Rtp (I use rtpengine) is flowing currectly.
>
>
>
> Is there any known way to do that or should I figure out I way myself.
>
>
>
> *Ziv Gabel *l Professional services l CommuniTake Technologies Ltd.
>
>
>
> *M*: +972535265553 l *Skype*: ziv_gabel l *E*: ziv at communitake.com
>
> *T*: +972.4.696.8908 l *F*: +972.4.959.1654 l www.communitake.com
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From johan at democon.be  Mon Aug  8 12:53:45 2016
From: johan at democon.be (Johan De Clercq)
Date: Mon, 8 Aug 2016 12:53:45 +0200
Subject: [OpenSIPS-Users] Monitoring end to end solution.
In-Reply-To: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
References: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
Message-ID: <CAEVWGm9xU+RN3XuAKwenLPFatekUqFiTnoG7M9rOzE9y-j4H1Q@mail.gmail.com>

Maybe you can script everything out in sipp ?

2016-08-08 11:58 GMT+02:00 Ziv Gabel <ziv at communitake.com>:

> Hi,
>
> I?m running several opensips servers and I want to monitor the health of
> the servers.
>
> I can check for service status and an open port, but what I?m looking for
> is a way to make an ?end to end test? that will include a registration, and
> a call.
>
> The test should check that
>
> 1.       Registration is working.
>
> 2.       Call goes through
>
> 3.       Rtp (I use rtpengine) is flowing currectly.
>
>
>
> Is there any known way to do that or should I figure out I way myself.
>
>
>
> *Ziv Gabel *l Professional services l CommuniTake Technologies Ltd.
>
>
>
> *M*: +972535265553 l *Skype*: ziv_gabel l *E*: ziv at communitake.com
>
> *T*: +972.4.696.8908 l *F*: +972.4.959.1654 l www.communitake.com
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From ziv at communitake.com  Mon Aug  8 13:02:18 2016
From: ziv at communitake.com (Ziv Gabel)
Date: Mon, 8 Aug 2016 14:02:18 +0300
Subject: [OpenSIPS-Users] Monitoring end to end solution.
In-Reply-To: <CAEVWGm9xU+RN3XuAKwenLPFatekUqFiTnoG7M9rOzE9y-j4H1Q@mail.gmail.com>
References: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
	<CAEVWGm9xU+RN3XuAKwenLPFatekUqFiTnoG7M9rOzE9y-j4H1Q@mail.gmail.com>
Message-ID: <00ab01d1f164$55a7a9d0$00f6fd70$@communitake.com>

The problem with sipp is that it does not support 2 call-ids in the same script (register and invite).

 

Is it possible to have static registrations in opensips ?

I can then use one script for registration test and one for the invite.

 

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Johan De Clercq
Sent: ??? ? 08 ?????? 2016 13:54
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Monitoring end to end solution.

 

Maybe you can script everything out in sipp ?

 

2016-08-08 11:58 GMT+02:00 Ziv Gabel <ziv at communitake.com>:

Hi,

I?m running several opensips servers and I want to monitor the health of the servers.

I can check for service status and an open port, but what I?m looking for is a way to make an ?end to end test? that will include a registration, and a call.

The test should check that

1.       Registration is working.

2.       Call goes through 

3.       Rtp (I use rtpengine) is flowing currectly.

 

Is there any known way to do that or should I figure out I way myself.

 

Ziv Gabel l Professional services l CommuniTake Technologies Ltd.

 

M: +972535265553 <tel:%2B972535265553>  l Skype: ziv_gabel l E: ziv at communitake.com

T: +972.4.696.8908 <tel:%2B972.4.696.8908>  l F: +972.4.959.1654 <tel:%2B972.4.959.1654>  l  <http://www.communitake.com/> www.communitake.com

 


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

 

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From pimenta at inatel.br  Mon Aug  8 13:51:27 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Mon, 8 Aug 2016 11:51:27 +0000
Subject: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
 Internet, from the script.
In-Reply-To: <bf742840-f151-712f-8fe6-1cf886d24174@topgreen.co.uk>
References: <BY2PR02MB165570C1E0BB607F1367A66DB5070@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAEVWGm8EwO=6zr7GzNzuDK4EDBTt8e-1c6FyO_qRC=fUYgaZ5A@mail.gmail.com>,
	<bf742840-f151-712f-8fe6-1cf886d24174@topgreen.co.uk>
Message-ID: <BY2PR02MB16554310F19E4410CDAE5F38B51B0@BY2PR02MB1655.namprd02.prod.outlook.com>

Thank all of you.

Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Adrian Fretwell <adrian.fretwell at topgreen.co.uk>
Enviado: segunda-feira, 8 de agosto de 2016 04:27
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.


Johan,

If your internet connection is going up and down regularly, you may be better off executing your test from a timer route:

timer_route[internet_check, 300] {
    # -----------------------------------------------------------------
    # Timer Route every 5 minutes
    # -----------------------------------------------------------------

   exec("/usr/local/bin/some_check _script");
}


There are many different ways to check if you have an internet connection, the way you do it will depend on your environment and application, but here is a very simple shell script as an example:

#!/bin/bash
ping -c 2 8.8.8.8 > /dev/null
if [ $? -eq 0 ]; then
echo "Internet Alive $(date)";
else
echo "Internet Dead $(date)";
fi

Kind regards,

Adrian Fretwell

On 08/08/16 07:58, Johan De Clercq wrote:
create a start up route startup_route, the use module exec to f.e. get your pub ip with curl.

2016-08-04 15:21 GMT+02:00 Rodrigo Pimenta Carvalho <pimenta at inatel.br<mailto:pimenta at inatel.br>>:

Hi.


How to discover if OpenSIPS is connected do Internet, from its configuration script?


Sometimes the Internet Link is down and then just local calls will work. If I can discover if OpenSIPS is "online" on Internet, I will use this information to implement some specific logic in my script.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users





_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From faheem2084 at gmail.com  Mon Aug  8 17:40:42 2016
From: faheem2084 at gmail.com (Faheem Muhammad)
Date: Mon, 8 Aug 2016 20:40:42 +0500
Subject: [OpenSIPS-Users] Monitoring end to end solution.
In-Reply-To: <00ab01d1f164$55a7a9d0$00f6fd70$@communitake.com>
References: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
	<CAEVWGm9xU+RN3XuAKwenLPFatekUqFiTnoG7M9rOzE9y-j4H1Q@mail.gmail.com>
	<00ab01d1f164$55a7a9d0$00f6fd70$@communitake.com>
Message-ID: <CA++nn_y5eacfpPBtE9o1SejAgq-MKnLJJNtjd_i+pC7hyLTH-w@mail.gmail.com>

No, with SIPp you can create any type of call scenario including REGISTER +
INVITE scenarios.
Follow the below mentioned blog entry 'Scenario 2'.
I posted it few years back. Unfortunately original domain is no more
available, but archive link works.

http://web.archive.org/web/20130924010913/http://techvoiper.com/sipp-load-scenario-generation-overview/

Regards,
Muhammad Faheem

On Mon, Aug 8, 2016 at 4:02 PM, Ziv Gabel <ziv at communitake.com> wrote:

> The problem with sipp is that it does not support 2 call-ids in the same
> script (register and invite).
>
>
>
> Is it possible to have static registrations in opensips ?
>
> I can then use one script for registration test and one for the invite.
>
>
>
> *From:* users-bounces at lists.opensips.org [mailto:users-bounces at lists.
> opensips.org] *On Behalf Of *Johan De Clercq
> *Sent:* ??? ? 08 ?????? 2016 13:54
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Monitoring end to end solution.
>
>
>
> Maybe you can script everything out in sipp ?
>
>
>
> 2016-08-08 11:58 GMT+02:00 Ziv Gabel <ziv at communitake.com>:
>
> Hi,
>
> I?m running several opensips servers and I want to monitor the health of
> the servers.
>
> I can check for service status and an open port, but what I?m looking for
> is a way to make an ?end to end test? that will include a registration, and
> a call.
>
> The test should check that
>
> 1.       Registration is working.
>
> 2.       Call goes through
>
> 3.       Rtp (I use rtpengine) is flowing currectly.
>
>
>
> Is there any known way to do that or should I figure out I way myself.
>
>
>
> *Ziv Gabel *l Professional services l CommuniTake Technologies Ltd.
>
>
>
> *M*: +972535265553 l *Skype*: ziv_gabel l *E*: ziv at communitake.com
>
> *T*: +972.4.696.8908 l *F*: +972.4.959.1654 l www.communitake.com
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From faheem2084 at gmail.com  Mon Aug  8 17:42:52 2016
From: faheem2084 at gmail.com (Faheem Muhammad)
Date: Mon, 8 Aug 2016 20:42:52 +0500
Subject: [OpenSIPS-Users] Monitoring end to end solution.
In-Reply-To: <CA++nn_y5eacfpPBtE9o1SejAgq-MKnLJJNtjd_i+pC7hyLTH-w@mail.gmail.com>
References: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
	<CAEVWGm9xU+RN3XuAKwenLPFatekUqFiTnoG7M9rOzE9y-j4H1Q@mail.gmail.com>
	<00ab01d1f164$55a7a9d0$00f6fd70$@communitake.com>
	<CA++nn_y5eacfpPBtE9o1SejAgq-MKnLJJNtjd_i+pC7hyLTH-w@mail.gmail.com>
Message-ID: <CA++nn_xJe+H-jF3v+GMyraJoJX=M7tdyRqXxERXChWrtww8x+A@mail.gmail.com>

By the way,
You can do the same with NetSIP, a Perl library. This library can do the
same as SIPp, but it is very simple.
Try this below link.
http://web.archive.org/web/20130924023603/http://techvoiper.com/perl-netsip-sip-invite-example/

Regards,
Muhammad Faheem

On Mon, Aug 8, 2016 at 8:40 PM, Faheem Muhammad <faheem2084 at gmail.com>
wrote:

> No, with SIPp you can create any type of call scenario including REGISTER
> + INVITE scenarios.
> Follow the below mentioned blog entry 'Scenario 2'.
> I posted it few years back. Unfortunately original domain is no more
> available, but archive link works.
>
> http://web.archive.org/web/20130924010913/http://techvoiper.com/sipp-load-
> scenario-generation-overview/
>
> Regards,
> Muhammad Faheem
>
> On Mon, Aug 8, 2016 at 4:02 PM, Ziv Gabel <ziv at communitake.com> wrote:
>
>> The problem with sipp is that it does not support 2 call-ids in the same
>> script (register and invite).
>>
>>
>>
>> Is it possible to have static registrations in opensips ?
>>
>> I can then use one script for registration test and one for the invite.
>>
>>
>>
>> *From:* users-bounces at lists.opensips.org [mailto:users-bounces at lists.op
>> ensips.org] *On Behalf Of *Johan De Clercq
>> *Sent:* ??? ? 08 ?????? 2016 13:54
>> *To:* OpenSIPS users mailling list
>> *Subject:* Re: [OpenSIPS-Users] Monitoring end to end solution.
>>
>>
>>
>> Maybe you can script everything out in sipp ?
>>
>>
>>
>> 2016-08-08 11:58 GMT+02:00 Ziv Gabel <ziv at communitake.com>:
>>
>> Hi,
>>
>> I?m running several opensips servers and I want to monitor the health of
>> the servers.
>>
>> I can check for service status and an open port, but what I?m looking for
>> is a way to make an ?end to end test? that will include a registration, and
>> a call.
>>
>> The test should check that
>>
>> 1.       Registration is working.
>>
>> 2.       Call goes through
>>
>> 3.       Rtp (I use rtpengine) is flowing currectly.
>>
>>
>>
>> Is there any known way to do that or should I figure out I way myself.
>>
>>
>>
>> *Ziv Gabel *l Professional services l CommuniTake Technologies Ltd.
>>
>>
>>
>> *M*: +972535265553 l *Skype*: ziv_gabel l *E*: ziv at communitake.com
>>
>> *T*: +972.4.696.8908 l *F*: +972.4.959.1654 l www.communitake.com
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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From ziv at communitake.com  Mon Aug  8 17:52:49 2016
From: ziv at communitake.com (Ziv Gabel)
Date: Mon, 8 Aug 2016 18:52:49 +0300
Subject: [OpenSIPS-Users] Monitoring end to end solution.
In-Reply-To: <CA++nn_xJe+H-jF3v+GMyraJoJX=M7tdyRqXxERXChWrtww8x+A@mail.gmail.com>
References: <007a01d1f15b$7e612a30$7b237e90$@communitake.com>
	<CAEVWGm9xU+RN3XuAKwenLPFatekUqFiTnoG7M9rOzE9y-j4H1Q@mail.gmail.com>
	<00ab01d1f164$55a7a9d0$00f6fd70$@communitake.com>
	<CA++nn_y5eacfpPBtE9o1SejAgq-MKnLJJNtjd_i+pC7hyLTH-w@mail.gmail.com>
	<CA++nn_xJe+H-jF3v+GMyraJoJX=M7tdyRqXxERXChWrtww8x+A@mail.gmail.com>
Message-ID: <00e601d1f18c$eaac2240$c00466c0$@communitake.com>

10x,

Looks interesting.

I?ll test it and get back with the results

 

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Faheem Muhammad
Sent: ??? ? 08 ?????? 2016 18:43
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Monitoring end to end solution.

 

By the way,

You can do the same with NetSIP, a Perl library. This library can do the same as SIPp, but it is very simple.

Try this below link.

http://web.archive.org/web/20130924023603/http://techvoiper.com/perl-netsip-sip-invite-example/ <http://web.archive.org/web/20130924023603/http:/techvoiper.com/perl-netsip-sip-invite-example/> 

 

Regards,

Muhammad Faheem

 

On Mon, Aug 8, 2016 at 8:40 PM, Faheem Muhammad <faheem2084 at gmail.com> wrote:

No, with SIPp you can create any type of call scenario including REGISTER + INVITE scenarios.

Follow the below mentioned blog entry 'Scenario 2'.

I posted it few years back. Unfortunately original domain is no more available, but archive link works.

 

http://web.archive.org/web/20130924010913/http://techvoiper.com/sipp-load-scenario-generation-overview/ <http://web.archive.org/web/20130924010913/http:/techvoiper.com/sipp-load-scenario-generation-overview/> 

 

Regards,

Muhammad Faheem

 

On Mon, Aug 8, 2016 at 4:02 PM, Ziv Gabel <ziv at communitake.com> wrote:

The problem with sipp is that it does not support 2 call-ids in the same script (register and invite).

 

Is it possible to have static registrations in opensips ?

I can then use one script for registration test and one for the invite.

 

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Johan De Clercq
Sent: ??? ? 08 ?????? 2016 13:54
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Monitoring end to end solution.

 

Maybe you can script everything out in sipp ?

 

2016-08-08 11:58 GMT+02:00 Ziv Gabel <ziv at communitake.com>:

Hi,

I?m running several opensips servers and I want to monitor the health of the servers.

I can check for service status and an open port, but what I?m looking for is a way to make an ?end to end test? that will include a registration, and a call.

The test should check that

1.       Registration is working.

2.       Call goes through 

3.       Rtp (I use rtpengine) is flowing currectly.

 

Is there any known way to do that or should I figure out I way myself.

 

Ziv Gabel l Professional services l CommuniTake Technologies Ltd.

 

M: +972535265553 <tel:%2B972535265553>  l Skype: ziv_gabel l E: ziv at communitake.com

T: +972.4.696.8908 <tel:%2B972.4.696.8908>  l F: +972.4.959.1654 <tel:%2B972.4.959.1654>  l  <http://www.communitake.com/> www.communitake.com

 


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

 


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

 

 

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From rrobson at greenlightcrm.com  Tue Aug  9 12:50:22 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Tue, 9 Aug 2016 11:50:22 +0100
Subject: [OpenSIPS-Users] "gw_attrs_avp" not found in module drouting
Message-ID: <4bbffc93-6116-c1e3-5496-203d5ea8a9d6@greenlightcrm.com>

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From liviu at opensips.org  Tue Aug  9 13:22:49 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 9 Aug 2016 06:22:49 -0500
Subject: [OpenSIPS-Users] "gw_attrs_avp" not found in module drouting
In-Reply-To: <4bbffc93-6116-c1e3-5496-203d5ea8a9d6@greenlightcrm.com>
References: <4bbffc93-6116-c1e3-5496-203d5ea8a9d6@greenlightcrm.com>
Message-ID: <3db2750b-d55b-7cfa-64df-d57436e62a87@opensips.org>

Hi Richard,

The "gw_attrs_avp" was removed a while back ago, and replaced with an 
output parameter to do_routing() and friends.

Thank you for reporting! I've fixed the docs where needed.

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 09.08.2016 05:50, Richard Robson wrote:
>
> Hi,
>
>
> I'm trying to keep a channel count per trunk and want to keep the 
> value in the DB assigned to the particular trunk. I thought that I 
> could use the
> |gw_attrs_avp parameted in the DRouting module (which is doing the 
> routing).
> |
>
> |
> |
>
> |I've added this line to the module config ,|
>
> |modparam("drouting", "gw_attrs_avp", "$avp(trunkcalllimit)")
> |
>
> |and i'm storing the call limit in the attrs field of the dr_gateways 
> table
> |
>
> |but i'm getting this error in the logs:|
>
> |
> |
>
> |Aug  9 11:08:37 gl-sip-03 opensips: ERROR:core:set_mod_param_regex: 
> parameter <gw_attrs_avp> not found in module <drouting>
> Aug  9 11:08:37 gl-sip-03 opensips: CRITICAL:core:yyerror: parse error 
> in config file /etc/opensips/opensips.cfg, line 117, column 20-21: 
> Parameter <gw_attrs_avp> not found in module <drouting> - can't set
> Aug  9 11:08:37 gl-sip-03 opensips: ERROR:core:main: bad config file 
> (1 errors)
> Aug  9 11:08:37 gl-sip-03 opensips: NOTICE:core:main: Exiting....
> |
>
> |
> |
>
> |Has something changed or has the documentation not been updated.
> |
>
> |
> |
>
> |It's version 2.2.1|
>
> |
> |
>
> |Regards,|
>
> |
> |
>
> |Richard
> |
>
>
> -- 
> Richard Robson
> Greenlight Support
> 01382 843843
> support at greenlightcrm.com
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Ben.Newlin at inin.com  Fri Aug  5 22:00:20 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Fri, 5 Aug 2016 20:00:20 +0000
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <c929d518-9bb8-63c7-25d0-260b6989d9d1@opensips.org>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
	<204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
	<579B5127.7080805@opensips.org>
	<273E1DB2-DD7E-412E-B762-2E1826B2AC7D@inin.com>
	<8fa14a30-da35-3a54-257e-390ce097975e@opensips.org>
	<6CE77C34-7F1B-459F-8BDF-2C6ABDA56BC6@inin.com>
	<c929d518-9bb8-63c7-25d0-260b6989d9d1@opensips.org>
Message-ID: <9B9C770C-E693-419B-B472-389474282671@inin.com>

Bogdan,

I tried to reproduce this with some simple instructions, but it didn?t reproduce. There must be some dependency on other functions or configurations we are using. It would be too hard to try to figure out exactly what that is, so I will have to capture the data for you. It will just take a while for me to revive an older configuration that allows me to enable the memory debugger.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Tuesday, August 2, 2016 at 3:47 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Ben,

To make it easier, please send me the instructions on how to reproduce the crash.

Thanks and Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 01.08.2016 20:17, Newlin, Ben wrote:
Bogdan,

I am not familiar with gdb, so I double check what you?ve assessed. If there are some other steps with gdb you would like me to perform, just let me know what to do.

Is there a way to compile the memory debugger without using the interactive `make menuconfig` command? Our build system is completely automated, so it impossible for me to do it this way. Can I pass the options as build parameters or alter the makefile in some way?

I can provide a SIPp scenario which should reproduce the issue on any basic script that uses Dialog with topology hiding, if that would be easier.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Monday, August 1, 2016 at 10:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

According to the BT, the crash is in a pkg_malloc() call:
                    route = pkg_malloc(size);
Please double check this with gdb info.

If so, this indicate a memory corruption and we have 2 options here:
    - you compile with memory debugger (see my previous emails)
    - provide step-by-step indications on how to reproduce this crash.

Thanks and Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 29.07.2016 15:54, Newlin, Ben wrote:
This is 1.11.6, running on CentOS 7.


Ben Newlin

From: <users-bounces at lists.opensips.org><mailto:users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Date: Friday, July 29, 2016 at 8:50 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Ben,

What OpenSIPS version is this (the crashing one) ? 1.11 or 2.1 ?

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 19:02, Newlin, Ben wrote:
I have identified that these crashes are occurring when the far end system is not returning the Record-Route headers in the 200 OK response. The headers are present in the 180 response, but not the 200 OK. I have reproduced the scenario using SIPp and captured a SIP trace: http://pastebin.com/ckKk3EhY

The crash occurs on receipt of the ACK request and attempt to match the dialog.

I also captured a BT for this scenario as well, in case anything specific in the trace made the issue easier to find: http://pastebin.com/cM3FhPiw

I am working with the other system to try to fix their behavior.

Ideally the Record-Route headers from previous replies could be used in this case to allow the call to succeed, but I don?t know if that is possible.

Thanks,

Ben Newlin

From: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Date: Wednesday, July 27, 2016 at 9:44 AM
To: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Bogdan,

This is a different scenario than the other you responded to. As I said, we have two types of servers that work together. One is a load-balancer and runs as a proxy. It uses double Record-Route because it sends messages between public and private networks. Then we have our other servers using TH which receive those requests. We are not using TH and RR on the same server (although I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should not be called when using TH, I believe the documentation should reference that. It states that match_dialog() must be used with TH, but does not indicate that the other functions should not be used or that the functionality won?t work. There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), validate_dialog(), and fix_route_dialog() from my script. The crash still occurred and the BT still pointed to fix_route_dialog() function. So it must be getting called from within Dialog module somewhere. That BT is here: http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP concepts that overlaps. You either act as an end-point (by doing TH), either as a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and repair the routing information in the request (like Route and Contact headers). if you do TH, this routing info is actually hidden and added by OpenSIPS, so there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,






Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: <b0i2> (4) / <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)

Many times the ?URI? displayed in the error message is actually internal OpenSIPS variables, as in the last error above. When they are from the SIP message, I have verified that the messages themselves are correctly formatted. This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, specifically the ?double_rr? mechanism for when multiple interfaces exist. The Record-Routing is occurring on different servers which have not crashed. Only the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the topology_hiding() functionality of the Dialog module. Are validate_dialog() and fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
    setflag(SEQ_REQUEST);

  if (has_totag())
  {
    loose_route();

    if (match_dialog())
    {
      if (!validate_dialog())
        fix_route_dialog();

      if (is_method("BYE"))
        setflag(ACC_FLAG);

      setflag(SEQ_REQUEST);
    }
    else if (!isflagset(SEQ_REQUEST))
    {
      if (!is_method("ACK")) {
        route(rlog, LV_ERROR, "check_sequential", "Sequential request not matched");
        route(reply_error, "481", "Call Does Not Exist");
      }

      return(EXIT);
    }
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin








_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users







_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From bogdan at opensips.org  Tue Aug  9 16:18:36 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 09 Aug 2016 09:18:36 -0500
Subject: [OpenSIPS-Users] t_replicate and dispatcher module to forward
 REGISTER messages
Message-ID: <1krpq2rchbm7rjpm0f4t3nx5.1470752316158@email.android.com>

Thank you Jonathan!
Regards,
Sent from my Samsung Galaxy smartphone.
-------- Original message --------From: Jonathan Hunter <hunterj91 at hotmail.com> Date: 8/8/16  04:43  (GMT-06:00) To: Bogdan-Andrei Iancu <bogdan at opensips.org>, OpenSIPS users mailling list <users at lists.opensips.org> Subject: RE: [OpenSIPS-Users] t_replicate and dispatcher module to forward REGISTER messages 
Issue opened on github, thanks Bogdan.
#943.
Jon

Subject: Re: [OpenSIPS-Users] t_replicate and dispatcher module to forward REGISTER messages
To: users at lists.opensips.org; hunterj91 at hotmail.com
From: bogdan at opensips.org
Date: Sun, 7 Aug 2016 20:39:59 +0300


  
    
  
  
    Hi Jonathan,

      

      Unfortunately this is a design problem - the t_replicate() is
      actually a t_reply() but without passing the received replies back
      to the caller. So t_replicate() behaves on parallel forking
      exactly as t_reply().

      

      Some additional work may be required to change this and have
      t_replicate() properly replicating to all destination. Could you
      please open a ticket on github about this ?

      

      Thanks and Regards,

    
    Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
    On 03.08.2016 19:15, Jonathan Hunter
      wrote:

    
    
      <!--
.ExternalClass .ecxhmmessage P {
padding:0px;
}

.ExternalClass body.ecxhmmessage {
font-size:12pt;
font-family:Calibri;
}

-->
      
        Hi Guys,
        

        
        We have an old platform we are looking to redesign but in
          the meantime we need to fix a problem we have.
        

        
        Currently we have an opensips proxy/registrar with 2 x
          Asterisk boxes behind it.
        

        
        Now currently we use t_replicate and the dispatcher module
          (ds_next_domain()) to send REGISTER requests in parallel to
          both Asterisk Servers so they can update their sip peers.
        

        
        This works great but we sometimes (an issue we are
          investigating) dont always get a 200ok response back from one
          or both Asterisk servers for the REGISTER message.
        

        
        Therefore I am looking to resend the REGISTER whenever this
          may happen.
        

        
        This is fine using the fr timers should both Asterisk
          Servers not respond, however if only 1 Server doesnt respond,
          and the other does, this problem is lost, due to the parallel
          forking being applied to the t_replicate scenario.
        

        
        I can capture 200 ok messages in onreply routes to confirm
          which Asterisk servers have responded, however I cannot
          t_relay again from this type of route.
        

        
        Is there another way to do this? Whereby I can remain
          transaction stateful, and resend a REGISTER message if only
          one of the Asterisk Servers doesnt respond to a REGISTER
          message?
        

        
        Thanks
        

        
        Jon
      
      

      
      

      _______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

    
    
 		 	   		  
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From rrobson at greenlightcrm.com  Tue Aug  9 17:26:19 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Tue, 9 Aug 2016 16:26:19 +0100
Subject: [OpenSIPS-Users] "gw_attrs_avp" not found in module drouting
In-Reply-To: <3db2750b-d55b-7cfa-64df-d57436e62a87@opensips.org>
References: <4bbffc93-6116-c1e3-5496-203d5ea8a9d6@greenlightcrm.com>
	<3db2750b-d55b-7cfa-64df-d57436e62a87@opensips.org>
Message-ID: <fd751f1e-c55d-9731-6dad-dfbfd1c18c3d@greenlightcrm.com>

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From bogdan at opensips.org  Tue Aug  9 17:29:19 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 9 Aug 2016 18:29:19 +0300
Subject: [OpenSIPS-Users] "gw_attrs_avp" not found in module drouting
In-Reply-To: <fd751f1e-c55d-9731-6dad-dfbfd1c18c3d@greenlightcrm.com>
References: <4bbffc93-6116-c1e3-5496-203d5ea8a9d6@greenlightcrm.com>
	<3db2750b-d55b-7cfa-64df-d57436e62a87@opensips.org>
	<fd751f1e-c55d-9731-6dad-dfbfd1c18c3d@greenlightcrm.com>
Message-ID: <b4d09246-cc9e-372d-c204-94ce277881f4@opensips.org>

Hi Richard,

Still there, in the auth module:
http://www.opensips.org/html/docs/modules/2.2.x/auth.html#id294332

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 09.08.2016 18:26, Richard Robson wrote:
> is pv_proxy_authorize deprecated too?
>
>  unknown command <pv_proxy_authorize>, missing loadmodule?
>
> auth module is loaded
>
> Regards,
>
> Richard*
>
> *
> On 09/08/2016 12:22, Liviu Chircu wrote:
>>
>> Hi Richard,
>>
>> The "gw_attrs_avp" was removed a while back ago, and replaced with an 
>> output parameter to do_routing() and friends.
>>
>> Thank you for reporting! I've fixed the docs where needed.
>>
>> Liviu Chircu
>> OpenSIPS Developer
>> http://www.opensips-solutions.com
>> On 09.08.2016 05:50, Richard Robson wrote:
>>>
>>> Hi,
>>>
>>>
>>> I'm trying to keep a channel count per trunk and want to keep the 
>>> value in the DB assigned to the particular trunk. I thought that I 
>>> could use the
>>> |gw_attrs_avp parameted in the DRouting module (which is doing the 
>>> routing).
>>> |
>>>
>>> |
>>> |
>>>
>>> |I've added this line to the module config ,|
>>>
>>> |modparam("drouting", "gw_attrs_avp", "$avp(trunkcalllimit)")
>>> |
>>>
>>> |and i'm storing the call limit in the attrs field of the 
>>> dr_gateways table
>>> |
>>>
>>> |but i'm getting this error in the logs:|
>>>
>>> |
>>> |
>>>
>>> |Aug  9 11:08:37 gl-sip-03 opensips: ERROR:core:set_mod_param_regex: 
>>> parameter <gw_attrs_avp> not found in module <drouting>
>>> Aug  9 11:08:37 gl-sip-03 opensips: CRITICAL:core:yyerror: parse 
>>> error in config file /etc/opensips/opensips.cfg, line 117, column 
>>> 20-21: Parameter <gw_attrs_avp> not found in module <drouting> - 
>>> can't set
>>> Aug  9 11:08:37 gl-sip-03 opensips: ERROR:core:main: bad config file 
>>> (1 errors)
>>> Aug  9 11:08:37 gl-sip-03 opensips: NOTICE:core:main: Exiting....
>>> |
>>>
>>> |
>>> |
>>>
>>> |Has something changed or has the documentation not been updated.
>>> |
>>>
>>> |
>>> |
>>>
>>> |It's version 2.2.1|
>>>
>>> |
>>> |
>>>
>>> |Regards,|
>>>
>>> |
>>> |
>>>
>>> |Richard
>>> |
>>>
>>>
>>> -- 
>>> Richard Robson
>>> Greenlight Support
>>> 01382 843843
>>> support at greenlightcrm.com
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> -- 
> Richard Robson
> Greenlight Support
> 01382 843843
> support at greenlightcrm.com
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From rrobson at greenlightcrm.com  Tue Aug  9 17:40:58 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Tue, 9 Aug 2016 16:40:58 +0100
Subject: [OpenSIPS-Users] "gw_attrs_avp" not found in module drouting
In-Reply-To: <b4d09246-cc9e-372d-c204-94ce277881f4@opensips.org>
References: <4bbffc93-6116-c1e3-5496-203d5ea8a9d6@greenlightcrm.com>
	<3db2750b-d55b-7cfa-64df-d57436e62a87@opensips.org>
	<fd751f1e-c55d-9731-6dad-dfbfd1c18c3d@greenlightcrm.com>
	<b4d09246-cc9e-372d-c204-94ce277881f4@opensips.org>
Message-ID: <d8b4e5a5-b0dd-684a-c134-790099af05a6@greenlightcrm.com>

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From Agalya_Ramachandran at comcast.com  Tue Aug  9 21:36:43 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 9 Aug 2016 19:36:43 +0000
Subject: [OpenSIPS-Users] Issue in opensips 2.1.4
Message-ID: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>

Hi team,

I am trying to use opensips 2.1.4 .
I have made some changes in opensips.cfg file for sip trace module, and opensips does not start after that change.
The same change for sip trace module is working good in opensips 1.11.3.

Error am getting is:

Aug  9 19:30:21: WARNING:core:solve_module_dependencies: module siptrace depends on an sqldb module, but none was loaded!
Aug  9 19:30:21 :ERROR:core:main: failed to solve module dependencies

Below is the newly added stuff for sip trace module:
loadmodule "siptrace.so"

modparam("siptrace", "db_url","mysql://opensips:opensipsrw at localhost/opensips")
modparam("siptrace", "duplicate_uri", "sip:100.10.10.1:9060")
modparam("siptrace", "duplicate_with_hep", 1)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "traced_user_avp", "$avp(traced_user)")

Please let me know if am missing something.

Regards,
Agalya
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From Agalya_Ramachandran at comcast.com  Tue Aug  9 21:47:20 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 9 Aug 2016 19:47:20 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
In-Reply-To: <d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
Message-ID: <8620c7c900ef4873a2fce16b469b1641@COPDCEX28.cable.comcast.com>

Hi Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c

If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

But I never see rest_put() is called. Rather am getting below error continuously.
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution

Am I missing something here? I am sure it is a small thing. Do we need to do anything else before calling async in the script?

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.


Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From adrian.fretwell at topgreen.co.uk  Tue Aug  9 21:48:52 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Tue, 9 Aug 2016 20:48:52 +0100
Subject: [OpenSIPS-Users] Issue in opensips 2.1.4
In-Reply-To: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
References: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
Message-ID: <726824f4-1eb6-c248-5c05-a6cf058ea258@topgreen.co.uk>

You need a database module, as your first module parameter "db_url" 
specifies mysql you should probably load the mysql module.

http://www.opensips.org/html/docs/modules/2.1.x/db_mysql.html

Kind regards,

Adrian Fretwell

On 09/08/16 20:36, Ramachandran, Agalya (Contractor) wrote:
>
> Hi team,
>
> I am trying to use opensips 2.1.4 .
>
> I have made some changes in opensips.cfg file for sip trace module, 
> and opensips does not start after that change.
>
> The same change for sip trace module is working good in opensips *1.11.3.*
>
> Error am getting is:
>
> Aug  9 19:30:21: WARNING:core:solve_module_dependencies: module 
> siptrace depends on an sqldb module, but none was loaded!
>
> Aug  9 19:30:21 :ERROR:core:main: failed to solve module dependencies
>
> Below is the newly added stuff for sip trace module:
>
> loadmodule "siptrace.so"
>
> modparam("siptrace", 
> "db_url","mysql://opensips:opensipsrw at localhost/opensips")
>
> modparam("siptrace", "duplicate_uri", "sip:100.10.10.1:9060")
>
> modparam("siptrace", "duplicate_with_hep", 1)
>
> modparam("siptrace", "trace_to_database", 0)
>
> modparam("siptrace", "trace_on", 1)
>
> modparam("siptrace", "hep_version", 2)
>
> modparam("siptrace", "traced_user_avp", "$avp(traced_user)")
>
> Please let me know if am missing something.
>
> Regards,
>
> Agalya
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From nabeelshikder at gmail.com  Wed Aug 10 10:24:57 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Wed, 10 Aug 2016 09:24:57 +0100
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
In-Reply-To: <CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>
References: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
	<698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>
	<CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>
Message-ID: <CA+vx6K+hzs0_Jut1YooX8qPugDMFdEn7iAp6okLKb_uyXV008g@mail.gmail.com>

Is there any way to make OpenSIPS handle a call with voicemail in exactly
the same way as a call with another SIP user?
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From rrobson at greenlightcrm.com  Wed Aug 10 13:05:01 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Wed, 10 Aug 2016 12:05:01 +0100
Subject: [OpenSIPS-Users] problem with $avp pulled from local db cache
Message-ID: <f2ff03de-2d00-1993-a391-9eac648382d6@greenlightcrm.com>

Hi All,

I'm using  the local dbcache to record an accountcode password and two 
variables. these are used for authentoiaction and rate and channe 
limiting the customer.


the account code is a 4 digit number e.g. 0001

this authenticates correctly in the auth_db and auth modules. (its 
chached after the first attempt)

once authenticated the accountcode is uses as a ratelimit pipe, which is 
working fine after the auth_db, when  the values are cached in the local 
cache.

on a subsequent call the cached credentials are there and are pulled 
from the cache and the pv_proxy_auth passes. this is then ratelimited 
(with the same code block) but here we get an error.

rl_check("$avp(accountcode)", "$avp(customerratelimit)", "TAILDROP")

ERROR:core:fixup_get_ivalue: no valid PV value found (error in scripts)
ERROR:ratelimit:w_rl_check_3: cannot retrieve limit

I suspect that the avp that is being used is casting the value somehow 
ans the ratelimit is not recognising the value, but i'm logging the avp 
and it show 0001 in both cases.

[relay_customer] Account code: 0001 -  Channel Limit: 20 - ratelimit: 10

xlog("[relay_customer] Account code: $avp(accountcode) -  Channel Limit: 
$avp(customerchannellimit) -  ratelimit: $avp(customerratelimit)");

this is the fetch:

cache_fetch("local","username_$si",$avp(accountcode)

this is the  store:

cache_store("local","username_$si","$avp(accountcode)",1200)

here the avp is in quotes in one and not the other, which may be the 
problem but the script fails when the quotes are added to the fetch or 
removed from the store.


version 2.2.1



-- 
Richard Robson
Greenlight Support
01382 843843
support at greenlightcrm.com



From chandan.pr at webshar.org  Wed Aug 10 14:16:39 2016
From: chandan.pr at webshar.org (Chandan PR)
Date: Wed, 10 Aug 2016 17:46:39 +0530
Subject: [OpenSIPS-Users] OpenSips rejecting too many calls
Message-ID: <CADh=FyiKZRy-EqdiL2nkXSHmUY+gvytnVMd9C8rs8ETYVmTGNw@mail.gmail.com>

Hi Guys,

We are using OpenSips, 1.9 Load Balancer with Probing Interval set to 15
seconds.

Yesterday we observed huge number of calls being rejected by Open Sips as
Destinations were full.

But the number of error codes which results in disabling the gateway was
pretty less on the gateway end.

The log file was not rolled over and it was of size ~2.5G.

Any one has faced similar issue?

I see a new event "E_LOAD_BALANCER_STATUS" available in latest versions,
which helps in knowing when the destination was disabled or enabled.

Any other possible options in 1.9.x to know the details about when the
destination was disabled or enabled?

How easy or difficult it is to migrate from 1.9.x to 2.2?

Any migration tutorials/links/docs for the same?

Thanks,
Chandan
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From daniel.zanutti at gmail.com  Wed Aug 10 14:56:52 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Wed, 10 Aug 2016 09:56:52 -0300
Subject: [OpenSIPS-Users] OpenSips rejecting too many calls
In-Reply-To: <CADh=FyiKZRy-EqdiL2nkXSHmUY+gvytnVMd9C8rs8ETYVmTGNw@mail.gmail.com>
References: <CADh=FyiKZRy-EqdiL2nkXSHmUY+gvytnVMd9C8rs8ETYVmTGNw@mail.gmail.com>
Message-ID: <CAN0jgiN09j08jTx-ABoGkr=QDg6TT21zaHTfuFH+ezyCrj6ViA@mail.gmail.com>

Hi Chandan

I would suggest upgrading to 1.11.X first, it will be very straightforward.
If you still have problems, migrating to 2.2 will probably not solve them
and you may find new ones.

Regards

On Wed, Aug 10, 2016 at 9:16 AM, Chandan PR <chandan.pr at webshar.org> wrote:

> Hi Guys,
>
> We are using OpenSips, 1.9 Load Balancer with Probing Interval set to 15
> seconds.
>
> Yesterday we observed huge number of calls being rejected by Open Sips as
> Destinations were full.
>
> But the number of error codes which results in disabling the gateway was
> pretty less on the gateway end.
>
> The log file was not rolled over and it was of size ~2.5G.
>
> Any one has faced similar issue?
>
> I see a new event "E_LOAD_BALANCER_STATUS" available in latest versions,
> which helps in knowing when the destination was disabled or enabled.
>
> Any other possible options in 1.9.x to know the details about when the
> destination was disabled or enabled?
>
> How easy or difficult it is to migrate from 1.9.x to 2.2?
>
> Any migration tutorials/links/docs for the same?
>
> Thanks,
> Chandan
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From bogdan at opensips.org  Wed Aug 10 15:18:12 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 10 Aug 2016 16:18:12 +0300
Subject: [OpenSIPS-Users] Issue in opensips 2.1.4
In-Reply-To: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
References: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
Message-ID: <a7d5a89e-96bc-976d-c54e-6ad93ae768b9@opensips.org>

Hi Agalya,

Have you loaded the db_mysql module before the siptrace module ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 09.08.2016 22:36, Ramachandran, Agalya (Contractor) wrote:
>
> Hi team,
>
> I am trying to use opensips 2.1.4 .
>
> I have made some changes in opensips.cfg file for sip trace module, 
> and opensips does not start after that change.
>
> The same change for sip trace module is working good in opensips *1.11.3.*
>
> Error am getting is:
>
> Aug  9 19:30:21: WARNING:core:solve_module_dependencies: module 
> siptrace depends on an sqldb module, but none was loaded!
>
> Aug  9 19:30:21 :ERROR:core:main: failed to solve module dependencies
>
> Below is the newly added stuff for sip trace module:
>
> loadmodule "siptrace.so"
>
> modparam("siptrace", 
> "db_url","mysql://opensips:opensipsrw at localhost/opensips")
>
> modparam("siptrace", "duplicate_uri", "sip:100.10.10.1:9060")
>
> modparam("siptrace", "duplicate_with_hep", 1)
>
> modparam("siptrace", "trace_to_database", 0)
>
> modparam("siptrace", "trace_on", 1)
>
> modparam("siptrace", "hep_version", 2)
>
> modparam("siptrace", "traced_user_avp", "$avp(traced_user)")
>
> Please let me know if am missing something.
>
> Regards,
>
> Agalya
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Agalya_Ramachandran at comcast.com  Wed Aug 10 15:21:39 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 10 Aug 2016 13:21:39 +0000
Subject: [OpenSIPS-Users] Issue in opensips 2.1.4
In-Reply-To: <a7d5a89e-96bc-976d-c54e-6ad93ae768b9@opensips.org>
References: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
	<a7d5a89e-96bc-976d-c54e-6ad93ae768b9@opensips.org>
Message-ID: <3283e221053b486ea5ed845d1e62b2a9@COPDCEX28.cable.comcast.com>

Hi Bogdan,

I didn't load db_msql module in opensips 1.11. But it worked for me.
The same config file when am trying to use in 2.1.4 it says me the error.

Now I have tried including the db_msql module and its good now.
I thought it will work the same way in 1.11 and 2.1.4.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, August 10, 2016 9:18 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Issue in opensips 2.1.4

Hi Agalya,

Have you loaded the db_mysql module before the siptrace module ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 09.08.2016 22:36, Ramachandran, Agalya (Contractor) wrote:
Hi team,

I am trying to use opensips 2.1.4 .
I have made some changes in opensips.cfg file for sip trace module, and opensips does not start after that change.
The same change for sip trace module is working good in opensips 1.11.3.

Error am getting is:

Aug  9 19:30:21: WARNING:core:solve_module_dependencies: module siptrace depends on an sqldb module, but none was loaded!
Aug  9 19:30:21 :ERROR:core:main: failed to solve module dependencies

Below is the newly added stuff for sip trace module:
loadmodule "siptrace.so"

modparam("siptrace", "db_url","mysql://opensips:opensipsrw at localhost/opensips")
modparam("siptrace", "duplicate_uri", "sip:100.10.10.1:9060")
modparam("siptrace", "duplicate_with_hep", 1)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "traced_user_avp", "$avp(traced_user)")

Please let me know if am missing something.

Regards,
Agalya




_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From rrobson at greenlightcrm.com  Wed Aug 10 18:07:51 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Wed, 10 Aug 2016 17:07:51 +0100
Subject: [OpenSIPS-Users] problem with $avp pulled from local db cache
In-Reply-To: <f2ff03de-2d00-1993-a391-9eac648382d6@greenlightcrm.com>
References: <f2ff03de-2d00-1993-a391-9eac648382d6@greenlightcrm.com>
Message-ID: <03a5cef9-feb8-d49b-cc97-b3dc9b9c5ad7@greenlightcrm.com>

Ok,

I've solved this.

The problem was the dbcache had stored or fetched the 
$avp(customerratelimit) as string and the rl_limit function was 
expecting an int.
If I cast the asp to an int $(avp(customerratelimit){s.int}) it works 
fine with the fetched value.

I had made the assumption that the dbcaching would leave the value with 
the same type.
Regards,

Richard


On 10/08/2016 12:05, Richard Robson wrote:
> Hi All,
>
> I'm using  the local dbcache to record an accountcode password and two 
> variables. these are used for authentoiaction and rate and channe 
> limiting the customer.
>
>
> the account code is a 4 digit number e.g. 0001
>
> this authenticates correctly in the auth_db and auth modules. (its 
> chached after the first attempt)
>
> once authenticated the accountcode is uses as a ratelimit pipe, which 
> is working fine after the auth_db, when  the values are cached in the 
> local cache.
>
> on a subsequent call the cached credentials are there and are pulled 
> from the cache and the pv_proxy_auth passes. this is then ratelimited 
> (with the same code block) but here we get an error.
>
> rl_check("$avp(accountcode)", "$avp(customerratelimit)", "TAILDROP")
>
> ERROR:core:fixup_get_ivalue: no valid PV value found (error in scripts)
> ERROR:ratelimit:w_rl_check_3: cannot retrieve limit
>
> I suspect that the avp that is being used is casting the value somehow 
> ans the ratelimit is not recognising the value, but i'm logging the 
> avp and it show 0001 in both cases.
>
> [relay_customer] Account code: 0001 -  Channel Limit: 20 - ratelimit: 10
>
> xlog("[relay_customer] Account code: $avp(accountcode) -  Channel 
> Limit: $avp(customerchannellimit) -  ratelimit: 
> $avp(customerratelimit)");
>
> this is the fetch:
>
> cache_fetch("local","username_$si",$avp(accountcode)
>
> this is the  store:
>
> cache_store("local","username_$si","$avp(accountcode)",1200)
>
> here the avp is in quotes in one and not the other, which may be the 
> problem but the script fails when the quotes are added to the fetch or 
> removed from the store.
>
>
> version 2.2.1
>
>
>


-- 
Richard Robson
Greenlight Support
01382 843843
support at greenlightcrm.com



From bogdan at opensips.org  Wed Aug 10 20:23:48 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 10 Aug 2016 21:23:48 +0300
Subject: [OpenSIPS-Users] Issue in opensips 2.1.4
In-Reply-To: <3283e221053b486ea5ed845d1e62b2a9@COPDCEX28.cable.comcast.com>
References: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
	<a7d5a89e-96bc-976d-c54e-6ad93ae768b9@opensips.org>
	<3283e221053b486ea5ed845d1e62b2a9@COPDCEX28.cable.comcast.com>
Message-ID: <f5d762fc-4739-8b8d-ed64-39daf9bb4ffb@opensips.org>

Hi,

I see you have the db_url parameter (in siptrace)pointing to the 
db_mysql module, so this is a must to load.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.08.2016 16:21, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> I didn?t load db_msql module in opensips 1.11. But it worked for me.
>
> The same config file when am trying to use in 2.1.4 it says me the error.
>
> Now I have tried including the db_msql module and its good now.
>
> I thought it will work the same way in 1.11 and 2.1.4.
>
> Regards,
>
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Wednesday, August 10, 2016 9:18 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; 
> Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
> *Subject:* Re: [OpenSIPS-Users] Issue in opensips 2.1.4
>
> Hi Agalya,
>
> Have you loaded the db_mysql module before the siptrace module ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 09.08.2016 22:36, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi team,
>
>     I am trying to use opensips 2.1.4 .
>
>     I have made some changes in opensips.cfg file for sip trace
>     module, and opensips does not start after that change.
>
>     The same change for sip trace module is working good in opensips
>     *1.11.3.*
>
>     Error am getting is:
>
>     Aug  9 19:30:21: WARNING:core:solve_module_dependencies: module
>     siptrace depends on an sqldb module, but none was loaded!
>
>     Aug  9 19:30:21 :ERROR:core:main: failed to solve module dependencies
>
>     Below is the newly added stuff for sip trace module:
>
>     loadmodule "siptrace.so"
>
>     modparam("siptrace",
>     "db_url","mysql://opensips:opensipsrw at localhost/opensips")
>
>     modparam("siptrace", "duplicate_uri", "sip:100.10.10.1:9060")
>
>     modparam("siptrace", "duplicate_with_hep", 1)
>
>     modparam("siptrace", "trace_to_database", 0)
>
>     modparam("siptrace", "trace_on", 1)
>
>     modparam("siptrace", "hep_version", 2)
>
>     modparam("siptrace", "traced_user_avp", "$avp(traced_user)")
>
>     Please let me know if am missing something.
>
>     Regards,
>
>     Agalya
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Wed Aug 10 20:38:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 10 Aug 2016 21:38:08 +0300
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
In-Reply-To: <CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>
References: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
	<698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>
	<CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>
Message-ID: <507034b8-575f-b8da-35db-e9b36f2977ec@opensips.org>

Hi Nabeel,

OpenSIPS does not assume anything by default. If you want to have any 
new calls to user A rejected (if A already in a call, with other users 
or any service), you need to script this.
You should use dialog profiles to count the ongoing calls for the user A 
and when you get a new call for A and the profile counter is <> 0, just 
do the reject.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 08.08.2016 01:44, Nabeel wrote:
>
> Hi,
>
> The busy reply should be generated in exactly the same way as when the 
> callee is busy on the phone with another SIP user (from the OpenSIPS 
> subscriber table). The only difference is that the callee is on the 
> phone with asterisk voicemail instead of a SIP user from the 
> subscriber table.  I'm not sure who exactly should generate the reply 
> but I think it depends on how OpenSIPS is interacting with the 
> Voicemail server during a call to voicemail. Perhaps it's not treating 
> the voicemail service as another 'user' so the caller didn't receive 
> the busy signal.
>
> Nabeel
>
>
> On 7 Aug 2016 6:57 p.m., "Bogdan-Andrei Iancu" <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Nabeel,
>
>     Who should generate the 486 Busy reply ?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://www.opensips-solutions.com>
>
>     On 06.08.2016 02:56, Nabeel wrote:
>>     Hi,
>>     OpenSIPS does not receive a '486 Busy' signal when a callee is
>>     using the Asterisk voicemail service. If a user is currently
>>     listening to voicemail or leaving a voice message, an attempt to
>>     call that user should result in a '486 Busy' signal. From that
>>     response, I will be able to play a busy tone on the caller's
>>     side. Please let me know how to fix this.
>>     Nabeel
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
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From nabeelshikder at gmail.com  Wed Aug 10 20:58:22 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Wed, 10 Aug 2016 19:58:22 +0100
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
In-Reply-To: <507034b8-575f-b8da-35db-e9b36f2977ec@opensips.org>
References: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
	<698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>
	<CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>
	<507034b8-575f-b8da-35db-e9b36f2977ec@opensips.org>
Message-ID: <CA+vx6KK6gtzZ4Ts36aOAa4WNOWZzB6A2TYWX7bypto1wZ5gxqg@mail.gmail.com>

On 10 August 2016 at 19:38, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:

> Hi Nabeel,
>
> OpenSIPS does not assume anything by default. If you want to have any new
> calls to user A rejected (if A already in a call, with other users or any
> service), you need to script this.
>


In the case of user A being in a call with other users, I didn't have to
script this in order to receive a 486 busy response.  I don't see this in
the cfg script. This seems to be default OpenSIPS behaviour in the case of
calling other users.
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From Agalya_Ramachandran at comcast.com  Wed Aug 10 21:57:42 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 10 Aug 2016 19:57:42 +0000
Subject: [OpenSIPS-Users] Issue in opensips 2.1.4
In-Reply-To: <f5d762fc-4739-8b8d-ed64-39daf9bb4ffb@opensips.org>
References: <cc52cf2cbb9d48fabea7972c82c10096@COPDCEX28.cable.comcast.com>
	<a7d5a89e-96bc-976d-c54e-6ad93ae768b9@opensips.org>
	<3283e221053b486ea5ed845d1e62b2a9@COPDCEX28.cable.comcast.com>
	<f5d762fc-4739-8b8d-ed64-39daf9bb4ffb@opensips.org>
Message-ID: <1cbdccaf4e6945b2829422b7829cdcab@COPDCEX28.cable.comcast.com>

Hi,

Yes, after loading db_mysql it worked.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, August 10, 2016 2:24 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Issue in opensips 2.1.4

Hi,

I see you have the db_url parameter (in siptrace) pointing to the db_mysql module, so this is a must to load.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 10.08.2016 16:21, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

I didn't load db_msql module in opensips 1.11. But it worked for me.
The same config file when am trying to use in 2.1.4 it says me the error.

Now I have tried including the db_msql module and its good now.
I thought it will work the same way in 1.11 and 2.1.4.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, August 10, 2016 9:18 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Issue in opensips 2.1.4

Hi Agalya,

Have you loaded the db_mysql module before the siptrace module ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 09.08.2016 22:36, Ramachandran, Agalya (Contractor) wrote:
Hi team,

I am trying to use opensips 2.1.4 .
I have made some changes in opensips.cfg file for sip trace module, and opensips does not start after that change.
The same change for sip trace module is working good in opensips 1.11.3.

Error am getting is:

Aug  9 19:30:21: WARNING:core:solve_module_dependencies: module siptrace depends on an sqldb module, but none was loaded!
Aug  9 19:30:21 :ERROR:core:main: failed to solve module dependencies

Below is the newly added stuff for sip trace module:
loadmodule "siptrace.so"

modparam("siptrace", "db_url","mysql://opensips:opensipsrw at localhost/opensips")
modparam("siptrace", "duplicate_uri", "sip:100.10.10.1:9060")
modparam("siptrace", "duplicate_with_hep", 1)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "traced_user_avp", "$avp(traced_user)")

Please let me know if am missing something.

Regards,
Agalya





_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From bogdan at opensips.org  Thu Aug 11 07:00:05 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 11 Aug 2016 08:00:05 +0300
Subject: [OpenSIPS-Users] Problem with basic Opensips installation
In-Reply-To: <57A74527.9010103@alainfaure.net>
References: <57A74527.9010103@alainfaure.net>
Message-ID: <23a6d7b5-81fa-30bd-8393-ae720d478439@opensips.org>

Hi,

The missing modules are not compiled by default by the make file as they 
require external dependency (as libs). So, if you want to use those 
module from the script, before compiling, run "make menuconfig" in order 
to enable the compiling for those modules too.
See: http://www.opensips.org/Documentation/Install-CompileAndInstall-2-2

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.08.2016 17:26, Alain Faure wrote:
> Hi,
>
> I would like to use Opensips.
>
> 1) i use :
> sudo apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B
> sudo echo "deb http://apt.opensips.org xenial 2.2-releases" 
> >>/etc/apt/sources.list
> but that was a fail (pbm with public key)
>
> 2) I try an installation on Ubuntu 16.04 of the Opensips 2.2.1.
> I use the basic server configuration for Ubuntu 16.04. package 
> "Standard system utilities"
>
> cd /usr/src
> sudo wget http://opensips.org/pub/opensips/latest/opensips-2.2.1.tar.gz
> tar -zxvf opensips-2.2.1.tar.gz
> cd opensips-2.2.1
> sudo apt-get install build-essential openssl bison flex
> sudo apt-get install perl libdbi-perl libdbd-mysql-perl libdbd-pg-perl 
> libfrontier-rpc-perl libterm-readline-gnu-perl libberkeleydb-perl 
> ncurses-dev
> sudo apt-get install mysql-server libmysqlclient-dev
> sudo apt install make
> sudo make all
> sudo make install
> sudo mkdir /var/run/opensips
> sudo cd packaging/debian/
> sudo cp opensips.default /etc/default/opensips
> sudo cp opensips.init /etc/init.d/opensips
> sudo chmod +x /etc/init.d/opensips
> sudo useradd opensips
> sudo vim /etc/default/opensips
> RUN_OpenSIPS' to 'Yes'
> shared memory to minimum 128
>
> OpenSIP and change it location in its startup script and update its 
> state from 'off' to 'on'
> sudo vim /etc/init.d/opensips
>
> /etc/init.d/opensips start
> That was a fail.
> he could not find :
> db_mysql.so
> xcap.so
> presence.so
> presence_xml.so
> xcaplpresence.so
> proto_tls.so
>
>
> That's really important for me : 3) As the system is a new one i can i 
> can choose a new one.
> Can you give a working configuration that you really use to test 
> Opensips 2.2.1 ?
> Distrib : Debian or Ubuntu
> Version of the Distrib : 12.04, 14.04, 15.01, 15.02 or 16.04
>
> And, Is it possible for you to give the full procedure you use to 
> setup from scratch-to-run a very basic opensips server ?
> (with the tested choice of a Linux distribution)
> It could help a lot of people i hope.
>
> Thank's a lot for your pateince.
>
> Best regards



From spanda at 3clogic.com  Thu Aug 11 10:00:09 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Thu, 11 Aug 2016 13:30:09 +0530
Subject: [OpenSIPS-Users] mi_xmlrpc module related query .
Message-ID: <CALf-Q47sAcKNZaRSrri9JyYNBJ5jB+o2qOo406P1XAszJpHLpg@mail.gmail.com>

Hi All ,

    I wanted to use this module in opensips-1.11 . And in the doc it given
that
*"libxmlrpc-c3** version 0.9.10 - 0.9.42. (using other versions may be
dangerous and lead to opensips blocking)  *" .

          I am not getting this particular  version anywhere . I have tried
to install "*libxmlrpc-core-c3_1.16.33-3*." version . but while compiling
opensips , its giving warning : " *you are using a different version which
is not recommended so compile in ur own risk "* .
              After installation I am not getting the  " mi_xmlrpc.so "
 file in the module directory .

      Can anybody tell me where will get the same version . Or is there any
other way to fix my problem .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From Agalya_Ramachandran at comcast.com  Thu Aug 11 15:21:55 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 11 Aug 2016 13:21:55 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org> 
Message-ID: <0ac06054cede4b53b96dc146a463efe8@COPDCEX28.cable.comcast.com>

Hi team,

Can anyone help me or point to what mistake am doing when using async() in the script?

Regards,
Agalya

From: Ramachandran, Agalya (Contractor)
Sent: Tuesday, August 09, 2016 3:47 PM
To: users at lists.opensips.org
Subject: RE: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c

If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** <http://***url***%20> ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

But I never see rest_put() is called. Rather am getting below error continuously.
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution

Am I missing something here? I am sure it is a small thing. Do we need to do anything else before calling async in the script?

Regards,
Agalya

From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From adrian.fretwell at topgreen.co.uk  Thu Aug 11 21:57:20 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Thu, 11 Aug 2016 20:57:20 +0100
Subject: [OpenSIPS-Users] Opensips v2.2.1 Dialog reinvite ping
Message-ID: <cdf359d8-dd3f-aa41-b420-f16bb0cb7b02@topgreen.co.uk>

I wonder if any Opensips experts and shed any light on this.

I have just upgraded to Opensips 2.2.1 to take advantage of the new 
dialog reinvite ping, but what I am seeing is every time my Opensips 
proxy sends an invite ping the UAC (tried polycom and Yealink phones) 
responds with a 500 Internal Server Error.  I can't see anything wrong 
with the invite packet that Opensips is sending out.  Any ideas how I 
can diagnose this further?

Kind regards,

Adrian Fretwell
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From Agalya_Ramachandran at comcast.com  Thu Aug 11 22:00:31 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 11 Aug 2016 20:00:31 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org> 
Message-ID: <0d741665da5047d68756f1f31c95e156@COPDCEX28.cable.comcast.com>

Hi,

Am getting the below error if I enable debug logs.
Seems, w_async_rest_put function is called and after that it didn't process the async(rest_put)).
Do we need to configure anything specific when async() is called in config file?

Aug 11 19:49:52 DBG:tm:_reply_light: finished
Aug 11 19:49:52 DBG:rest_client:w_async_rest_put: async rest put 'http://***URL***?' 0x7f3df7848538 0x7f3df78485d0 0x7f3df7848650
Aug 11 19:50:12 DBG:rest_client:start_async_http_req: done, no need for async!
Aug 11 19:50:12 DBG:rest_client:set_output_pv_params: Last response code: 0
Aug 11 19:50:12 DBG:core:fm_free: free(0) called
Aug 11 19:50:12 DBG:core:comp_scriptvar: int 26 : -4 / 0
Aug 11 19:50:12 Return code is -4
Aug 11 19:50:12 Error code 0 in HTTP PUT!

Regards,
Agalya

From: Ramachandran, Agalya (Contractor)
Sent: Tuesday, August 09, 2016 3:47 PM
To: users at lists.opensips.org
Subject: RE: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c

If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** <http://***url***%20> ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

But I never see rest_put() is called. Rather am getting below error continuously.
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution

Am I missing something here? I am sure it is a small thing. Do we need to do anything else before calling async in the script?

Regards,
Agalya

From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From johan at democon.be  Fri Aug 12 07:08:30 2016
From: johan at democon.be (Johan De Clercq)
Date: Fri, 12 Aug 2016 05:08:30 +0000 (UTC)
Subject: [OpenSIPS-Users] Opensips v2.2.1 Dialog reinvite ping
In-Reply-To: <cdf359d8-dd3f-aa41-b420-f16bb0cb7b02@topgreen.co.uk>
References: <cdf359d8-dd3f-aa41-b420-f16bb0cb7b02@topgreen.co.uk>
Message-ID: <5AB35FAD44B8F4E0.0F182BD8-D935-4049-868C-930636D42977@mail.outlook.com>

Do you have a trac? ?

Get Outlook for iOS




On Thu, Aug 11, 2016 at 9:58 PM +0200, "Adrian Fretwell" <adrian.fretwell at topgreen.co.uk> wrote:











  

    
  
  
    

I wonder if any Opensips experts and shed any
        light on this.
    

I have just upgraded to Opensips 2.2.1 to take
        advantage of the new dialog reinvite ping, but what I am seeing
        is every time my Opensips proxy sends an invite ping the UAC
        (tried polycom and Yealink phones) responds with a 500 Internal
        Server Error.? I can't see anything wrong with the invite packet
        that Opensips is sending out.? Any ideas how I can diagnose this
        further?

    
    Kind regards,

    

          Adrian Fretwell

        
    
  






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From adrian.fretwell at topgreen.co.uk  Fri Aug 12 23:08:17 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Fri, 12 Aug 2016 22:08:17 +0100
Subject: [OpenSIPS-Users] Opensips v2.2.1 Dialog reinvite ping
In-Reply-To: <5AB35FAD44B8F4E0.0F182BD8-D935-4049-868C-930636D42977@mail.outlook.com>
References: <cdf359d8-dd3f-aa41-b420-f16bb0cb7b02@topgreen.co.uk>
	<5AB35FAD44B8F4E0.0F182BD8-D935-4049-868C-930636D42977@mail.outlook.com>
Message-ID: <bc250502-038d-e0bc-53da-9fa6c760358a@topgreen.co.uk>

I think I have got to the bottom of this.  There was also an options 
ping going on and the 500 internal server error was generated if the 
options ping arrived before the ack for the 200 ok response to the 
invite ping.

Kind regards,

Adrian Fretwell

On 12/08/16 06:08, Johan De Clercq wrote:
> Do you have a trac? ?
>
> Get Outlook for iOS <https://aka.ms/o0ukef>
>
>
>
>
> On Thu, Aug 11, 2016 at 9:58 PM +0200, "Adrian Fretwell" 
> <adrian.fretwell at topgreen.co.uk 
> <mailto:adrian.fretwell at topgreen.co.uk>> wrote:
>
>     I wonder if any Opensips experts and shed any light on this.
>
>     I have just upgraded to Opensips 2.2.1 to take advantage of the
>     new dialog reinvite ping, but what I am seeing is every time my
>     Opensips proxy sends an invite ping the UAC (tried polycom and
>     Yealink phones) responds with a 500 Internal Server Error.  I
>     can't see anything wrong with the invite packet that Opensips is
>     sending out.  Any ideas how I can diagnose this further?
>
>     Kind regards,
>
>     Adrian Fretwell
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Agalya_Ramachandran at comcast.com  Mon Aug 15 15:31:51 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Mon, 15 Aug 2016 13:31:51 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org> 
Message-ID: <64550c2d1fc944d7b88b5cd2f527db75@COPDCEX28.cable.comcast.com>

Hi Bogdan/Liviu,

Please help on this async() to get worked at config file.
Am using rest_put() and I have added the needed functions and declarations for the same as like rest_post().

Am getting the below error if I enable debug logs.
Seems, w_async_rest_put function is called and after that it didn't process the async(rest_put)).
Do we need to configure anything specific when async() is called in config file?

Aug 11 19:49:52 DBG:tm:_reply_light: finished
Aug 11 19:49:52 DBG:rest_client:w_async_rest_put: async rest put 'http://***URL***?' 0x7f3df7848538 0x7f3df78485d0 0x7f3df7848650
Aug 11 19:50:12 DBG:rest_client:start_async_http_req: done, no need for async!
Aug 11 19:50:12 DBG:rest_client:set_output_pv_params: Last response code: 0
Aug 11 19:50:12 DBG:core:fm_free: free(0) called
Aug 11 19:50:12 DBG:core:comp_scriptvar: int 26 : -4 / 0
Aug 11 19:50:12 Return code is -4
Aug 11 19:50:12 Error code 0 in HTTP PUT!

Regards,
Agalya


From: Ramachandran, Agalya (Contractor)
Sent: Tuesday, August 09, 2016 3:47 PM
To: users at lists.opensips.org
Subject: RE: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c

If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** <http://***url***%20> ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

But I never see rest_put() is called. Rather am getting below error continuously.
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution

Am I missing something here? I am sure it is a small thing. Do we need to do anything else before calling async in the script?

Regards,
Agalya

From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From chandan.pr at webshar.org  Mon Aug 15 20:34:35 2016
From: chandan.pr at webshar.org (Chandan PR)
Date: Tue, 16 Aug 2016 00:04:35 +0530
Subject: [OpenSIPS-Users] OpenSips rejecting too many calls
In-Reply-To: <CAN0jgiN09j08jTx-ABoGkr=QDg6TT21zaHTfuFH+ezyCrj6ViA@mail.gmail.com>
References: <CADh=FyiKZRy-EqdiL2nkXSHmUY+gvytnVMd9C8rs8ETYVmTGNw@mail.gmail.com>
	<CAN0jgiN09j08jTx-ABoGkr=QDg6TT21zaHTfuFH+ezyCrj6ViA@mail.gmail.com>
Message-ID: <CADh=FyisEVgzt8tadXLx0DBCZkwLDR8DccqKQwc5XbB3JuzKfg@mail.gmail.com>

Thanks for the input Daniel.

Guys,

Right now we are observing crazy amount of rejections...But the destination
was disabled only a few times..

Any additional data that would be useful to collect?

Regards,
Chandan

On Wed, Aug 10, 2016 at 6:26 PM, Daniel Zanutti <daniel.zanutti at gmail.com>
wrote:

> Hi Chandan
>
> I would suggest upgrading to 1.11.X first, it will be very
> straightforward. If you still have problems, migrating to 2.2 will probably
> not solve them and you may find new ones.
>
> Regards
>
> On Wed, Aug 10, 2016 at 9:16 AM, Chandan PR <chandan.pr at webshar.org>
> wrote:
>
>> Hi Guys,
>>
>> We are using OpenSips, 1.9 Load Balancer with Probing Interval set to 15
>> seconds.
>>
>> Yesterday we observed huge number of calls being rejected by Open Sips as
>> Destinations were full.
>>
>> But the number of error codes which results in disabling the gateway was
>> pretty less on the gateway end.
>>
>> The log file was not rolled over and it was of size ~2.5G.
>>
>> Any one has faced similar issue?
>>
>> I see a new event "E_LOAD_BALANCER_STATUS" available in latest versions,
>> which helps in knowing when the destination was disabled or enabled.
>>
>> Any other possible options in 1.9.x to know the details about when the
>> destination was disabled or enabled?
>>
>> How easy or difficult it is to migrate from 1.9.x to 2.2?
>>
>> Any migration tutorials/links/docs for the same?
>>
>> Thanks,
>> Chandan
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From chandan.pr at webshar.org  Mon Aug 15 22:06:38 2016
From: chandan.pr at webshar.org (Chandan PR)
Date: Tue, 16 Aug 2016 01:36:38 +0530
Subject: [OpenSIPS-Users] OpenSips rejecting calls even though the
	destination is not disabled
Message-ID: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>

Hi Guys,

We are observing an issue where OpenSips is rejecting calls as Destination
Full, even when the destination is not disabled...

We are not sending too many calls for all the destinations to be full.

The script is as below:

#
# $Id: opensips_loadbalancer.m4 9723 2013-02-01 15:10:50Z bogdan_iancu $
#
# OpenSIPS loadbalancer script
#     by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
#   the "Load Balancer" scenario.
# You can enable / disable more features / functionalities by
#   re-generating the scenario with different options.
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL1

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
   based on revers DNS on IPs */
auto_aliases=no


listen=udp:192.168.21.111:5060   # CUSTOMIZE ME

disable_tcp=no
listen=tcp:192.168.21.111:5060   # CUSTOMIZE ME

#disable_tls=yes




####### Modules Section ########

#set module path
mpath="/usr/local/opensips/lib64/opensips/modules/"



#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)

#### MYSQL module
loadmodule "db_mysql.so"

#### AVPOPS module
loadmodule "avpops.so"

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", "ACC_FAILED")
/* account triggers (flags) */
modparam("acc", "log_flag", "ACC_DO")
modparam("acc", "log_missed_flag", "ACC_MISSED")


#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME


#### LOAD BALANCER module
loadmodule "load_balancer.so"
modparam("load_balancer", "db_url",
"mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
modparam("load_balancer", "probing_method", "OPTIONS")

modparam("load_balancer", "probing_interval", 15)





####### Routing Logic ########


# main request routing logic
route{
if (!mf_process_maxfwd_header("3")) {
send_reply("483","looping");
exit;
}


if ( has_totag() ) {
# sequential request -> obey Route indication
xlog("Loose Route for $avp(iurl) over $avp(dest) $avp(calltype) \n");
loose_route();
                t_relay();
                exit;
        }

        # handle cancel and re-transmissions
if ( is_method("CANCEL") ) {
if ( t_check_trans() )
t_relay();
exit;
}


        # from now on we have only the initial requests
        if (!is_method("INVITE")) {
                send_reply("405","Method Not Allowed");
                exit;
        }

        # initial request
record_route();
$avp(iurl) = $(hdr(P-Nuance-Initial-URL));
$avp(calltype) = $(hdr(P-Nuance-Call-Type));
        xlog("Sending call $fu $avp(iurl) $avp(calltype) \n");
# LB function returns negative if no suitable destination (for requested
resources) is found,
        # or if all destinations are full
        if ( !load_balance("1","channel") ) {
    xlog("All Destinations are full. Sending 500! $fu
$(hdr(P-Nuance-Initial-URL))");
             send_reply("500","Service full");
             exit;
        }
$avp(dest) = $du;
xlog("Call sent over $avp(dest) $fu $(hdr(P-Nuance-Initial-URL))\n");
t_on_reply("test");
        # arm a failure route for be able to catch a failure event and to
do
        # failover to the next available destination
        t_on_failure("LB_failed");

        # send it out
if (!t_relay()) {
sl_reply_error();
}

}

#onreply_route {
# xlog("OpenSIPS received a reply $avp(iurl) rs= $rs | fu= $fu | si= $si |
Ri= $Ri ===");
#}

onreply_route[test] {
xlog("Reply $rs received from $si for the call $avp(iurl)");
}

failure_route[LB_failed]
{
        # skip if call was canceled
if (t_was_cancelled()) {
exit;
}
xlog("Call $avp(iurl) sent over $avp(dest) failed with code
$T_reply_code\n");

# Checking only for Destination Failure
# If the destination is failed, disable it and reroute the call
        #if ( t_check_status("[56][0-9][0-9]")) {
#Comment the below line and put the above condition to probe for all
500-699 errors
        if ( t_check_status("503")) {
       # this is a case for failover
                xlog("Marking $avp(dest) disabled as it failed for call
$avp(iurl) with code $T_reply_code\n");
                # mark failed destination as disabled
                lb_disable();
                # try to re-route to next available destination
                #if ( !load_balance("1","channel") ) {
#      xlog("RETRY FAILED SENDING 500");
                #      send_reply("500","Service Full");
                #      exit;
                #}
                #xlog("REPORT: re-routing call to $du \n");
                #t_relay();
        }
send_reply("$T_reply_code","Call Failed");
}


Please do let me know if any comments..

Regards,
Chandan
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From stefano.pisani at omnianet.it  Tue Aug 16 09:52:56 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Tue, 16 Aug 2016 09:52:56 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory
	allocation functions
In-Reply-To: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
Message-ID: <d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>

I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS

version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
main.c compiled on  with gcc 5.3.1

It's not working:
Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !!
Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize module 
tls_mgm

OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016

The opensips installed packages are:
opensips                                        install
opensips-tls-module:amd64                       install
opensips-tlsmgm-module:amd64                    install
opensips-wss-module:amd64                       install


There is something I can do?
Thanks







From chandan.pr at webshar.org  Tue Aug 16 11:47:36 2016
From: chandan.pr at webshar.org (Chandan PR)
Date: Tue, 16 Aug 2016 15:17:36 +0530
Subject: [OpenSIPS-Users] Load Balancer - Delay in clearing dialogs
Message-ID: <CADh=Fyicu3n2wPYGsNYNAEC=na2VWqFqNjmdsCcXEP=k3AxL1w@mail.gmail.com>

Hi Guys,

Opendsips Version : 1.9.x

We have configured the load balancer with 8 ports.

When the calls are ongoing, the lb_list shows the load balancer is running
full.

root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl fifo
lb_list
Destination:: sip:192.168.248.62:5060 id=152 group=1 enabled=yes auto-re=on
Resource:: channel max=4 load=4
Destination:: sip:192.168.248.64:5060 id=153 group=1 enabled=yes auto-re=on
Resource:: channel max=4 load=4

When I check the dialog_list, I see some older dialogs which are yet to be
cleared.

root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl fifo
dlg_list | grep "callid"
callid:: 14713396610041471339775929_ZTLT_931
callid:: 14713396610151471339775929_ZTLT_937
callid:: 14713396610221471339775929_ZTLT_942
callid:: 14713396610081471339775929_ZTLT_933
callid:: 14713396610181471339775929_ZTLT_939
callid:: 14713396610191471339775929_ZTLT_940
callid:: 14713396610131471339775929_ZTLT_936
callid:: 14713396610031471339775929_ZTLT_930
callid:: 14713396610061471339775929_ZTLT_932
callid:: 14713396610161471339775929_ZTLT_938
callid:: 14713396610271471339775929_ZTLT_945
callid:: 14713396610371471339775929_ZTLT_951
callid:: 14713396610101471339775929_ZTLT_934
callid:: 14713396610311471339775929_ZTLT_947
callid:: 14713396610211471339775929_ZTLT_941
callid:: 14713396610241471339775929_ZTLT_943
callid:: 14713396610341471339775929_ZTLT_949
callid:: 14713396610011471339775929_ZTLT_929
callid:: 14713396610111471339775929_ZTLT_935
root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl fifo
dlg_list | grep "callid" | wc -l
13
root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin#

*Even when the lb_list shows the load as 0, I see some dialogs present and
getting cleared after around 10-12 seconds.*

What would be the impact of this delay in clearing the dialogs? Can this
lead to lot many calls rejected with "Destination Full" (We are
experiencing this issue in our production, where a huge number of calls
rejected)?

Is there any way to reduce this delay or clear the dialogs immediately?

*Dialog Module Settings:*
#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw at localhost/opensips") #


*Load Balancer Script:*
#
# $Id: opensips_loadbalancer.m4 9723 2013-02-01 15:10:50Z bogdan_iancu $
#
# OpenSIPS loadbalancer script
#     by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
#   the "Load Balancer" scenario.
# You can enable / disable more features / functionalities by
#   re-generating the scenario with different options.
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL1

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
   based on revers DNS on IPs */
auto_aliases=no


listen=udp:192.168.248.111:5060   # CUSTOMIZE ME

disable_tcp=no
listen=tcp:192.168.248.111:5060   # CUSTOMIZE ME

disable_tls=yes




####### Modules Section ########

#set module path
mpath="/usr/local/opensips/lib64/opensips/modules/"



#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)

#### MYSQL module
loadmodule "db_mysql.so"

#### AVPOPS module
loadmodule "avpops.so"

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", "ACC_FAILED")
/* account triggers (flags) */
modparam("acc", "log_flag", "ACC_DO")
modparam("acc", "log_missed_flag", "ACC_MISSED")


#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME


#### LOAD BALANCER module
loadmodule "load_balancer.so"
modparam("load_balancer", "db_url",
"mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
modparam("load_balancer", "probing_method", "OPTIONS")

modparam("load_balancer", "probing_interval", 15)





####### Routing Logic ########


# main request routing logic
route{
if (!mf_process_maxfwd_header("3")) {
send_reply("483","looping");
exit;
}


if ( has_totag() ) {
# sequential request -> obey Route indication
xlog("Loose Route for $avp(iurl) over $avp(dest) $avp(calltype) \n");
loose_route();
                t_relay();
                exit;
        }

        # handle cancel and re-transmissions
if ( is_method("CANCEL") ) {
if ( t_check_trans() )
t_relay();
exit;
}


        # from now on we have only the initial requests
        if (!is_method("INVITE")) {
                send_reply("405","Method Not Allowed");
                exit;
        }

        # initial request
record_route();
$avp(iurl) = $(hdr(P-Nuance-Initial-URL));
$avp(calltype) = $(hdr(P-Nuance-Call-Type));
        xlog("Sending call $fu $avp(iurl) $avp(calltype) \n");
xlog("Current Dialogs count $DLG_count \n");
# LB function returns negative if no suitable destination (for requested
resources) is found,
        # or if all destinations are full
        if ( !load_balance("1","channel") ) {
    xlog("All Destinations are full. Sending 500! $fu
$(hdr(P-Nuance-Initial-URL)) CURRENT DIALOGS $DLG_count \n");
             send_reply("500","Service full");
             exit;
        }
$avp(dest) = $du;
xlog("Call sent over $avp(dest) $fu $(hdr(P-Nuance-Initial-URL))\n");
t_on_reply("test");
        # arm a failure route for be able to catch a failure event and to
do
        # failover to the next available destination
        t_on_failure("LB_failed");

        # send it out
if (!t_relay()) {
sl_reply_error();
}

}

#onreply_route {
# xlog("OpenSIPS received a reply $avp(iurl) rs= $rs | fu= $fu | si= $si |
Ri= $Ri ===");
#}

onreply_route[test] {
xlog("Reply $rs received from $si for the call $avp(iurl)");
}

failure_route[LB_failed]
{
        # skip if call was canceled
if (t_was_cancelled()) {
exit;
}
xlog("Call $avp(iurl) sent over $avp(dest) failed with code
$T_reply_code\n");

# Checking only for Destination Failure
# If the destination is failed, disable it and reroute the call
        #if ( t_check_status("[56][0-9][0-9]")) {
        if ( t_check_status("503")) {
       # this is a case for failover
                xlog("Marking $avp(dest) disabled as it failed for call
$avp(iurl) with code $T_reply_code\n");
                #xlog("call $avp(iurl) failed with code $T_reply_code from
$avp(dest)\n");
# mark failed destination as disabled
                lb_disable();
                # try to re-route to next available destination
                #if ( !load_balance("1","channel") ) {
#      xlog("RETRY FAILED SENDING 500");
                #      send_reply("500","Service Full");
                #      exit;
                #}
                #xlog("REPORT: re-routing call to $du \n");
                #t_relay();
        }
send_reply("$T_reply_code","Call Failed");
}


Regards,
Chandan
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From rrobson at greenlightcrm.com  Tue Aug 16 13:18:38 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Tue, 16 Aug 2016 12:18:38 +0100
Subject: [OpenSIPS-Users] rate limiter error (2.2.1)
Message-ID: <2ab4cf3b-57f6-ae70-2930-588b13648847@greenlightcrm.com>

I'm using the rl module to limit customers using an account code and 
calls to a trunk via its name.

i'm getting the following error in the tunk one only:

ERROR:ratelimit:pv_get_rl_count: Cannot get counter's value

the module creates both the PIPEs as expected:

opensipsctl fifo rl_list
PIPE::  id=gl-agw-01.greenlightcrm.local algorithm=TAILDROP limit=10 
counter=0
PIPE::  id=0001 algorithm=TAILDROP limit=20 counter=0
PIPE::  drop_rate=0


but the error is for the top one (id=gl-agw-01.greenlightcrm.local) as 
the other one is OK and returns a value and does not error


-- 
Richard Robson
Greenlight Support
01382 843843
support at greenlightcrm.com



From rrobson at greenlightcrm.com  Tue Aug 16 13:30:14 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Tue, 16 Aug 2016 12:30:14 +0100
Subject: [OpenSIPS-Users] rate limiter error (2.2.1)
In-Reply-To: <2ab4cf3b-57f6-ae70-2930-588b13648847@greenlightcrm.com>
References: <2ab4cf3b-57f6-ae70-2930-588b13648847@greenlightcrm.com>
Message-ID: <ff828b20-fa9d-08e0-ef55-118f6d4cc030@greenlightcrm.com>

Please ignore post, I found the error I was calling the count with a 
different pv

On 16/08/2016 12:18, Richard Robson wrote:
> I'm using the rl module to limit customers using an account code and 
> calls to a trunk via its name.
>
> i'm getting the following error in the tunk one only:
>
> ERROR:ratelimit:pv_get_rl_count: Cannot get counter's value
>
> the module creates both the PIPEs as expected:
>
> opensipsctl fifo rl_list
> PIPE::  id=gl-agw-01.greenlightcrm.local algorithm=TAILDROP limit=10 
> counter=0
> PIPE::  id=0001 algorithm=TAILDROP limit=20 counter=0
> PIPE::  drop_rate=0
>
>
> but the error is for the top one (id=gl-agw-01.greenlightcrm.local) as 
> the other one is OK and returns a value and does not error
>
>


-- 
Richard Robson
Greenlight Support
01382 843843
support at greenlightcrm.com



From Agalya_Ramachandran at comcast.com  Mon Aug 15 22:35:14 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Mon, 15 Aug 2016 20:35:14 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
In-Reply-To: <64550c2d1fc944d7b88b5cd2f527db75@COPDCEX28.cable.comcast.com>
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
	<64550c2d1fc944d7b88b5cd2f527db75@COPDCEX28.cable.comcast.com>
Message-ID: <99a79810857c45078538a12eb0230bf9@COPDCEX28.cable.comcast.com>

Hi Bogdan/Liviu,

Further debugging I got some info that I can share with you so that you can help me out easily.
Instead of calling rest_put asynchronously I have tried to call rest_post async . I could able to see that HTTP request went out(from wireshark traces) and I got 4xx response for the same.
With my code changes in place and when I call rest_put, am not seeing HTTP request sent out.
Assuming rest_put only is not working, there should be some libcurl stuffs not handled properly in 'start_async_http_req' for 'REST_CLIENT_PUT' case.
Am attaching the files I have modified. Can you please take a look guide me here.

And regarding the below error, am not seeing it every time now. It's coming now and then. Sometimes in my debug I could able to see the async_start and async_resume has executed for rest_put as well.
But the HTTP request never went to the destination.
Please let me know if you need more information on this.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Ramachandran, Agalya (Contractor)
Sent: Monday, August 15, 2016 9:32 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Bogdan/Liviu,

Please help on this async() to get worked at config file.
Am using rest_put() and I have added the needed functions and declarations for the same as like rest_post().

Am getting the below error if I enable debug logs.
Seems, w_async_rest_put function is called and after that it didn't process the async(rest_put)).
Do we need to configure anything specific when async() is called in config file?

Aug 11 19:49:52 DBG:tm:_reply_light: finished
Aug 11 19:49:52 DBG:rest_client:w_async_rest_put: async rest put 'http://***URL***?' 0x7f3df7848538 0x7f3df78485d0 0x7f3df7848650
Aug 11 19:50:12 DBG:rest_client:start_async_http_req: done, no need for async!
Aug 11 19:50:12 DBG:rest_client:set_output_pv_params: Last response code: 0
Aug 11 19:50:12 DBG:core:fm_free: free(0) called
Aug 11 19:50:12 DBG:core:comp_scriptvar: int 26 : -4 / 0
Aug 11 19:50:12 Return code is -4
Aug 11 19:50:12 Error code 0 in HTTP PUT!

Regards,
Agalya


From: Ramachandran, Agalya (Contractor)
Sent: Tuesday, August 09, 2016 3:47 PM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: RE: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c

If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** <http://***url***%20> ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

But I never see rest_put() is called. Rather am getting below error continuously.
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution
Aug  9 17:27:47 sturn-test /usr/local/sbin/opensips[18525]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 403840000 us delay in execution

Am I missing something here? I am sure it is a small thing. Do we need to do anything else before calling async in the script?

Regards,
Agalya

From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From Agalya_Ramachandran at comcast.com  Tue Aug 16 16:23:23 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 16 Aug 2016 14:23:23 +0000
Subject: [OpenSIPS-Users] Asynchronous operation for REST queries
In-Reply-To: <d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
Message-ID: <b87dda888e6f43a5a6cf143481c27b6b@COPDCEX28.cable.comcast.com>

Hi Bogdan/Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c
If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** <http://***url***%20> ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

Further debugging I got some info that I can share with you so that you can help me out easily.
Instead of calling rest_put asynchronously I have tried to call rest_post async . I could able to see that HTTP request went out(from wireshark traces) and I got 4xx response for the same.

With my code changes in place and when I call rest_put, am not seeing HTTP request sent out.
Assuming rest_put only is not working, there should be some libcurl stuffs not handled properly in 'start_async_http_req' for 'REST_CLIENT_PUT' case.

Am attaching the files. It contains code for rest_put only. I have removed other code to reduce the file size. Can you please take a look guide me here.
Please let me know if you need more information on this.

Regards,
Agalya


From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.


Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From razvan at opensips.org  Tue Aug 16 17:18:30 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 16 Aug 2016 18:18:30 +0300
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
Message-ID: <98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>

Stefano!

I've added some extra debugging in the master branch that might help you 
debug this. Could you please take the latest git version/deb and try to 
run again?

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016 10:52 AM, Stefano Pisani wrote:
> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>
> version: opensips 2.2.1 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> main.c compiled on  with gcc 5.3.1
>
> It's not working:
> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS protocol
> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
> memory allocation functions
> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you have 
> openssl 1.0.1e-fips, as this version is know to be broken; if so, you 
> need to upgrade or downgrade to a differen openssl version !!
> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
> module tls_mgm
>
> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>
> The opensips installed packages are:
> opensips                                        install
> opensips-tls-module:amd64                       install
> opensips-tlsmgm-module:amd64                    install
> opensips-wss-module:amd64                       install
>
>
> There is something I can do?
> Thanks
>
>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From stefano.pisani at omnianet.it  Tue Aug 16 17:56:27 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Tue, 16 Aug 2016 17:56:27 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
Message-ID: <4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>

I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head the 
error appears the same Aug 16 11:55:14 [25630] INFO:tls_mgm:mod_init: 
initializing TLS protocol Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: unable to set the memory allocation functions 
Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !! Aug 16 
11:55:14 [25630] ERROR:core:init_mod: failed to initialize module 
tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error while 
initializing modules Aug 16 11:55:14 [25630] INFO:core:cleanup: cleanup 
Aug 16 11:55:14 [25630] DBG:dispatcher:destroy: destroying module ... 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 
11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 
0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table Aug 16 
11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 16 11:55:14 
[25630] DBG:tm:tm_shutdown: removing semaphores Aug 16 11:55:14 [25630] 
DBG:tm:tm_shutdown: destroying callback lists Aug 16 11:55:14 [25630] 
DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 11:55:14 [25630] 
DBG:core:shm_mem_destroy: destroying the shared memory lock Aug 16 
11:55:14 [25630] NOTICE:core:main: Exiting.... Thanks Stefano |


Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
> Stefano!
>
> I've added some extra debugging in the master branch that might help 
> you debug this. Could you please take the latest git version/deb and 
> try to run again?
>
> Best regards,
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>
>> version: opensips 2.2.1 (x86_64/linux)
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>> main.c compiled on  with gcc 5.3.1
>>
>> It's not working:
>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS protocol
>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
>> memory allocation functions
>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
>> have openssl 1.0.1e-fips, as this version is know to be broken; if 
>> so, you need to upgrade or downgrade to a differen openssl version !!
>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
>> module tls_mgm
>>
>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>
>> The opensips installed packages are:
>> opensips                                        install
>> opensips-tls-module:amd64                       install
>> opensips-tlsmgm-module:amd64                    install
>> opensips-wss-module:amd64                       install
>>
>>
>> There is something I can do?
>> Thanks
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-------------- next part --------------
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From razvan at opensips.org  Tue Aug 16 18:23:43 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 16 Aug 2016 19:23:43 +0300
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
Message-ID: <b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>

Hi, Stefano!

I don't think you are using the latest sources. Either you have not 
installed them, or you are using a wrong path. Could you please 
double-check?

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>
> I cloned the source from
>
> |git clone https://github.com/OpenSIPS/opensips.git opensips_head the 
> error appears the same Aug 16 11:55:14 [25630] INFO:tls_mgm:mod_init: 
> initializing TLS protocol Aug 16 11:55:14 [25630] 
> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions 
> Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: check if you 
> have openssl 1.0.1e-fips, as this version is know to be broken; if so, 
> you need to upgrade or downgrade to a differen openssl version !! Aug 
> 16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize module 
> tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error while 
> initializing modules Aug 16 11:55:14 [25630] INFO:core:cleanup: 
> cleanup Aug 16 11:55:14 [25630] DBG:dispatcher:destroy: destroying 
> module ... Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : 
> start Aug 16 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying 
> DELETE list for set 0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
> emptying hash table Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
> releasing timers Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing 
> semaphores Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying 
> callback lists Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown 
> : done Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying 
> the shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
> Exiting.... Thanks Stefano |
>
> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>> Stefano!
>>
>> I've added some extra debugging in the master branch that might help 
>> you debug this. Could you please take the latest git version/deb and 
>> try to run again?
>>
>> Best regards,
>>
>> R?zvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>>
>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>
>>> version: opensips 2.2.1 (x86_64/linux)
>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>> main.c compiled on  with gcc 5.3.1
>>>
>>> It's not working:
>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS protocol
>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
>>> memory allocation functions
>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>> have openssl 1.0.1e-fips, as this version is know to be broken; if 
>>> so, you need to upgrade or downgrade to a differen openssl version !!
>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
>>> module tls_mgm
>>>
>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>
>>> The opensips installed packages are:
>>> opensips                                        install
>>> opensips-tls-module:amd64                       install
>>> opensips-tlsmgm-module:amd64                    install
>>> opensips-wss-module:amd64                       install
>>>
>>>
>>> There is something I can do?
>>> Thanks
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From stefano.pisani at omnianet.it  Tue Aug 16 18:26:51 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Tue, 16 Aug 2016 18:26:51 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
Message-ID: <8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>
> Hi, Stefano!
>
> I don't think you are using the latest sources. Either you have not 
> installed them, or you are using a wrong path. Could you please 
> double-check?
>
> Best regards,
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>
>> I cloned the source from
>>
>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head the 
>> error appears the same Aug 16 11:55:14 [25630] INFO:tls_mgm:mod_init: 
>> initializing TLS protocol Aug 16 11:55:14 [25630] 
>> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions 
>> Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: check if you 
>> have openssl 1.0.1e-fips, as this version is know to be broken; if 
>> so, you need to upgrade or downgrade to a differen openssl version !! 
>> Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
>> module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error while 
>> initializing modules Aug 16 11:55:14 [25630] INFO:core:cleanup: 
>> cleanup Aug 16 11:55:14 [25630] DBG:dispatcher:destroy: destroying 
>> module ... Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : 
>> start Aug 16 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying 
>> DELETE list for set 0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>> emptying hash table Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>> releasing timers Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing 
>> semaphores Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying 
>> callback lists Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>> tm_shutdown : done Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: 
>> destroying the shared memory lock Aug 16 11:55:14 [25630] 
>> NOTICE:core:main: Exiting.... Thanks Stefano |
>>
>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>> Stefano!
>>>
>>> I've added some extra debugging in the master branch that might help 
>>> you debug this. Could you please take the latest git version/deb and 
>>> try to run again?
>>>
>>> Best regards,
>>>
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>>
>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>
>>>> version: opensips 2.2.1 (x86_64/linux)
>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> main.c compiled on  with gcc 5.3.1
>>>>
>>>> It's not working:
>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>> protocol
>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
>>>> memory allocation functions
>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>> have openssl 1.0.1e-fips, as this version is know to be broken; if 
>>>> so, you need to upgrade or downgrade to a differen openssl version !!
>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
>>>> module tls_mgm
>>>>
>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>
>>>> The opensips installed packages are:
>>>> opensips                                        install
>>>> opensips-tls-module:amd64                       install
>>>> opensips-tlsmgm-module:amd64                    install
>>>> opensips-wss-module:amd64                       install
>>>>
>>>>
>>>> There is something I can do?
>>>> Thanks
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From acmicrox at gmail.com  Wed Aug 17 09:36:13 2016
From: acmicrox at gmail.com (Chen-Che Huang)
Date: Wed, 17 Aug 2016 00:36:13 -0700 (MST)
Subject: [OpenSIPS-Users] Dispatcher command question
Message-ID: <1471419373119-7604038.post@n2.nabble.com>

Hi all,

I'm using the dispatcher module with DBTEXT. With the preconfigured
/etc/opensips/dispatcher file, the front OpenSIPS proxy works with two
backend SIP proxies. Now, I would like to add a new backend SIP proxy at the
run time of the front OpenSIPS proxy. I have tried to modified the
dispatcher file and then call 'sudo opensipsctl fifo ds_reload'. But I could
not see the added SIP proxy by running sudo opensipsctl fifo ds_list'. I
also try call `sudo opensipsctl dispatcher addgw'. However, I can see the
added SIP proxy by `sudo opensipsctl dispatcher show' but not in the result
of `sudo opensipsctl dispatcher show'. It seems that  the content of
text-based db does not sync with the memory. I'm using opensips version
1.11.6. I wonder it is feasible to add a new backend proxy during the run
time in the dispatcher module? Any comment is appreciated.

Best regards,
Chen-Che



--
View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-command-question-tp7604038.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.


From razvan at opensips.org  Wed Aug 17 09:53:17 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 17 Aug 2016 10:53:17 +0300
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
Message-ID: <179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>

Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone and try 
again?

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016 07:26 PM, Stefano Pisani wrote:
> I have this result.
>
> # ./opensips -V
> version: opensips 2.3.0-dev (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> git revision: 058991d
> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>
>
>
> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>
>> Hi, Stefano!
>>
>> I don't think you are using the latest sources. Either you have not 
>> installed them, or you are using a wrong path. Could you please 
>> double-check?
>>
>> Best regards,
>>
>> R?zvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>
>>> I cloned the source from
>>>
>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head the 
>>> error appears the same Aug 16 11:55:14 [25630] 
>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory allocation 
>>> functions Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: 
>>> check if you have openssl 1.0.1e-fips, as this version is know to be 
>>> broken; if so, you need to upgrade or downgrade to a differen 
>>> openssl version !! Aug 16 11:55:14 [25630] ERROR:core:init_mod: 
>>> failed to initialize module tls_mgm Aug 16 11:55:14 [25630] 
>>> ERROR:core:main: error while initializing modules Aug 16 11:55:14 
>>> [25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
>>> DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
>>> [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
>>> [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table Aug 
>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 16 
>>> 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 16 
>>> 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists Aug 
>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 
>>> 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the shared 
>>> memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: Exiting.... 
>>> Thanks Stefano |
>>>
>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>> Stefano!
>>>>
>>>> I've added some extra debugging in the master branch that might 
>>>> help you debug this. Could you please take the latest git 
>>>> version/deb and try to run again?
>>>>
>>>> Best regards,
>>>>
>>>> R?zvan Crainea
>>>> OpenSIPS Solutions
>>>> www.opensips-solutions.com
>>>>
>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>
>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>> main.c compiled on  with gcc 5.3.1
>>>>>
>>>>> It's not working:
>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>> protocol
>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
>>>>> memory allocation functions
>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>>> have openssl 1.0.1e-fips, as this version is know to be broken; if 
>>>>> so, you need to upgrade or downgrade to a differen openssl version !!
>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
>>>>> module tls_mgm
>>>>>
>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>
>>>>> The opensips installed packages are:
>>>>> opensips                                        install
>>>>> opensips-tls-module:amd64                       install
>>>>> opensips-tlsmgm-module:amd64                    install
>>>>> opensips-wss-module:amd64                       install
>>>>>
>>>>>
>>>>> There is something I can do?
>>>>> Thanks
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From stefano.pisani at omnianet.it  Wed Aug 17 11:06:16 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Wed, 17 Aug 2016 11:06:16 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
Message-ID: <9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>

Hi R?zvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
need to upgrade or downgrade to a different openssl version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 free=0x7f0cdb08462c/0x7f0ce10d8a70 
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize module 
tls_mgm
Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules

Thanks.
Stefano

Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>
> Hi, Stefano!
>
> Apologies, I did not push the changes. Could you please re-clone and 
> try again?
>
> Best regards,
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>> I have this result.
>>
>> # ./opensips -V
>> version: opensips 2.3.0-dev (x86_64/linux)
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>> git revision: 058991d
>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>
>>
>>
>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>
>>> Hi, Stefano!
>>>
>>> I don't think you are using the latest sources. Either you have not 
>>> installed them, or you are using a wrong path. Could you please 
>>> double-check?
>>>
>>> Best regards,
>>>
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>
>>>> I cloned the source from
>>>>
>>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head 
>>>> the error appears the same Aug 16 11:55:14 [25630] 
>>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory allocation 
>>>> functions Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: 
>>>> check if you have openssl 1.0.1e-fips, as this version is know to 
>>>> be broken; if so, you need to upgrade or downgrade to a differen 
>>>> openssl version !! Aug 16 11:55:14 [25630] ERROR:core:init_mod: 
>>>> failed to initialize module tls_mgm Aug 16 11:55:14 [25630] 
>>>> ERROR:core:main: error while initializing modules Aug 16 11:55:14 
>>>> [25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
>>>> DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
>>>> [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
>>>> [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table Aug 
>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 16 
>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 16 
>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists Aug 
>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 
>>>> 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the shared 
>>>> memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: Exiting.... 
>>>> Thanks Stefano |
>>>>
>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>> Stefano!
>>>>>
>>>>> I've added some extra debugging in the master branch that might 
>>>>> help you debug this. Could you please take the latest git 
>>>>> version/deb and try to run again?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> R?zvan Crainea
>>>>> OpenSIPS Solutions
>>>>> www.opensips-solutions.com
>>>>>
>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>>
>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>>>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>
>>>>>> It's not working:
>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>> protocol
>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
>>>>>> memory allocation functions
>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>>>> have openssl 1.0.1e-fips, as this version is know to be broken; 
>>>>>> if so, you need to upgrade or downgrade to a differen openssl 
>>>>>> version !!
>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
>>>>>> module tls_mgm
>>>>>>
>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>
>>>>>> The opensips installed packages are:
>>>>>> opensips                                        install
>>>>>> opensips-tls-module:amd64                       install
>>>>>> opensips-tlsmgm-module:amd64                    install
>>>>>> opensips-wss-module:amd64                       install
>>>>>>
>>>>>>
>>>>>> There is something I can do?
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From razvan at opensips.org  Wed Aug 17 11:16:13 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 17 Aug 2016 12:16:13 +0300
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
Message-ID: <1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>

Can you run 'gdb ./opensips' and run the following commands:
info line *0x7f0ce10d8a70
info line *0x7f0ce10d8550

Thanks,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>
> Hi R?zvan.
>
> These are the new logs. I hope it helps to fix this.
>
> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
> memory allocation functions
> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you have 
> openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
> need to upgrade or downgrade to a different openssl version !!
> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips 1 Mar 2016
> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
> module tls_mgm
> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules
>
> Thanks.
> Stefano
>
> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>
>> Hi, Stefano!
>>
>> Apologies, I did not push the changes. Could you please re-clone and 
>> try again?
>>
>> Best regards,
>>
>> R?zvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>> I have this result.
>>>
>>> # ./opensips -V
>>> version: opensips 2.3.0-dev (x86_64/linux)
>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>> git revision: 058991d
>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>
>>>
>>>
>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>
>>>> Hi, Stefano!
>>>>
>>>> I don't think you are using the latest sources. Either you have not 
>>>> installed them, or you are using a wrong path. Could you please 
>>>> double-check?
>>>>
>>>> Best regards,
>>>>
>>>> R?zvan Crainea
>>>> OpenSIPS Solutions
>>>> www.opensips-solutions.com
>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>
>>>>> I cloned the source from
>>>>>
>>>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head 
>>>>> the error appears the same Aug 16 11:55:14 [25630] 
>>>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory 
>>>>> allocation functions Aug 16 11:55:14 [25630] 
>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you need 
>>>>> to upgrade or downgrade to a differen openssl version !! Aug 16 
>>>>> 11:55:14 [25630] ERROR:core:init_mod: failed to initialize module 
>>>>> tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error while 
>>>>> initializing modules Aug 16 11:55:14 [25630] INFO:core:cleanup: 
>>>>> cleanup Aug 16 11:55:14 [25630] DBG:dispatcher:destroy: destroying 
>>>>> module ... Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown 
>>>>> : start Aug 16 11:55:14 [25630] DBG:tm:unlink_timer_lists: 
>>>>> emptying DELETE list for set 0 Aug 16 11:55:14 [25630] 
>>>>> DBG:tm:tm_shutdown: emptying hash table Aug 16 11:55:14 [25630] 
>>>>> DBG:tm:tm_shutdown: releasing timers Aug 16 11:55:14 [25630] 
>>>>> DBG:tm:tm_shutdown: removing semaphores Aug 16 11:55:14 [25630] 
>>>>> DBG:tm:tm_shutdown: destroying callback lists Aug 16 11:55:14 
>>>>> [25630] DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 11:55:14 
>>>>> [25630] DBG:core:shm_mem_destroy: destroying the shared memory 
>>>>> lock Aug 16 11:55:14 [25630] NOTICE:core:main: Exiting.... Thanks 
>>>>> Stefano |
>>>>>
>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>> Stefano!
>>>>>>
>>>>>> I've added some extra debugging in the master branch that might 
>>>>>> help you debug this. Could you please take the latest git 
>>>>>> version/deb and try to run again?
>>>>>>
>>>>>> Best regards,
>>>>>>
>>>>>> R?zvan Crainea
>>>>>> OpenSIPS Solutions
>>>>>> www.opensips-solutions.com
>>>>>>
>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>>>
>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>
>>>>>>> It's not working:
>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>>> protocol
>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
>>>>>>> memory allocation functions
>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if 
>>>>>>> you have openssl 1.0.1e-fips, as this version is know to be 
>>>>>>> broken; if so, you need to upgrade or downgrade to a differen 
>>>>>>> openssl version !!
>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
>>>>>>> module tls_mgm
>>>>>>>
>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>
>>>>>>> The opensips installed packages are:
>>>>>>> opensips install
>>>>>>> opensips-tls-module:amd64 install
>>>>>>> opensips-tlsmgm-module:amd64 install
>>>>>>> opensips-wss-module:amd64 install
>>>>>>>
>>>>>>>
>>>>>>> There is something I can do?
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From stefano.pisani at omnianet.it  Wed Aug 17 11:20:46 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Wed, 17 Aug 2016 11:20:46 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
Message-ID: <b0a6b8b0-817e-4f24-efc2-bffed8b2d84a@omnianet.it>

# gdb ./opensips
GNU gdb (Ubuntu 7.11.1-0ubuntu1~16.04) 7.11.1
Copyright (C) 2016 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later 
<http://gnu.org/licenses/gpl.html>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
Type "show configuration" for configuration details.
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>.
Find the GDB manual and other documentation resources online at:
<http://www.gnu.org/software/gdb/documentation/>.
For help, type "help".
Type "apropos word" to search for commands related to "word"...
Reading symbols from ./opensips...done.
(gdb) info line *0x7f0ce10d8a70
No line number information available for address 0x7f0ce10d8a70
(gdb) info line *0x7f0ce10d8550
No line number information available for address 0x7f0ce10d8550
(gdb)

Thanks
Stefano


Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
> Can you run 'gdb ./opensips' and run the following commands:
> info line *0x7f0ce10d8a70
> info line *0x7f0ce10d8550
>
> Thanks,
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>
>> Hi R?zvan.
>>
>> These are the new logs. I hope it helps to fix this.
>>
>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>> memory allocation functions
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>> so, you need to upgrade or downgrade to a different openssl version !!
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 Mar 
>> 2016
>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>> module tls_mgm
>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules
>>
>> Thanks.
>> Stefano
>>
>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>
>>> Hi, Stefano!
>>>
>>> Apologies, I did not push the changes. Could you please re-clone and 
>>> try again?
>>>
>>> Best regards,
>>>
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>> I have this result.
>>>>
>>>> # ./opensips -V
>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> git revision: 058991d
>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>
>>>>
>>>>
>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>
>>>>> Hi, Stefano!
>>>>>
>>>>> I don't think you are using the latest sources. Either you have 
>>>>> not installed them, or you are using a wrong path. Could you 
>>>>> please double-check?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> R?zvan Crainea
>>>>> OpenSIPS Solutions
>>>>> www.opensips-solutions.com
>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>
>>>>>> I cloned the source from
>>>>>>
>>>>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head 
>>>>>> the error appears the same Aug 16 11:55:14 [25630] 
>>>>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>>>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory 
>>>>>> allocation functions Aug 16 11:55:14 [25630] 
>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>> need to upgrade or downgrade to a differen openssl version !! Aug 
>>>>>> 16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
>>>>>> module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error 
>>>>>> while initializing modules Aug 16 11:55:14 [25630] 
>>>>>> INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
>>>>>> DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
>>>>>> [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
>>>>>> [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 
>>>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 
>>>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done 
>>>>>> Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the 
>>>>>> shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
>>>>>> Exiting.... Thanks Stefano |
>>>>>>
>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>> Stefano!
>>>>>>>
>>>>>>> I've added some extra debugging in the master branch that might 
>>>>>>> help you debug this. Could you please take the latest git 
>>>>>>> version/deb and try to run again?
>>>>>>>
>>>>>>> Best regards,
>>>>>>>
>>>>>>> R?zvan Crainea
>>>>>>> OpenSIPS Solutions
>>>>>>> www.opensips-solutions.com
>>>>>>>
>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>>>>
>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>
>>>>>>>> It's not working:
>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>>>> protocol
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set 
>>>>>>>> the memory allocation functions
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if 
>>>>>>>> you have openssl 1.0.1e-fips, as this version is know to be 
>>>>>>>> broken; if so, you need to upgrade or downgrade to a differen 
>>>>>>>> openssl version !!
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>> initialize module tls_mgm
>>>>>>>>
>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>
>>>>>>>> The opensips installed packages are:
>>>>>>>> opensips install
>>>>>>>> opensips-tls-module:amd64 install
>>>>>>>> opensips-tlsmgm-module:amd64 install
>>>>>>>> opensips-wss-module:amd64 install
>>>>>>>>
>>>>>>>>
>>>>>>>> There is something I can do?
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From bogdan at opensips.org  Wed Aug 17 11:26:30 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 17 Aug 2016 12:26:30 +0300
Subject: [OpenSIPS-Users] OpenSips rejecting calls even though the
 destination is not disabled
In-Reply-To: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
Message-ID: <bfd0db92-a5ae-12fb-1c73-13163ca664c2@opensips.org>

Hi Chandan,

What is the actual rejection reply from OpenSIPS ? Is it something from 
script or something that OpenSIPS receives from upstream ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 15.08.2016 23:06, Chandan PR wrote:
> Hi Guys,
>
> We are observing an issue where OpenSips is rejecting calls as 
> Destination Full, even when the destination is not disabled...
>
> We are not sending too many calls for all the destinations to be full.
>
> The script is as below:
>
> #
> # $Id: opensips_loadbalancer.m4 9723 2013-02-01 15:10:50Z bogdan_iancu $
> #
> # OpenSIPS loadbalancer script
> #     by OpenSIPS Solutions <team at opensips-solutions.com 
> <mailto:team at opensips-solutions.com>>
> #
> # This script was generated via "make menuconfig", from
> #   the "Load Balancer" scenario.
> # You can enable / disable more features / functionalities by
> #   re-generating the scenario with different options.
> #
> # Please refer to the Core CookBook at:
> # http://www.opensips.org/Resources/DocsCookbooks
> # for a explanation of possible statements, functions and parameters.
> #
>
>
> ####### Global Parameters #########
>
> debug=3
> log_stderror=no
> log_facility=LOG_LOCAL1
>
> fork=yes
> children=4
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
> /* uncomment the next line to enable the auto temporary blacklisting of
>    not available destinations (default disabled) */
> #disable_dns_blacklist=no
>
> /* uncomment the next line to enable IPv6 lookup after IPv4 dns
>    lookup failures (default disabled) */
> #dns_try_ipv6=yes
>
> /* comment the next line to enable the auto discovery of local aliases
>    based on revers DNS on IPs */
> auto_aliases=no
>
>
> listen=udp:192.168.21.111:5060 <http://192.168.21.111:5060>   # 
> CUSTOMIZE ME
>
> disable_tcp=no
> listen=tcp:192.168.21.111:5060 <http://192.168.21.111:5060>   # 
> CUSTOMIZE ME
>
> #disable_tls=yes
>
>
>
>
> ####### Modules Section ########
>
> #set module path
> mpath="/usr/local/opensips/lib64/opensips/modules/"
>
>
>
> #### SIGNALING module
> loadmodule "signaling.so"
>
> #### StateLess module
> loadmodule "sl.so"
>
> #### Transaction Module
> loadmodule "tm.so"
> modparam("tm", "fr_timer", 30)
> modparam("tm", "fr_inv_timer", 120)
> modparam("tm", "restart_fr_on_each_reply", 0)
> modparam("tm", "onreply_avp_mode", 1)
>
> #### Record Route Module
> loadmodule "rr.so"
> /* do not append from tag to the RR (no need for this script) */
> modparam("rr", "append_fromtag", 0)
>
> #### MAX ForWarD module
> loadmodule "maxfwd.so"
>
> #### SIP MSG OPerationS module
> loadmodule "sipmsgops.so"
>
> #### FIFO Management Interface
> loadmodule "mi_fifo.so"
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0666)
>
> #### URI module
> loadmodule "uri.so"
> modparam("uri", "use_uri_table", 0)
>
> #### MYSQL module
> loadmodule "db_mysql.so"
>
> #### AVPOPS module
> loadmodule "avpops.so"
>
> #### ACCounting module
> loadmodule "acc.so"
> /* what special events should be accounted ? */
> modparam("acc", "early_media", 0)
> modparam("acc", "report_cancels", 0)
> /* by default we do not adjust the direct of the sequential requests.
>    if you enable this parameter, be sure the enable "append_fromtag"
>    in "rr" module */
> modparam("acc", "detect_direction", 0)
> modparam("acc", "failed_transaction_flag", "ACC_FAILED")
> /* account triggers (flags) */
> modparam("acc", "log_flag", "ACC_DO")
> modparam("acc", "log_missed_flag", "ACC_MISSED")
>
>
> #### DIALOG module
> loadmodule "dialog.so"
> modparam("dialog", "dlg_match_mode", 1)
> modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
> modparam("dialog", "db_mode", 2)
> modparam("dialog", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
>
>
> #### LOAD BALANCER module
> loadmodule "load_balancer.so"
> modparam("load_balancer", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
> modparam("load_balancer", "probing_method", "OPTIONS")
>
> modparam("load_balancer", "probing_interval", 15)
>
>
>
>
>
> ####### Routing Logic ########
>
>
> # main request routing logic
> route{
> if (!mf_process_maxfwd_header("3")) {
> send_reply("483","looping");
> exit;
> }
>
>
> if ( has_totag() ) {
> # sequential request -> obey Route indication
> xlog("Loose Route for $avp(iurl) over $avp(dest) $avp(calltype) \n");
> loose_route();
>                 t_relay();
>                 exit;
>         }
>
>         # handle cancel and re-transmissions
> if ( is_method("CANCEL") ) {
> if ( t_check_trans() )
> t_relay();
> exit;
> }
>
>
>         # from now on we have only the initial requests
>         if (!is_method("INVITE")) {
>                 send_reply("405","Method Not Allowed");
>                 exit;
>         }
>
>         # initial request
> record_route();
> $avp(iurl) = $(hdr(P-Nuance-Initial-URL));
> $avp(calltype) = $(hdr(P-Nuance-Call-Type));
>         xlog("Sending call $fu $avp(iurl) $avp(calltype) \n");
> # LB function returns negative if no suitable destination (for 
> requested resources) is found,
>         # or if all destinations are full
>         if ( !load_balance("1","channel") ) {
> xlog("All Destinations are full. Sending 500! $fu 
> $(hdr(P-Nuance-Initial-URL))");
>              send_reply("500","Service full");
>              exit;
>         }
> $avp(dest) = $du;
> xlog("Call sent over $avp(dest) $fu $(hdr(P-Nuance-Initial-URL))\n");
> t_on_reply("test");
>         # arm a failure route for be able to catch a failure event and 
> to do
>         # failover to the next available destination
>         t_on_failure("LB_failed");
>
>         # send it out
> if (!t_relay()) {
> sl_reply_error();
> }
>
> }
>
> #onreply_route {
> #xlog("OpenSIPS received a reply $avp(iurl) rs= $rs | fu= $fu | si= 
> $si | Ri= $Ri ===");
> #}
>
> onreply_route[test] {
> xlog("Reply $rs received from $si for the call $avp(iurl)");
> }
>
> failure_route[LB_failed]
> {
>         # skip if call was canceled
> if (t_was_cancelled()) {
> exit;
> }
> xlog("Call $avp(iurl) sent over $avp(dest) failed with code 
> $T_reply_code\n");
> # Checking only for Destination Failure
> # If the destination is failed, disable it and reroute the call
>         #if ( t_check_status("[56][0-9][0-9]")) {
> #Comment the below line and put the above condition to probe for all 
> 500-699 errors
>         if ( t_check_status("503")) {
>        # this is a case for failover
>                 xlog("Marking $avp(dest) disabled as it failed for 
> call $avp(iurl) with code $T_reply_code\n");
>                 # mark failed destination as disabled
>                 lb_disable();
>                 # try to re-route to next available destination
>                 #if ( !load_balance("1","channel") ) {
> #  xlog("RETRY FAILED SENDING 500");
>                 #      send_reply("500","Service Full");
>                 #      exit;
>                 #}
>                 #xlog("REPORT: re-routing call to $du \n");
>                 #t_relay();
>         }
> send_reply("$T_reply_code","Call Failed");
> }
>
>
> Please do let me know if any comments..
>
> Regards,
> Chandan
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Aug 17 11:29:01 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 17 Aug 2016 12:29:01 +0300
Subject: [OpenSIPS-Users] Dispatcher command question
In-Reply-To: <1471419373119-7604038.post@n2.nabble.com>
References: <1471419373119-7604038.post@n2.nabble.com>
Message-ID: <5e6b5457-e9a5-293b-41d9-f0873017d456@opensips.org>

Hi Chen-Che,

What is the db_mode you use for the db_text module ? see :
http://www.opensips.org/html/docs/modules/2.2.x/db_text.html#id293561

If in mode 0, the db_text (as sql backend) will not read the text file 
during runtime unless you run the MI dbt_reload command.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 17.08.2016 10:36, Chen-Che Huang wrote:
> Hi all,
>
> I'm using the dispatcher module with DBTEXT. With the preconfigured
> /etc/opensips/dispatcher file, the front OpenSIPS proxy works with two
> backend SIP proxies. Now, I would like to add a new backend SIP proxy at the
> run time of the front OpenSIPS proxy. I have tried to modified the
> dispatcher file and then call 'sudo opensipsctl fifo ds_reload'. But I could
> not see the added SIP proxy by running sudo opensipsctl fifo ds_list'. I
> also try call `sudo opensipsctl dispatcher addgw'. However, I can see the
> added SIP proxy by `sudo opensipsctl dispatcher show' but not in the result
> of `sudo opensipsctl dispatcher show'. It seems that  the content of
> text-based db does not sync with the memory. I'm using opensips version
> 1.11.6. I wonder it is feasible to add a new backend proxy during the run
> time in the dispatcher module? Any comment is appreciated.
>
> Best regards,
> Chen-Che
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-command-question-tp7604038.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



From stefano.pisani at omnianet.it  Wed Aug 17 11:44:27 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Wed, 17 Aug 2016 11:44:27 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
Message-ID: <4b5c6ad8-8db6-07fb-865e-0b0093b263a2@omnianet.it>

I hope this helps.

Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
need to upgrade or downgrade to a different openssl version !!
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7ffff161b5aa/0x7ffff7672550 
realloc=0x7ffff161b5d6/0x7ffff7672c40 free=0x7ffff161b60e/0x7ffff7672a70 
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize module 
tls_mgm
Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
(gdb) info line *0x7ffff7672550
Line 2903 of "malloc.c" starts at address 0x7ffff7672550 
<__GI___libc_malloc> and ends at 0x7ffff7672556 <__GI___libc_malloc+6>.
(gdb) info line *0x7ffff7672c40
Line 2975 of "malloc.c" starts at address 0x7ffff7672c40 
<__GI___libc_realloc> and ends at 0x7ffff7672c54 <__GI___libc_realloc+20>.
(gdb) info line *0x7ffff161b5aa
Line 89 of "tls.h" starts at address 0x7ffff161b5aa <os_malloc> and ends 
at 0x7ffff161b5b6 <os_malloc+12>.
(gdb) info line *0x7ffff161b5d6
Line 95 of "tls.h" starts at address 0x7ffff161b5d6 <os_realloc> and 
ends at 0x7ffff161b5e6 <os_realloc+16>.


Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
> Can you run 'gdb ./opensips' and run the following commands:
> info line *0x7f0ce10d8a70
> info line *0x7f0ce10d8550
>
> Thanks,
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>
>> Hi R?zvan.
>>
>> These are the new logs. I hope it helps to fix this.
>>
>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>> memory allocation functions
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>> so, you need to upgrade or downgrade to a different openssl version !!
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 Mar 
>> 2016
>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>> module tls_mgm
>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules
>>
>> Thanks.
>> Stefano
>>
>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>
>>> Hi, Stefano!
>>>
>>> Apologies, I did not push the changes. Could you please re-clone and 
>>> try again?
>>>
>>> Best regards,
>>>
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>> I have this result.
>>>>
>>>> # ./opensips -V
>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> git revision: 058991d
>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>
>>>>
>>>>
>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>
>>>>> Hi, Stefano!
>>>>>
>>>>> I don't think you are using the latest sources. Either you have 
>>>>> not installed them, or you are using a wrong path. Could you 
>>>>> please double-check?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> R?zvan Crainea
>>>>> OpenSIPS Solutions
>>>>> www.opensips-solutions.com
>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>
>>>>>> I cloned the source from
>>>>>>
>>>>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head 
>>>>>> the error appears the same Aug 16 11:55:14 [25630] 
>>>>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>>>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory 
>>>>>> allocation functions Aug 16 11:55:14 [25630] 
>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>> need to upgrade or downgrade to a differen openssl version !! Aug 
>>>>>> 16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
>>>>>> module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error 
>>>>>> while initializing modules Aug 16 11:55:14 [25630] 
>>>>>> INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
>>>>>> DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
>>>>>> [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
>>>>>> [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 
>>>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 
>>>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done 
>>>>>> Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the 
>>>>>> shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
>>>>>> Exiting.... Thanks Stefano |
>>>>>>
>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>> Stefano!
>>>>>>>
>>>>>>> I've added some extra debugging in the master branch that might 
>>>>>>> help you debug this. Could you please take the latest git 
>>>>>>> version/deb and try to run again?
>>>>>>>
>>>>>>> Best regards,
>>>>>>>
>>>>>>> R?zvan Crainea
>>>>>>> OpenSIPS Solutions
>>>>>>> www.opensips-solutions.com
>>>>>>>
>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>>>>
>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>
>>>>>>>> It's not working:
>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>>>> protocol
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set 
>>>>>>>> the memory allocation functions
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if 
>>>>>>>> you have openssl 1.0.1e-fips, as this version is know to be 
>>>>>>>> broken; if so, you need to upgrade or downgrade to a differen 
>>>>>>>> openssl version !!
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>> initialize module tls_mgm
>>>>>>>>
>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>
>>>>>>>> The opensips installed packages are:
>>>>>>>> opensips install
>>>>>>>> opensips-tls-module:amd64 install
>>>>>>>> opensips-tlsmgm-module:amd64 install
>>>>>>>> opensips-wss-module:amd64 install
>>>>>>>>
>>>>>>>>
>>>>>>>> There is something I can do?
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Aug 17 11:45:24 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 17 Aug 2016 12:45:24 +0300
Subject: [OpenSIPS-Users] Load Balancer - Delay in clearing dialogs
In-Reply-To: <CADh=Fyicu3n2wPYGsNYNAEC=na2VWqFqNjmdsCcXEP=k3AxL1w@mail.gmail.com>
References: <CADh=Fyicu3n2wPYGsNYNAEC=na2VWqFqNjmdsCcXEP=k3AxL1w@mail.gmail.com>
Message-ID: <b2d41fbc-d6a7-8ffe-bf10-5c902cbc11d8@opensips.org>

Hi Chandan,

Could you check what is the status of the calls (see the "State" field) 
? I guess the call are listed in state 5 (terminated) - and this is the 
reason why you do not see them as counted by LB.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 16.08.2016 12:47, Chandan PR wrote:
> Hi Guys,
>
> Opendsips Version : 1.9.x
>
> We have configured the load balancer with 8 ports.
>
> When the calls are ongoing, the lb_list shows the load balancer is 
> running full.
>
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl 
> fifo lb_list
> Destination:: sip:192.168.248.62:5060 <http://192.168.248.62:5060> 
> id=152 group=1 enabled=yes auto-re=on
> Resource:: channel max=4 load=4
> Destination:: sip:192.168.248.64:5060 <http://192.168.248.64:5060> 
> id=153 group=1 enabled=yes auto-re=on
> Resource:: channel max=4 load=4
>
> When I check the dialog_list, I see some older dialogs which are yet 
> to be cleared.
>
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl 
> fifo dlg_list | grep "callid"
> callid:: 14713396610041471339775929_ZTLT_931
> callid:: 14713396610151471339775929_ZTLT_937
> callid:: 14713396610221471339775929_ZTLT_942
> callid:: 14713396610081471339775929_ZTLT_933
> callid:: 14713396610181471339775929_ZTLT_939
> callid:: 14713396610191471339775929_ZTLT_940
> callid:: 14713396610131471339775929_ZTLT_936
> callid:: 14713396610031471339775929_ZTLT_930
> callid:: 14713396610061471339775929_ZTLT_932
> callid:: 14713396610161471339775929_ZTLT_938
> callid:: 14713396610271471339775929_ZTLT_945
> callid:: 14713396610371471339775929_ZTLT_951
> callid:: 14713396610101471339775929_ZTLT_934
> callid:: 14713396610311471339775929_ZTLT_947
> callid:: 14713396610211471339775929_ZTLT_941
> callid:: 14713396610241471339775929_ZTLT_943
> callid:: 14713396610341471339775929_ZTLT_949
> callid:: 14713396610011471339775929_ZTLT_929
> callid:: 14713396610111471339775929_ZTLT_935
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl 
> fifo dlg_list | grep "callid" | wc -l
> 13
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin#
>
> *Even when the lb_list shows the load as 0, I see some dialogs present 
> and getting cleared after around 10-12 seconds.*
>
> What would be the impact of this delay in clearing the dialogs? Can 
> this lead to lot many calls rejected with "Destination Full" (We are 
> experiencing this issue in our production, where a huge number of 
> calls rejected)?
>
> Is there any way to reduce this delay or clear the dialogs immediately?
>
> *_Dialog Module Settings:_*
> #### DIALOG module
> loadmodule "dialog.so"
> modparam("dialog", "dlg_match_mode", 1)
> modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
> modparam("dialog", "db_mode", 2)
> modparam("dialog", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") #
>
>
> *_Load Balancer Script:_*
> #
> # $Id: opensips_loadbalancer.m4 9723 2013-02-01 15:10:50Z bogdan_iancu $
> #
> # OpenSIPS loadbalancer script
> #     by OpenSIPS Solutions <team at opensips-solutions.com 
> <mailto:team at opensips-solutions.com>>
> #
> # This script was generated via "make menuconfig", from
> #   the "Load Balancer" scenario.
> # You can enable / disable more features / functionalities by
> #   re-generating the scenario with different options.
> #
> # Please refer to the Core CookBook at:
> # http://www.opensips.org/Resources/DocsCookbooks
> # for a explanation of possible statements, functions and parameters.
> #
>
>
> ####### Global Parameters #########
>
> debug=3
> log_stderror=no
> log_facility=LOG_LOCAL1
>
> fork=yes
> children=4
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
> /* uncomment the next line to enable the auto temporary blacklisting of
>    not available destinations (default disabled) */
> #disable_dns_blacklist=no
>
> /* uncomment the next line to enable IPv6 lookup after IPv4 dns
>    lookup failures (default disabled) */
> #dns_try_ipv6=yes
>
> /* comment the next line to enable the auto discovery of local aliases
>    based on revers DNS on IPs */
> auto_aliases=no
>
>
> listen=udp:192.168.248.111:5060 <http://192.168.248.111:5060>   # 
> CUSTOMIZE ME
>
> disable_tcp=no
> listen=tcp:192.168.248.111:5060 <http://192.168.248.111:5060>   # 
> CUSTOMIZE ME
>
> disable_tls=yes
>
>
>
>
> ####### Modules Section ########
>
> #set module path
> mpath="/usr/local/opensips/lib64/opensips/modules/"
>
>
>
> #### SIGNALING module
> loadmodule "signaling.so"
>
> #### StateLess module
> loadmodule "sl.so"
>
> #### Transaction Module
> loadmodule "tm.so"
> modparam("tm", "fr_timer", 30)
> modparam("tm", "fr_inv_timer", 120)
> modparam("tm", "restart_fr_on_each_reply", 0)
> modparam("tm", "onreply_avp_mode", 1)
>
> #### Record Route Module
> loadmodule "rr.so"
> /* do not append from tag to the RR (no need for this script) */
> modparam("rr", "append_fromtag", 0)
>
> #### MAX ForWarD module
> loadmodule "maxfwd.so"
>
> #### SIP MSG OPerationS module
> loadmodule "sipmsgops.so"
>
> #### FIFO Management Interface
> loadmodule "mi_fifo.so"
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0666)
>
> #### URI module
> loadmodule "uri.so"
> modparam("uri", "use_uri_table", 0)
>
> #### MYSQL module
> loadmodule "db_mysql.so"
>
> #### AVPOPS module
> loadmodule "avpops.so"
>
> #### ACCounting module
> loadmodule "acc.so"
> /* what special events should be accounted ? */
> modparam("acc", "early_media", 0)
> modparam("acc", "report_cancels", 0)
> /* by default we do not adjust the direct of the sequential requests.
>    if you enable this parameter, be sure the enable "append_fromtag"
>    in "rr" module */
> modparam("acc", "detect_direction", 0)
> modparam("acc", "failed_transaction_flag", "ACC_FAILED")
> /* account triggers (flags) */
> modparam("acc", "log_flag", "ACC_DO")
> modparam("acc", "log_missed_flag", "ACC_MISSED")
>
>
> #### DIALOG module
> loadmodule "dialog.so"
> modparam("dialog", "dlg_match_mode", 1)
> modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
> modparam("dialog", "db_mode", 2)
> modparam("dialog", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
>
>
> #### LOAD BALANCER module
> loadmodule "load_balancer.so"
> modparam("load_balancer", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
> modparam("load_balancer", "probing_method", "OPTIONS")
>
> modparam("load_balancer", "probing_interval", 15)
>
>
>
>
>
> ####### Routing Logic ########
>
>
> # main request routing logic
> route{
> if (!mf_process_maxfwd_header("3")) {
> send_reply("483","looping");
> exit;
> }
>
>
> if ( has_totag() ) {
> # sequential request -> obey Route indication
> xlog("Loose Route for $avp(iurl) over $avp(dest) $avp(calltype) \n");
> loose_route();
>                 t_relay();
>                 exit;
>         }
>
>         # handle cancel and re-transmissions
> if ( is_method("CANCEL") ) {
> if ( t_check_trans() )
> t_relay();
> exit;
> }
>
>
>         # from now on we have only the initial requests
>         if (!is_method("INVITE")) {
>                 send_reply("405","Method Not Allowed");
>                 exit;
>         }
>
>         # initial request
> record_route();
> $avp(iurl) = $(hdr(P-Nuance-Initial-URL));
> $avp(calltype) = $(hdr(P-Nuance-Call-Type));
>         xlog("Sending call $fu $avp(iurl) $avp(calltype) \n");
> xlog("Current Dialogs count $DLG_count \n");
> # LB function returns negative if no suitable destination (for 
> requested resources) is found,
>         # or if all destinations are full
>         if ( !load_balance("1","channel") ) {
> xlog("All Destinations are full. Sending 500! $fu 
> $(hdr(P-Nuance-Initial-URL)) CURRENT DIALOGS $DLG_count \n");
>              send_reply("500","Service full");
>              exit;
>         }
> $avp(dest) = $du;
> xlog("Call sent over $avp(dest) $fu $(hdr(P-Nuance-Initial-URL))\n");
> t_on_reply("test");
>         # arm a failure route for be able to catch a failure event and 
> to do
>         # failover to the next available destination
>         t_on_failure("LB_failed");
>
>         # send it out
> if (!t_relay()) {
> sl_reply_error();
> }
>
> }
>
> #onreply_route {
> #xlog("OpenSIPS received a reply $avp(iurl) rs= $rs | fu= $fu | si= 
> $si | Ri= $Ri ===");
> #}
>
> onreply_route[test] {
> xlog("Reply $rs received from $si for the call $avp(iurl)");
> }
>
> failure_route[LB_failed]
> {
>         # skip if call was canceled
> if (t_was_cancelled()) {
> exit;
> }
> xlog("Call $avp(iurl) sent over $avp(dest) failed with code 
> $T_reply_code\n");
> # Checking only for Destination Failure
> # If the destination is failed, disable it and reroute the call
>         #if ( t_check_status("[56][0-9][0-9]")) {
>         if ( t_check_status("503")) {
>        # this is a case for failover
>                 xlog("Marking $avp(dest) disabled as it failed for 
> call $avp(iurl) with code $T_reply_code\n");
>                 #xlog("call $avp(iurl) failed with code $T_reply_code 
> from $avp(dest)\n");
> # mark failed destination as disabled
>                 lb_disable();
>                 # try to re-route to next available destination
>                 #if ( !load_balance("1","channel") ) {
> #  xlog("RETRY FAILED SENDING 500");
>                 #      send_reply("500","Service Full");
>                 #      exit;
>                 #}
>                 #xlog("REPORT: re-routing call to $du \n");
>                 #t_relay();
>         }
> send_reply("$T_reply_code","Call Failed");
> }
>
>
> Regards,
> Chandan
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Aug 17 11:47:44 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 17 Aug 2016 12:47:44 +0300
Subject: [OpenSIPS-Users] mi_xmlrpc module related query .
In-Reply-To: <CALf-Q47sAcKNZaRSrri9JyYNBJ5jB+o2qOo406P1XAszJpHLpg@mail.gmail.com>
References: <CALf-Q47sAcKNZaRSrri9JyYNBJ5jB+o2qOo406P1XAszJpHLpg@mail.gmail.com>
Message-ID: <7afeb0f9-5d73-0591-15a5-7522b5eced9e@opensips.org>

Hi Sasmita,

Please avoid using the mi_xmlrpc module and use the mi_xmlrpc_ng (with 
the httpd module). The mi_xmlrpc has some issues in its relation to the 
libxmlrpc-c3 lib (some threads versus processes issue)

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.08.2016 11:00, Sasmita Panda wrote:
> Hi All ,
>
>     I wanted to use this module in opensips-1.11 . And in the doc it 
> given that
> /"*libxmlrpc-c3*/*version 0.9.10 - 0.9.42. (using other versions may 
> be dangerous and lead to opensips blocking) *" .
>
>           I am not getting this particular  version anywhere . I have 
> tried to install "*libxmlrpc-core-c3_1.16.33-3*." version . but while 
> compiling opensips , its giving warning : " *you are using a different 
> version which is not recommended so compile in ur own risk "* .
>               After installation I am not getting the  " mi_xmlrpc.so 
> "  file in the module directory .
>
>       Can anybody tell me where will get the same version . Or is 
> there any other way to fix my problem .
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From stefano.pisani at omnianet.it  Wed Aug 17 11:48:52 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Wed, 17 Aug 2016 11:48:52 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
Message-ID: <b02afc50-49a4-5ead-3f7e-18a6f8337750@omnianet.it>

I miss the last two lines

(gdb) info line *0x7ffff7672a70
Line 2935 of "malloc.c" starts at address 0x7ffff7672a70 
<__GI___libc_free> and ends at 0x7ffff7672a7a <__GI___libc_free+10>.
(gdb) info line *0x7ffff161b60e
Line 101 of "tls.h" starts at address 0x7ffff161b60e <os_free> and ends 
at 0x7ffff161b61a <os_free+12>.

I hope this helps.

Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
need to upgrade or downgrade to a different openssl version !!
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7ffff161b5aa/0x7ffff7672550 
realloc=0x7ffff161b5d6/0x7ffff7672c40 free=0x7ffff161b60e/0x7ffff7672a70 
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize module 
tls_mgm
Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
(gdb) info line *0x7ffff7672550
Line 2903 of "malloc.c" starts at address 0x7ffff7672550 
<__GI___libc_malloc> and ends at 0x7ffff7672556 <__GI___libc_malloc+6>.
(gdb) info line *0x7ffff7672c40
Line 2975 of "malloc.c" starts at address 0x7ffff7672c40 
<__GI___libc_realloc> and ends at 0x7ffff7672c54 <__GI___libc_realloc+20>.
(gdb) info line *0x7ffff161b5aa
Line 89 of "tls.h" starts at address 0x7ffff161b5aa <os_malloc> and ends 
at 0x7ffff161b5b6 <os_malloc+12>.
(gdb) info line *0x7ffff161b5d6
Line 95 of "tls.h" starts at address 0x7ffff161b5d6 <os_realloc> and 
ends at 0x7ffff161b5e6 <os_realloc+16>.


Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
> Can you run 'gdb ./opensips' and run the following commands:
> info line *0x7f0ce10d8a70
> info line *0x7f0ce10d8550
>
> Thanks,
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>
>> Hi R?zvan.
>>
>> These are the new logs. I hope it helps to fix this.
>>
>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>> memory allocation functions
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>> so, you need to upgrade or downgrade to a different openssl version !!
>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 Mar 
>> 2016
>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>> module tls_mgm
>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules
>>
>> Thanks.
>> Stefano
>>
>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>
>>> Hi, Stefano!
>>>
>>> Apologies, I did not push the changes. Could you please re-clone and 
>>> try again?
>>>
>>> Best regards,
>>>
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>> I have this result.
>>>>
>>>> # ./opensips -V
>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> git revision: 058991d
>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>
>>>>
>>>>
>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>
>>>>> Hi, Stefano!
>>>>>
>>>>> I don't think you are using the latest sources. Either you have 
>>>>> not installed them, or you are using a wrong path. Could you 
>>>>> please double-check?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> R?zvan Crainea
>>>>> OpenSIPS Solutions
>>>>> www.opensips-solutions.com
>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>
>>>>>> I cloned the source from
>>>>>>
>>>>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head 
>>>>>> the error appears the same Aug 16 11:55:14 [25630] 
>>>>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>>>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory 
>>>>>> allocation functions Aug 16 11:55:14 [25630] 
>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>> need to upgrade or downgrade to a differen openssl version !! Aug 
>>>>>> 16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
>>>>>> module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error 
>>>>>> while initializing modules Aug 16 11:55:14 [25630] 
>>>>>> INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
>>>>>> DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
>>>>>> [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
>>>>>> [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 
>>>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 
>>>>>> 16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists 
>>>>>> Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done 
>>>>>> Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the 
>>>>>> shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
>>>>>> Exiting.... Thanks Stefano |
>>>>>>
>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>> Stefano!
>>>>>>>
>>>>>>> I've added some extra debugging in the master branch that might 
>>>>>>> help you debug this. Could you please take the latest git 
>>>>>>> version/deb and try to run again?
>>>>>>>
>>>>>>> Best regards,
>>>>>>>
>>>>>>> R?zvan Crainea
>>>>>>> OpenSIPS Solutions
>>>>>>> www.opensips-solutions.com
>>>>>>>
>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>>>>
>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>
>>>>>>>> It's not working:
>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>>>> protocol
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set 
>>>>>>>> the memory allocation functions
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if 
>>>>>>>> you have openssl 1.0.1e-fips, as this version is know to be 
>>>>>>>> broken; if so, you need to upgrade or downgrade to a differen 
>>>>>>>> openssl version !!
>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>> initialize module tls_mgm
>>>>>>>>
>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>
>>>>>>>> The opensips installed packages are:
>>>>>>>> opensips install
>>>>>>>> opensips-tls-module:amd64 install
>>>>>>>> opensips-tlsmgm-module:amd64 install
>>>>>>>> opensips-wss-module:amd64 install
>>>>>>>>
>>>>>>>>
>>>>>>>> There is something I can do?
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From chandan.pr at webshar.org  Wed Aug 17 13:00:37 2016
From: chandan.pr at webshar.org (Chandan PR)
Date: Wed, 17 Aug 2016 16:30:37 +0530
Subject: [OpenSIPS-Users] Load Balancer - Delay in clearing dialogs
	<SOLVED>
Message-ID: <CADh=FyhfZ69qH_V7VBHFu3YqWJud-BLbuRS72rm596KKWmND6w@mail.gmail.com>

Thanks for the input Bogdan.

We found that the problem was due to some calls failing to get the response.

Since we had the dialog timeout of 6 hours, which was not in sync with
application logic, we were sending more calls than available destinations.

We have updated the dialog timeout in par with the application and this
should resolve the issue for us.

Regards,
Chandan

On Wed, Aug 17, 2016 at 3:15 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Chandan,
>
> Could you check what is the status of the calls (see the "State" field) ?
> I guess the call are listed in state 5 (terminated) - and this is the
> reason why you do not see them as counted by LB.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 16.08.2016 12:47, Chandan PR wrote:
>
> Hi Guys,
>
> Opendsips Version : 1.9.x
>
> We have configured the load balancer with 8 ports.
>
> When the calls are ongoing, the lb_list shows the load balancer is running
> full.
>
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl fifo
> lb_list
> Destination:: sip:192.168.248.62:5060 id=152 group=1 enabled=yes
> auto-re=on
> Resource:: channel max=4 load=4
> Destination:: sip:192.168.248.64:5060 id=153 group=1 enabled=yes
> auto-re=on
> Resource:: channel max=4 load=4
>
> When I check the dialog_list, I see some older dialogs which are yet to be
> cleared.
>
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl fifo
> dlg_list | grep "callid"
> callid:: 14713396610041471339775929_ZTLT_931
> callid:: 14713396610151471339775929_ZTLT_937
> callid:: 14713396610221471339775929_ZTLT_942
> callid:: 14713396610081471339775929_ZTLT_933
> callid:: 14713396610181471339775929_ZTLT_939
> callid:: 14713396610191471339775929_ZTLT_940
> callid:: 14713396610131471339775929_ZTLT_936
> callid:: 14713396610031471339775929_ZTLT_930
> callid:: 14713396610061471339775929_ZTLT_932
> callid:: 14713396610161471339775929_ZTLT_938
> callid:: 14713396610271471339775929_ZTLT_945
> callid:: 14713396610371471339775929_ZTLT_951
> callid:: 14713396610101471339775929_ZTLT_934
> callid:: 14713396610311471339775929_ZTLT_947
> callid:: 14713396610211471339775929_ZTLT_941
> callid:: 14713396610241471339775929_ZTLT_943
> callid:: 14713396610341471339775929_ZTLT_949
> callid:: 14713396610011471339775929_ZTLT_929
> callid:: 14713396610111471339775929_ZTLT_935
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin# ./opensipsctl fifo
> dlg_list | grep "callid" | wc -l
> 13
> root at Dev-Chandan-Opensips-LB:/usr/local/opensips/sbin#
>
> *Even when the lb_list shows the load as 0, I see some dialogs present and
> getting cleared after around 10-12 seconds.*
>
> What would be the impact of this delay in clearing the dialogs? Can this
> lead to lot many calls rejected with "Destination Full" (We are
> experiencing this issue in our production, where a huge number of calls
> rejected)?
>
> Is there any way to reduce this delay or clear the dialogs immediately?
>
> *Dialog Module Settings:*
> #### DIALOG module
> loadmodule "dialog.so"
> modparam("dialog", "dlg_match_mode", 1)
> modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
> modparam("dialog", "db_mode", 2)
> modparam("dialog", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") #
>
>
> *Load Balancer Script:*
> #
> # $Id: opensips_loadbalancer.m4 9723 2013-02-01 15:10:50Z bogdan_iancu $
> #
> # OpenSIPS loadbalancer script
> #     by OpenSIPS Solutions <team at opensips-solutions.com>
> #
> # This script was generated via "make menuconfig", from
> #   the "Load Balancer" scenario.
> # You can enable / disable more features / functionalities by
> #   re-generating the scenario with different options.
> #
> # Please refer to the Core CookBook at:
> #      http://www.opensips.org/Resources/DocsCookbooks
> # for a explanation of possible statements, functions and parameters.
> #
>
>
> ####### Global Parameters #########
>
> debug=3
> log_stderror=no
> log_facility=LOG_LOCAL1
>
> fork=yes
> children=4
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
> /* uncomment the next line to enable the auto temporary blacklisting of
>    not available destinations (default disabled) */
> #disable_dns_blacklist=no
>
> /* uncomment the next line to enable IPv6 lookup after IPv4 dns
>    lookup failures (default disabled) */
> #dns_try_ipv6=yes
>
> /* comment the next line to enable the auto discovery of local aliases
>    based on revers DNS on IPs */
> auto_aliases=no
>
>
> listen=udp:192.168.248.111:5060   # CUSTOMIZE ME
>
> disable_tcp=no
> listen=tcp:192.168.248.111:5060   # CUSTOMIZE ME
>
> disable_tls=yes
>
>
>
>
> ####### Modules Section ########
>
> #set module path
> mpath="/usr/local/opensips/lib64/opensips/modules/"
>
>
>
> #### SIGNALING module
> loadmodule "signaling.so"
>
> #### StateLess module
> loadmodule "sl.so"
>
> #### Transaction Module
> loadmodule "tm.so"
> modparam("tm", "fr_timer", 30)
> modparam("tm", "fr_inv_timer", 120)
> modparam("tm", "restart_fr_on_each_reply", 0)
> modparam("tm", "onreply_avp_mode", 1)
>
> #### Record Route Module
> loadmodule "rr.so"
> /* do not append from tag to the RR (no need for this script) */
> modparam("rr", "append_fromtag", 0)
>
> #### MAX ForWarD module
> loadmodule "maxfwd.so"
>
> #### SIP MSG OPerationS module
> loadmodule "sipmsgops.so"
>
> #### FIFO Management Interface
> loadmodule "mi_fifo.so"
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0666)
>
> #### URI module
> loadmodule "uri.so"
> modparam("uri", "use_uri_table", 0)
>
> #### MYSQL module
> loadmodule "db_mysql.so"
>
> #### AVPOPS module
> loadmodule "avpops.so"
>
> #### ACCounting module
> loadmodule "acc.so"
> /* what special events should be accounted ? */
> modparam("acc", "early_media", 0)
> modparam("acc", "report_cancels", 0)
> /* by default we do not adjust the direct of the sequential requests.
>    if you enable this parameter, be sure the enable "append_fromtag"
>    in "rr" module */
> modparam("acc", "detect_direction", 0)
> modparam("acc", "failed_transaction_flag", "ACC_FAILED")
> /* account triggers (flags) */
> modparam("acc", "log_flag", "ACC_DO")
> modparam("acc", "log_missed_flag", "ACC_MISSED")
>
>
> #### DIALOG module
> loadmodule "dialog.so"
> modparam("dialog", "dlg_match_mode", 1)
> modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
> modparam("dialog", "db_mode", 2)
> modparam("dialog", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
>
>
> #### LOAD BALANCER module
> loadmodule "load_balancer.so"
> modparam("load_balancer", "db_url",
> "mysql://opensips:opensipsrw at localhost/opensips") # CUSTOMIZE ME
> modparam("load_balancer", "probing_method", "OPTIONS")
>
> modparam("load_balancer", "probing_interval", 15)
>
>
>
>
>
> ####### Routing Logic ########
>
>
> # main request routing logic
> route{
> if (!mf_process_maxfwd_header("3")) {
> send_reply("483","looping");
> exit;
> }
>
>
> if ( has_totag() ) {
> # sequential request -> obey Route indication
> xlog("Loose Route for $avp(iurl) over $avp(dest) $avp(calltype) \n");
> loose_route();
>                 t_relay();
>                 exit;
>         }
>
>         # handle cancel and re-transmissions
> if ( is_method("CANCEL") ) {
> if ( t_check_trans() )
> t_relay();
> exit;
> }
>
>
>         # from now on we have only the initial requests
>         if (!is_method("INVITE")) {
>                 send_reply("405","Method Not Allowed");
>                 exit;
>         }
>
>         # initial request
> record_route();
> $avp(iurl) = $(hdr(P-Nuance-Initial-URL));
> $avp(calltype) = $(hdr(P-Nuance-Call-Type));
>         xlog("Sending call $fu $avp(iurl) $avp(calltype) \n");
> xlog("Current Dialogs count $DLG_count \n");
> # LB function returns negative if no suitable destination (for requested
> resources) is found,
>         # or if all destinations are full
>         if ( !load_balance("1","channel") ) {
>     xlog("All Destinations are full. Sending 500! $fu
> $(hdr(P-Nuance-Initial-URL)) CURRENT DIALOGS $DLG_count \n");
>              send_reply("500","Service full");
>              exit;
>         }
> $avp(dest) = $du;
> xlog("Call sent over $avp(dest) $fu $(hdr(P-Nuance-Initial-URL))\n");
> t_on_reply("test");
>         # arm a failure route for be able to catch a failure event and to
> do
>         # failover to the next available destination
>         t_on_failure("LB_failed");
>
>         # send it out
> if (!t_relay()) {
> sl_reply_error();
> }
>
> }
>
> #onreply_route {
> # xlog("OpenSIPS received a reply $avp(iurl) rs= $rs | fu= $fu | si= $si
> | Ri= $Ri ===");
> #}
>
> onreply_route[test] {
> xlog("Reply $rs received from $si for the call $avp(iurl)");
> }
>
> failure_route[LB_failed]
> {
>         # skip if call was canceled
> if (t_was_cancelled()) {
> exit;
> }
> xlog("Call $avp(iurl) sent over $avp(dest) failed with code
> $T_reply_code\n");
>
> # Checking only for Destination Failure
> # If the destination is failed, disable it and reroute the call
>         #if ( t_check_status("[56][0-9][0-9]")) {
>         if ( t_check_status("503")) {
>        # this is a case for failover
>                 xlog("Marking $avp(dest) disabled as it failed for call
> $avp(iurl) with code $T_reply_code\n");
>                 #xlog("call $avp(iurl) failed with code $T_reply_code from
> $avp(dest)\n");
> # mark failed destination as disabled
>                 lb_disable();
>                 # try to re-route to next available destination
>                 #if ( !load_balance("1","channel") ) {
> #      xlog("RETRY FAILED SENDING 500");
>                 #      send_reply("500","Service Full");
>                 #      exit;
>                 #}
>                 #xlog("REPORT: re-routing call to $du \n");
>                 #t_relay();
>         }
> send_reply("$T_reply_code","Call Failed");
> }
>
>
> Regards,
> Chandan
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From spanda at 3clogic.com  Wed Aug 17 13:42:19 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 17 Aug 2016 17:12:19 +0530
Subject: [OpenSIPS-Users] mi_xmlrpc module related query .
In-Reply-To: <7afeb0f9-5d73-0591-15a5-7522b5eced9e@opensips.org>
References: <CALf-Q47sAcKNZaRSrri9JyYNBJ5jB+o2qOo406P1XAszJpHLpg@mail.gmail.com>
	<7afeb0f9-5d73-0591-15a5-7522b5eced9e@opensips.org>
Message-ID: <CALf-Q47xwqa7jJQC7qq3m2pYM3PfPH2GMzCFy6qbUvJSJFD_-Q@mail.gmail.com>

Thank you so much Bogdan . Its really helpful .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Wed, Aug 17, 2016 at 3:17 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Sasmita,
>
> Please avoid using the mi_xmlrpc module and use the mi_xmlrpc_ng (with the
> httpd module). The mi_xmlrpc has some issues in its relation to the
> libxmlrpc-c3 lib (some threads versus processes issue)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 11.08.2016 11:00, Sasmita Panda wrote:
>
> Hi All ,
>
>     I wanted to use this module in opensips-1.11 . And in the doc it given
> that
> *"libxmlrpc-c3** version 0.9.10 - 0.9.42. (using other versions may be
> dangerous and lead to opensips blocking)  *" .
>
>           I am not getting this particular  version anywhere . I have
> tried to install "*libxmlrpc-core-c3_1.16.33-3*." version . but while
> compiling opensips , its giving warning : " *you are using a different
> version which is not recommended so compile in ur own risk "* .
>               After installation I am not getting the  " mi_xmlrpc.so "
>  file in the module directory .
>
>       Can anybody tell me where will get the same version . Or is there
> any other way to fix my problem .
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From razvan at opensips.org  Wed Aug 17 14:52:28 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 17 Aug 2016 15:52:28 +0300
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <b02afc50-49a4-5ead-3f7e-18a6f8337750@omnianet.it>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
	<b02afc50-49a4-5ead-3f7e-18a6f8337750@omnianet.it>
Message-ID: <3b2cfe7f-3bb7-ca3f-cd5e-ac4e5cb78ff0@opensips.org>

Seems to be a similar problem to the 1.0.1e-fips library. Could you try 
to upgrade the openssl package?

Thanks,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/17/2016 12:48 PM, Stefano Pisani wrote:
>
> I miss the last two lines
>
> (gdb) info line *0x7ffff7672a70
> Line 2935 of "malloc.c" starts at address 0x7ffff7672a70 
> <__GI___libc_free> and ends at 0x7ffff7672a7a <__GI___libc_free+10>.
> (gdb) info line *0x7ffff161b60e
> Line 101 of "tls.h" starts at address 0x7ffff161b60e <os_free> and 
> ends at 0x7ffff161b61a <os_free+12>.
>
> I hope this helps.
>
> Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the 
> memory allocation functions
> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you 
> have openssl 1.0.1e-fips, as this version is known to be broken; if 
> so, you need to upgrade or downgrade to a different openssl version !!
> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
> malloc=0x7ffff161b5aa/0x7ffff7672550 
> realloc=0x7ffff161b5d6/0x7ffff7672c40 
> free=0x7ffff161b60e/0x7ffff7672a70 version=OpenSSL 1.0.2g-fips 1 Mar 2016
> Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize 
> module tls_mgm
> Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
> (gdb) info line *0x7ffff7672550
> Line 2903 of "malloc.c" starts at address 0x7ffff7672550 
> <__GI___libc_malloc> and ends at 0x7ffff7672556 <__GI___libc_malloc+6>.
> (gdb) info line *0x7ffff7672c40
> Line 2975 of "malloc.c" starts at address 0x7ffff7672c40 
> <__GI___libc_realloc> and ends at 0x7ffff7672c54 <__GI___libc_realloc+20>.
> (gdb) info line *0x7ffff161b5aa
> Line 89 of "tls.h" starts at address 0x7ffff161b5aa <os_malloc> and 
> ends at 0x7ffff161b5b6 <os_malloc+12>.
> (gdb) info line *0x7ffff161b5d6
> Line 95 of "tls.h" starts at address 0x7ffff161b5d6 <os_realloc> and 
> ends at 0x7ffff161b5e6 <os_realloc+16>.
>
>
> Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
>> Can you run 'gdb ./opensips' and run the following commands:
>> info line *0x7f0ce10d8a70
>> info line *0x7f0ce10d8550
>>
>> Thanks,
>>
>> R?zvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>>
>>> Hi R?zvan.
>>>
>>> These are the new logs. I hope it helps to fix this.
>>>
>>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>>> memory allocation functions
>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>>> so, you need to upgrade or downgrade to a different openssl version !!
>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 
>>> Mar 2016
>>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>>> module tls_mgm
>>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules
>>>
>>> Thanks.
>>> Stefano
>>>
>>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>>
>>>> Hi, Stefano!
>>>>
>>>> Apologies, I did not push the changes. Could you please re-clone 
>>>> and try again?
>>>>
>>>> Best regards,
>>>>
>>>> R?zvan Crainea
>>>> OpenSIPS Solutions
>>>> www.opensips-solutions.com
>>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>>> I have this result.
>>>>>
>>>>> # ./opensips -V
>>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>> git revision: 058991d
>>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>>
>>>>>
>>>>>
>>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>>
>>>>>> Hi, Stefano!
>>>>>>
>>>>>> I don't think you are using the latest sources. Either you have 
>>>>>> not installed them, or you are using a wrong path. Could you 
>>>>>> please double-check?
>>>>>>
>>>>>> Best regards,
>>>>>>
>>>>>> R?zvan Crainea
>>>>>> OpenSIPS Solutions
>>>>>> www.opensips-solutions.com
>>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>>
>>>>>>> I cloned the source from
>>>>>>>
>>>>>>> |git clone https://github.com/OpenSIPS/opensips.git opensips_head 
>>>>>>> the error appears the same Aug 16 11:55:14 [25630] 
>>>>>>> INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
>>>>>>> [25630] ERROR:tls_mgm:mod_init: unable to set the memory 
>>>>>>> allocation functions Aug 16 11:55:14 [25630] 
>>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>>> need to upgrade or downgrade to a differen openssl version !! 
>>>>>>> Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to 
>>>>>>> initialize module tls_mgm Aug 16 11:55:14 [25630] 
>>>>>>> ERROR:core:main: error while initializing modules Aug 16 
>>>>>>> 11:55:14 [25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 
>>>>>>> [25630] DBG:dispatcher:destroy: destroying module ... Aug 16 
>>>>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 
>>>>>>> 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying DELETE list 
>>>>>>> for set 0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying 
>>>>>>> hash table Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing 
>>>>>>> timers Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing 
>>>>>>> semaphores Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>>>>>>> destroying callback lists Aug 16 11:55:14 [25630] 
>>>>>>> DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 11:55:14 [25630] 
>>>>>>> DBG:core:shm_mem_destroy: destroying the shared memory lock Aug 
>>>>>>> 16 11:55:14 [25630] NOTICE:core:main: Exiting.... Thanks Stefano |
>>>>>>>
>>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>>> Stefano!
>>>>>>>>
>>>>>>>> I've added some extra debugging in the master branch that might 
>>>>>>>> help you debug this. Could you please take the latest git 
>>>>>>>> version/deb and try to run again?
>>>>>>>>
>>>>>>>> Best regards,
>>>>>>>>
>>>>>>>> R?zvan Crainea
>>>>>>>> OpenSIPS Solutions
>>>>>>>> www.opensips-solutions.com
>>>>>>>>
>>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS
>>>>>>>>>
>>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>>
>>>>>>>>> It's not working:
>>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>>>>> protocol
>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set 
>>>>>>>>> the memory allocation functions
>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if 
>>>>>>>>> you have openssl 1.0.1e-fips, as this version is know to be 
>>>>>>>>> broken; if so, you need to upgrade or downgrade to a differen 
>>>>>>>>> openssl version !!
>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>>> initialize module tls_mgm
>>>>>>>>>
>>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>>
>>>>>>>>> The opensips installed packages are:
>>>>>>>>> opensips install
>>>>>>>>> opensips-tls-module:amd64 install
>>>>>>>>> opensips-tlsmgm-module:amd64 install
>>>>>>>>> opensips-wss-module:amd64 install
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> There is something I can do?
>>>>>>>>> Thanks
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Users mailing list
>>>>>>>>> Users at lists.opensips.org
>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From stefano.pisani at omnianet.it  Wed Aug 17 15:07:58 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Wed, 17 Aug 2016 15:07:58 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <3b2cfe7f-3bb7-ca3f-cd5e-ac4e5cb78ff0@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
	<b02afc50-49a4-5ead-3f7e-18a6f8337750@omnianet.it>
	<3b2cfe7f-3bb7-ca3f-cd5e-ac4e5cb78ff0@opensips.org>
Message-ID: <84872442-15f4-d607-c15a-fc053a6a64fc@omnianet.it>

It seems to be the last available version for ubuntu 16.04 LTS. What can 
I do?

#apt-get install openssl
Reading package lists... Done
Building dependency tree
Reading state information... Done
openssl is already the newest version (1.0.2g-1ubuntu4.1)

Thanks
Stefano




Il 17/08/2016 14:52, R?zvan Crainea ha scritto:
> Seems to be a similar problem to the 1.0.1e-fips library. Could you 
> try to upgrade the openssl package?
>
> Thanks,
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/17/2016 12:48 PM, Stefano Pisani wrote:
>>
>> I miss the last two lines
>>
>> (gdb) info line *0x7ffff7672a70
>> Line 2935 of "malloc.c" starts at address 0x7ffff7672a70 
>> <__GI___libc_free> and ends at 0x7ffff7672a7a <__GI___libc_free+10>.
>> (gdb) info line *0x7ffff161b60e
>> Line 101 of "tls.h" starts at address 0x7ffff161b60e <os_free> and 
>> ends at 0x7ffff161b61a <os_free+12>.
>>
>> I hope this helps.
>>
>> Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the 
>> memory allocation functions
>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you 
>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>> so, you need to upgrade or downgrade to a different openssl version !!
>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
>> malloc=0x7ffff161b5aa/0x7ffff7672550 
>> realloc=0x7ffff161b5d6/0x7ffff7672c40 
>> free=0x7ffff161b60e/0x7ffff7672a70 version=OpenSSL 1.0.2g-fips  1 Mar 
>> 2016
>> Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize 
>> module tls_mgm
>> Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
>> (gdb) info line *0x7ffff7672550
>> Line 2903 of "malloc.c" starts at address 0x7ffff7672550 
>> <__GI___libc_malloc> and ends at 0x7ffff7672556 <__GI___libc_malloc+6>.
>> (gdb) info line *0x7ffff7672c40
>> Line 2975 of "malloc.c" starts at address 0x7ffff7672c40 
>> <__GI___libc_realloc> and ends at 0x7ffff7672c54 
>> <__GI___libc_realloc+20>.
>> (gdb) info line *0x7ffff161b5aa
>> Line 89 of "tls.h" starts at address 0x7ffff161b5aa <os_malloc> and 
>> ends at 0x7ffff161b5b6 <os_malloc+12>.
>> (gdb) info line *0x7ffff161b5d6
>> Line 95 of "tls.h" starts at address 0x7ffff161b5d6 <os_realloc> and 
>> ends at 0x7ffff161b5e6 <os_realloc+16>.
>>
>>
>> Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
>>> Can you run 'gdb ./opensips' and run the following commands:
>>> info line *0x7f0ce10d8a70
>>> info line *0x7f0ce10d8550
>>>
>>> Thanks,
>>>
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>>>
>>>> Hi R?zvan.
>>>>
>>>> These are the new logs. I hope it helps to fix this.
>>>>
>>>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>>>> memory allocation functions
>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>>>> so, you need to upgrade or downgrade to a different openssl version !!
>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>>>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>>>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>>>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 
>>>> Mar 2016
>>>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>>>> module tls_mgm
>>>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing 
>>>> modules
>>>>
>>>> Thanks.
>>>> Stefano
>>>>
>>>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>>>
>>>>> Hi, Stefano!
>>>>>
>>>>> Apologies, I did not push the changes. Could you please re-clone 
>>>>> and try again?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> R?zvan Crainea
>>>>> OpenSIPS Solutions
>>>>> www.opensips-solutions.com
>>>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>>>> I have this result.
>>>>>>
>>>>>> # ./opensips -V
>>>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>>>>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 
>>>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>> git revision: 058991d
>>>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>>>
>>>>>>
>>>>>>
>>>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>>>
>>>>>>> Hi, Stefano!
>>>>>>>
>>>>>>> I don't think you are using the latest sources. Either you have 
>>>>>>> not installed them, or you are using a wrong path. Could you 
>>>>>>> please double-check?
>>>>>>>
>>>>>>> Best regards,
>>>>>>>
>>>>>>> R?zvan Crainea
>>>>>>> OpenSIPS Solutions
>>>>>>> www.opensips-solutions.com
>>>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>>>
>>>>>>>> I cloned the source from
>>>>>>>>
>>>>>>>> |git clone https://github.com/OpenSIPS/opensips.git 
>>>>>>>> opensips_head the error appears the same Aug 16 11:55:14 
>>>>>>>> [25630] INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 
>>>>>>>> 11:55:14 [25630] ERROR:tls_mgm:mod_init: unable to set the 
>>>>>>>> memory allocation functions Aug 16 11:55:14 [25630] 
>>>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>>>> need to upgrade or downgrade to a differen openssl version !! 
>>>>>>>> Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to 
>>>>>>>> initialize module tls_mgm Aug 16 11:55:14 [25630] 
>>>>>>>> ERROR:core:main: error while initializing modules Aug 16 
>>>>>>>> 11:55:14 [25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 
>>>>>>>> [25630] DBG:dispatcher:destroy: destroying module ... Aug 16 
>>>>>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 
>>>>>>>> 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying DELETE 
>>>>>>>> list for set 0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>>>>>>>> emptying hash table Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>>>>>>>> releasing timers Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>>>>>>>> removing semaphores Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>>>>>>>> destroying callback lists Aug 16 11:55:14 [25630] 
>>>>>>>> DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 11:55:14 [25630] 
>>>>>>>> DBG:core:shm_mem_destroy: destroying the shared memory lock Aug 
>>>>>>>> 16 11:55:14 [25630] NOTICE:core:main: Exiting.... Thanks Stefano |
>>>>>>>>
>>>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>>>> Stefano!
>>>>>>>>>
>>>>>>>>> I've added some extra debugging in the master branch that 
>>>>>>>>> might help you debug this. Could you please take the latest 
>>>>>>>>> git version/deb and try to run again?
>>>>>>>>>
>>>>>>>>> Best regards,
>>>>>>>>>
>>>>>>>>> R?zvan Crainea
>>>>>>>>> OpenSIPS Solutions
>>>>>>>>> www.opensips-solutions.com
>>>>>>>>>
>>>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 
>>>>>>>>>> LTS
>>>>>>>>>>
>>>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>>>
>>>>>>>>>> It's not working:
>>>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing 
>>>>>>>>>> TLS protocol
>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set 
>>>>>>>>>> the memory allocation functions
>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if 
>>>>>>>>>> you have openssl 1.0.1e-fips, as this version is know to be 
>>>>>>>>>> broken; if so, you need to upgrade or downgrade to a differen 
>>>>>>>>>> openssl version !!
>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>>>> initialize module tls_mgm
>>>>>>>>>>
>>>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>>>
>>>>>>>>>> The opensips installed packages are:
>>>>>>>>>> opensips install
>>>>>>>>>> opensips-tls-module:amd64 install
>>>>>>>>>> opensips-tlsmgm-module:amd64 install
>>>>>>>>>> opensips-wss-module:amd64 install
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> There is something I can do?
>>>>>>>>>> Thanks
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Users mailing list
>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Users mailing list
>>>>>>>>> Users at lists.opensips.org
>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From acmicrox at gmail.com  Wed Aug 17 16:03:37 2016
From: acmicrox at gmail.com (Chen-Che Huang)
Date: Wed, 17 Aug 2016 07:03:37 -0700 (MST)
Subject: [OpenSIPS-Users] Dispatcher command question
In-Reply-To: <5e6b5457-e9a5-293b-41d9-f0873017d456@opensips.org>
References: <1471419373119-7604038.post@n2.nabble.com>
	<5e6b5457-e9a5-293b-41d9-f0873017d456@opensips.org>
Message-ID: <1471442617130-7604055.post@n2.nabble.com>

Hi Bogdan-Andrei,

As you point out, the reason is that I set the db_mode to 0. When the
db_mode changes to 1, the ds_reload command works as I expect. I just
focused on the dispatcher module and overlooked the db_text module. Thanks
for your reply.

Best regards,
Chen-Che



--
View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-command-question-tp7604038p7604055.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.


From razvan at opensips.org  Wed Aug 17 16:09:46 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 17 Aug 2016 17:09:46 +0300
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <84872442-15f4-d607-c15a-fc053a6a64fc@omnianet.it>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
	<b02afc50-49a4-5ead-3f7e-18a6f8337750@omnianet.it>
	<3b2cfe7f-3bb7-ca3f-cd5e-ac4e5cb78ff0@opensips.org>
	<84872442-15f4-d607-c15a-fc053a6a64fc@omnianet.it>
Message-ID: <d7e14137-afb6-aeea-9250-f2c34c32cd31@opensips.org>

Can you try to downgrade it[1]? At least just for testing purposes.
BTW, I am using 1.0.2h and it works fine.

[1] 
http://askubuntu.com/questions/292314/how-to-downgrade-packages-on-ubuntu

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/17/2016 04:07 PM, Stefano Pisani wrote:
> It seems to be the last available version for ubuntu 16.04 LTS. What 
> can I do?
>
> #apt-get install openssl
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> openssl is already the newest version (1.0.2g-1ubuntu4.1)
>
> Thanks
> Stefano
>
>
>
>
> Il 17/08/2016 14:52, R?zvan Crainea ha scritto:
>> Seems to be a similar problem to the 1.0.1e-fips library. Could you 
>> try to upgrade the openssl package?
>>
>> Thanks,
>> R?zvan Crainea
>> OpenSIPS Solutions
>> www.opensips-solutions.com
>> On 08/17/2016 12:48 PM, Stefano Pisani wrote:
>>>
>>> I miss the last two lines
>>>
>>> (gdb) info line *0x7ffff7672a70
>>> Line 2935 of "malloc.c" starts at address 0x7ffff7672a70 
>>> <__GI___libc_free> and ends at 0x7ffff7672a7a <__GI___libc_free+10>.
>>> (gdb) info line *0x7ffff161b60e
>>> Line 101 of "tls.h" starts at address 0x7ffff161b60e <os_free> and 
>>> ends at 0x7ffff161b61a <os_free+12>.
>>>
>>> I hope this helps.
>>>
>>> Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
>>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the 
>>> memory allocation functions
>>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>>> so, you need to upgrade or downgrade to a different openssl version !!
>>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
>>> malloc=0x7ffff161b5aa/0x7ffff7672550 
>>> realloc=0x7ffff161b5d6/0x7ffff7672c40 
>>> free=0x7ffff161b60e/0x7ffff7672a70 version=OpenSSL 1.0.2g-fips  1 
>>> Mar 2016
>>> Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize 
>>> module tls_mgm
>>> Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing 
>>> modules
>>> (gdb) info line *0x7ffff7672550
>>> Line 2903 of "malloc.c" starts at address 0x7ffff7672550 
>>> <__GI___libc_malloc> and ends at 0x7ffff7672556 <__GI___libc_malloc+6>.
>>> (gdb) info line *0x7ffff7672c40
>>> Line 2975 of "malloc.c" starts at address 0x7ffff7672c40 
>>> <__GI___libc_realloc> and ends at 0x7ffff7672c54 
>>> <__GI___libc_realloc+20>.
>>> (gdb) info line *0x7ffff161b5aa
>>> Line 89 of "tls.h" starts at address 0x7ffff161b5aa <os_malloc> and 
>>> ends at 0x7ffff161b5b6 <os_malloc+12>.
>>> (gdb) info line *0x7ffff161b5d6
>>> Line 95 of "tls.h" starts at address 0x7ffff161b5d6 <os_realloc> and 
>>> ends at 0x7ffff161b5e6 <os_realloc+16>.
>>>
>>>
>>> Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
>>>> Can you run 'gdb ./opensips' and run the following commands:
>>>> info line *0x7f0ce10d8a70
>>>> info line *0x7f0ce10d8550
>>>>
>>>> Thanks,
>>>>
>>>> R?zvan Crainea
>>>> OpenSIPS Solutions
>>>> www.opensips-solutions.com
>>>> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>>>>
>>>>> Hi R?zvan.
>>>>>
>>>>> These are the new logs. I hope it helps to fix this.
>>>>>
>>>>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS 
>>>>> protocol
>>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>>>>> memory allocation functions
>>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>>> have openssl 1.0.1e-fips, as this version is known to be broken; 
>>>>> if so, you need to upgrade or downgrade to a different openssl 
>>>>> version !!
>>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>>>>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>>>>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>>>>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 
>>>>> Mar 2016
>>>>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>>>>> module tls_mgm
>>>>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing 
>>>>> modules
>>>>>
>>>>> Thanks.
>>>>> Stefano
>>>>>
>>>>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>>>>
>>>>>> Hi, Stefano!
>>>>>>
>>>>>> Apologies, I did not push the changes. Could you please re-clone 
>>>>>> and try again?
>>>>>>
>>>>>> Best regards,
>>>>>>
>>>>>> R?zvan Crainea
>>>>>> OpenSIPS Solutions
>>>>>> www.opensips-solutions.com
>>>>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>>>>> I have this result.
>>>>>>>
>>>>>>> # ./opensips -V
>>>>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>> PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>> git revision: 058991d
>>>>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>>>>
>>>>>>>> Hi, Stefano!
>>>>>>>>
>>>>>>>> I don't think you are using the latest sources. Either you have 
>>>>>>>> not installed them, or you are using a wrong path. Could you 
>>>>>>>> please double-check?
>>>>>>>>
>>>>>>>> Best regards,
>>>>>>>>
>>>>>>>> R?zvan Crainea
>>>>>>>> OpenSIPS Solutions
>>>>>>>> www.opensips-solutions.com
>>>>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>>>>
>>>>>>>>> I cloned the source from
>>>>>>>>>
>>>>>>>>> |git clone https://github.com/OpenSIPS/opensips.git 
>>>>>>>>> opensips_head the error appears the same Aug 16 11:55:14 
>>>>>>>>> [25630] INFO:tls_mgm:mod_init: initializing TLS protocol Aug 
>>>>>>>>> 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: unable to set the 
>>>>>>>>> memory allocation functions Aug 16 11:55:14 [25630] 
>>>>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>>>>> need to upgrade or downgrade to a differen openssl version !! 
>>>>>>>>> Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to 
>>>>>>>>> initialize module tls_mgm Aug 16 11:55:14 [25630] 
>>>>>>>>> ERROR:core:main: error while initializing modules Aug 16 
>>>>>>>>> 11:55:14 [25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 
>>>>>>>>> [25630] DBG:dispatcher:destroy: destroying module ... Aug 16 
>>>>>>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 
>>>>>>>>> 16 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying DELETE 
>>>>>>>>> list for set 0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
>>>>>>>>> emptying hash table Aug 16 11:55:14 [25630] 
>>>>>>>>> DBG:tm:tm_shutdown: releasing timers Aug 16 11:55:14 [25630] 
>>>>>>>>> DBG:tm:tm_shutdown: removing semaphores Aug 16 11:55:14 
>>>>>>>>> [25630] DBG:tm:tm_shutdown: destroying callback lists Aug 16 
>>>>>>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 
>>>>>>>>> 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the 
>>>>>>>>> shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
>>>>>>>>> Exiting.... Thanks Stefano |
>>>>>>>>>
>>>>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>>>>> Stefano!
>>>>>>>>>>
>>>>>>>>>> I've added some extra debugging in the master branch that 
>>>>>>>>>> might help you debug this. Could you please take the latest 
>>>>>>>>>> git version/deb and try to run again?
>>>>>>>>>>
>>>>>>>>>> Best regards,
>>>>>>>>>>
>>>>>>>>>> R?zvan Crainea
>>>>>>>>>> OpenSIPS Solutions
>>>>>>>>>> www.opensips-solutions.com
>>>>>>>>>>
>>>>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 
>>>>>>>>>>> 16.04.1 LTS
>>>>>>>>>>>
>>>>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, 
>>>>>>>>>>> select.
>>>>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>>>>
>>>>>>>>>>> It's not working:
>>>>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing 
>>>>>>>>>>> TLS protocol
>>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set 
>>>>>>>>>>> the memory allocation functions
>>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check 
>>>>>>>>>>> if you have openssl 1.0.1e-fips, as this version is know to 
>>>>>>>>>>> be broken; if so, you need to upgrade or downgrade to a 
>>>>>>>>>>> differen openssl version !!
>>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>>>>> initialize module tls_mgm
>>>>>>>>>>>
>>>>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>>>>
>>>>>>>>>>> The opensips installed packages are:
>>>>>>>>>>> opensips                                        install
>>>>>>>>>>> opensips-tls-module:amd64                       install
>>>>>>>>>>> opensips-tlsmgm-module:amd64                    install
>>>>>>>>>>> opensips-wss-module:amd64                       install
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> There is something I can do?
>>>>>>>>>>> Thanks
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Users mailing list
>>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Users mailing list
>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Users mailing list
>>>>>>>>> Users at lists.opensips.org
>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From stefano.pisani at omnianet.it  Wed Aug 17 18:51:25 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Wed, 17 Aug 2016 18:51:25 +0200
Subject: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the
 memory allocation functions
In-Reply-To: <d7e14137-afb6-aeea-9250-f2c34c32cd31@opensips.org>
References: <CADh=FygH3KzAdMsUq3FsLw-rZq3Ebwu+kjkr7P4LDD9aR05pVg@mail.gmail.com>
	<d25ce6fc-dec1-b084-4ad6-441af26bfdbc@omnianet.it>
	<98226c69-8774-cb13-95fd-e6ffa4d145d3@opensips.org>
	<4fb2f0d1-7a59-68b3-01b8-5f8ed6a9f340@omnianet.it>
	<b986979f-e86a-d5ce-94e4-25727dc195f7@opensips.org>
	<8a6e188f-251c-d214-72c6-070cb4c7b80d@omnianet.it>
	<179242c7-b202-e2ba-a767-ecc4435189e9@opensips.org>
	<9cf5fcf7-60fa-28e5-60ac-194667997186@omnianet.it>
	<1e01a826-628a-f094-aa25-56a22c2c7649@opensips.org>
	<b02afc50-49a4-5ead-3f7e-18a6f8337750@omnianet.it>
	<3b2cfe7f-3bb7-ca3f-cd5e-ac4e5cb78ff0@opensips.org>
	<84872442-15f4-d607-c15a-fc053a6a64fc@omnianet.it>
	<d7e14137-afb6-aeea-9250-f2c34c32cd31@opensips.org>
Message-ID: <3ca102c5-bc8e-1522-1222-0f9aadcebdf6@omnianet.it>

I upgraded openssl version to 1.0.2h.

The error now is different.

Aug 17 12:46:03 [5229] ERROR:core:sr_load_module: could not open module 
</usr/lib/x86_64-linux-gnu/opensips/modules/tls_mgm.so>: /lib/x86_64-l 
inux-gnu/libssl.so.1.0.0: undefined symbol: EVP_idea_cbc

It looks like it miss a cipher.

Stefano


Il 17/08/2016 16:09, R?zvan Crainea ha scritto:
> Can you try to downgrade it[1]? At least just for testing purposes.
> BTW, I am using 1.0.2h and it works fine.
>
> [1] 
> http://askubuntu.com/questions/292314/how-to-downgrade-packages-on-ubuntu
>
> Best regards,
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 08/17/2016 04:07 PM, Stefano Pisani wrote:
>> It seems to be the last available version for ubuntu 16.04 LTS. What 
>> can I do?
>>
>> #apt-get install openssl
>> Reading package lists... Done
>> Building dependency tree
>> Reading state information... Done
>> openssl is already the newest version (1.0.2g-1ubuntu4.1)
>>
>> Thanks
>> Stefano
>>
>>
>>
>>
>> Il 17/08/2016 14:52, R?zvan Crainea ha scritto:
>>> Seems to be a similar problem to the 1.0.1e-fips library. Could you 
>>> try to upgrade the openssl package?
>>>
>>> Thanks,
>>> R?zvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 08/17/2016 12:48 PM, Stefano Pisani wrote:
>>>>
>>>> I miss the last two lines
>>>>
>>>> (gdb) info line *0x7ffff7672a70
>>>> Line 2935 of "malloc.c" starts at address 0x7ffff7672a70 
>>>> <__GI___libc_free> and ends at 0x7ffff7672a7a <__GI___libc_free+10>.
>>>> (gdb) info line *0x7ffff161b60e
>>>> Line 101 of "tls.h" starts at address 0x7ffff161b60e <os_free> and 
>>>> ends at 0x7ffff161b61a <os_free+12>.
>>>>
>>>> I hope this helps.
>>>>
>>>> Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS 
>>>> protocol
>>>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the 
>>>> memory allocation functions
>>>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>> have openssl 1.0.1e-fips, as this version is known to be broken; if 
>>>> so, you need to upgrade or downgrade to a different openssl version !!
>>>> Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
>>>> malloc=0x7ffff161b5aa/0x7ffff7672550 
>>>> realloc=0x7ffff161b5d6/0x7ffff7672c40 
>>>> free=0x7ffff161b60e/0x7ffff7672a70 version=OpenSSL 1.0.2g-fips  1 
>>>> Mar 2016
>>>> Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize 
>>>> module tls_mgm
>>>> Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing 
>>>> modules
>>>> (gdb) info line *0x7ffff7672550
>>>> Line 2903 of "malloc.c" starts at address 0x7ffff7672550 
>>>> <__GI___libc_malloc> and ends at 0x7ffff7672556 <__GI___libc_malloc+6>.
>>>> (gdb) info line *0x7ffff7672c40
>>>> Line 2975 of "malloc.c" starts at address 0x7ffff7672c40 
>>>> <__GI___libc_realloc> and ends at 0x7ffff7672c54 
>>>> <__GI___libc_realloc+20>.
>>>> (gdb) info line *0x7ffff161b5aa
>>>> Line 89 of "tls.h" starts at address 0x7ffff161b5aa <os_malloc> and 
>>>> ends at 0x7ffff161b5b6 <os_malloc+12>.
>>>> (gdb) info line *0x7ffff161b5d6
>>>> Line 95 of "tls.h" starts at address 0x7ffff161b5d6 <os_realloc> 
>>>> and ends at 0x7ffff161b5e6 <os_realloc+16>.
>>>>
>>>>
>>>> Il 17/08/2016 11:16, R?zvan Crainea ha scritto:
>>>>> Can you run 'gdb ./opensips' and run the following commands:
>>>>> info line *0x7f0ce10d8a70
>>>>> info line *0x7f0ce10d8550
>>>>>
>>>>> Thanks,
>>>>>
>>>>> R?zvan Crainea
>>>>> OpenSIPS Solutions
>>>>> www.opensips-solutions.com
>>>>> On 08/17/2016 12:06 PM, Stefano Pisani wrote:
>>>>>>
>>>>>> Hi R?zvan.
>>>>>>
>>>>>> These are the new logs. I hope it helps to fix this.
>>>>>>
>>>>>> Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS 
>>>>>> protocol
>>>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
>>>>>> memory allocation functions
>>>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
>>>>>> have openssl 1.0.1e-fips, as this version is known to be broken; 
>>>>>> if so, you need to upgrade or downgrade to a different openssl 
>>>>>> version !!
>>>>>> Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
>>>>>> malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
>>>>>> realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
>>>>>> free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 
>>>>>> Mar 2016
>>>>>> Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
>>>>>> module tls_mgm
>>>>>> Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing 
>>>>>> modules
>>>>>>
>>>>>> Thanks.
>>>>>> Stefano
>>>>>>
>>>>>> Il 17/08/2016 09:53, R?zvan Crainea ha scritto:
>>>>>>>
>>>>>>> Hi, Stefano!
>>>>>>>
>>>>>>> Apologies, I did not push the changes. Could you please re-clone 
>>>>>>> and try again?
>>>>>>>
>>>>>>> Best regards,
>>>>>>>
>>>>>>> R?zvan Crainea
>>>>>>> OpenSIPS Solutions
>>>>>>> www.opensips-solutions.com
>>>>>>> On 08/16/2016 07:26 PM, Stefano Pisani wrote:
>>>>>>>> I have this result.
>>>>>>>>
>>>>>>>> # ./opensips -V
>>>>>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>> PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>>>>> git revision: 058991d
>>>>>>>> main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Il 16/08/2016 18:23, R?zvan Crainea ha scritto:
>>>>>>>>>
>>>>>>>>> Hi, Stefano!
>>>>>>>>>
>>>>>>>>> I don't think you are using the latest sources. Either you 
>>>>>>>>> have not installed them, or you are using a wrong path. Could 
>>>>>>>>> you please double-check?
>>>>>>>>>
>>>>>>>>> Best regards,
>>>>>>>>>
>>>>>>>>> R?zvan Crainea
>>>>>>>>> OpenSIPS Solutions
>>>>>>>>> www.opensips-solutions.com
>>>>>>>>> On 08/16/2016 06:56 PM, Stefano Pisani wrote:
>>>>>>>>>>
>>>>>>>>>> I cloned the source from
>>>>>>>>>>
>>>>>>>>>> |git clone https://github.com/OpenSIPS/opensips.git 
>>>>>>>>>> opensips_head the error appears the same Aug 16 11:55:14 
>>>>>>>>>> [25630] INFO:tls_mgm:mod_init: initializing TLS protocol Aug 
>>>>>>>>>> 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: unable to set the 
>>>>>>>>>> memory allocation functions Aug 16 11:55:14 [25630] 
>>>>>>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
>>>>>>>>>> 1.0.1e-fips, as this version is know to be broken; if so, you 
>>>>>>>>>> need to upgrade or downgrade to a differen openssl version !! 
>>>>>>>>>> Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to 
>>>>>>>>>> initialize module tls_mgm Aug 16 11:55:14 [25630] 
>>>>>>>>>> ERROR:core:main: error while initializing modules Aug 16 
>>>>>>>>>> 11:55:14 [25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 
>>>>>>>>>> [25630] DBG:dispatcher:destroy: destroying module ... Aug 16 
>>>>>>>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 
>>>>>>>>>> 16 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying 
>>>>>>>>>> DELETE list for set 0 Aug 16 11:55:14 [25630] 
>>>>>>>>>> DBG:tm:tm_shutdown: emptying hash table Aug 16 11:55:14 
>>>>>>>>>> [25630] DBG:tm:tm_shutdown: releasing timers Aug 16 11:55:14 
>>>>>>>>>> [25630] DBG:tm:tm_shutdown: removing semaphores Aug 16 
>>>>>>>>>> 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback 
>>>>>>>>>> lists Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown 
>>>>>>>>>> : done Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: 
>>>>>>>>>> destroying the shared memory lock Aug 16 11:55:14 [25630] 
>>>>>>>>>> NOTICE:core:main: Exiting.... Thanks Stefano |
>>>>>>>>>>
>>>>>>>>>> Il 16/08/2016 17:18, R?zvan Crainea ha scritto:
>>>>>>>>>>> Stefano!
>>>>>>>>>>>
>>>>>>>>>>> I've added some extra debugging in the master branch that 
>>>>>>>>>>> might help you debug this. Could you please take the latest 
>>>>>>>>>>> git version/deb and try to run again?
>>>>>>>>>>>
>>>>>>>>>>> Best regards,
>>>>>>>>>>>
>>>>>>>>>>> R?zvan Crainea
>>>>>>>>>>> OpenSIPS Solutions
>>>>>>>>>>> www.opensips-solutions.com
>>>>>>>>>>>
>>>>>>>>>>> On 08/16/2016 10:52 AM, Stefano Pisani wrote:
>>>>>>>>>>>> I'm trying to enable wss on opensips 2.2.1 and Ubuntu 
>>>>>>>>>>>> 16.04.1 LTS
>>>>>>>>>>>>
>>>>>>>>>>>> version: opensips 2.2.1 (x86_64/linux)
>>>>>>>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
>>>>>>>>>>>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>>>>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, 
>>>>>>>>>>>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>>>>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, 
>>>>>>>>>>>> select.
>>>>>>>>>>>> main.c compiled on  with gcc 5.3.1
>>>>>>>>>>>>
>>>>>>>>>>>> It's not working:
>>>>>>>>>>>> Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing 
>>>>>>>>>>>> TLS protocol
>>>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to 
>>>>>>>>>>>> set the memory allocation functions
>>>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check 
>>>>>>>>>>>> if you have openssl 1.0.1e-fips, as this version is know to 
>>>>>>>>>>>> be broken; if so, you need to upgrade or downgrade to a 
>>>>>>>>>>>> differen openssl version !!
>>>>>>>>>>>> Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to 
>>>>>>>>>>>> initialize module tls_mgm
>>>>>>>>>>>>
>>>>>>>>>>>> OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>>>>>>>>
>>>>>>>>>>>> The opensips installed packages are:
>>>>>>>>>>>> opensips                                        install
>>>>>>>>>>>> opensips-tls-module:amd64                       install
>>>>>>>>>>>> opensips-tlsmgm-module:amd64                    install
>>>>>>>>>>>> opensips-wss-module:amd64                       install
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> There is something I can do?
>>>>>>>>>>>> Thanks
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Users mailing list
>>>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Users mailing list
>>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Users mailing list
>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Users mailing list
>>>>>>>>> Users at lists.opensips.org
>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From adrian.fretwell at topgreen.co.uk  Wed Aug 17 20:25:56 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Wed, 17 Aug 2016 19:25:56 +0100
Subject: [OpenSIPS-Users] Dialog information
Message-ID: <4e3d7c04-bb4d-bfc7-4e08-cde036b238f3@topgreen.co.uk>

Hello all,  A simple question that I can't seem to work out for myself:

When I execute the MI command :dlg_list_ctx:  I can see lots of useful 
information about the dialog like caller_contact:: and callee_tag:: etc.

Is there a way of retrieving these values from within the script 
especially from another dialog?  I have tried get_dialog_info but this 
will only retrieve values that I have set in the script through 
store_dlg_value.  Or am I just missing a naming convention for the other 
dialog values and attributes?

Kind regards,

Adrian Fretwell
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From denis7979 at mail.ru  Thu Aug 18 10:58:30 2016
From: denis7979 at mail.ru (Denis)
Date: Thu, 18 Aug 2016 11:58:30 +0300
Subject: [OpenSIPS-Users] reject 3xx code
Message-ID: <686221444.20160818115830@ptl.ru>

Hello

I am using Opensips 2.1.2.
I want to block sending 3xx codes to caller.

In reply route i wrote such code 
 if (status=~"30[1,2]") {
  drop();
 }

but i see that 302, for example, still process successfully.

Thank you for any help.

-- 
mailto:denis7979 at mail.ru
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From adrian.fretwell at topgreen.co.uk  Thu Aug 18 11:03:19 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Thu, 18 Aug 2016 10:03:19 +0100
Subject: [OpenSIPS-Users] Dialog information
In-Reply-To: <4e3d7c04-bb4d-bfc7-4e08-cde036b238f3@topgreen.co.uk>
References: <4e3d7c04-bb4d-bfc7-4e08-cde036b238f3@topgreen.co.uk>
Message-ID: <75d08002-4040-42d0-b9ae-47a615bb7774@topgreen.co.uk>

Ok, To put it another way; It seems like an overhead for me to store say 
a to_tag so I can retrieve it later when the to_tag is already stored in 
the dialog, I just can't work out how to get at it...

     if ( t_check_status("200") ) {
             store_dlg_value("my_callee_tag","$tt");
     }
.......

get_dialog_info("my_callee_tag","$var(my_callee_tag)","my_caller_cid","$var(my_srch_call_id)");

Perhaps there is a way I don't need to store something that is already 
there?

Kind regards,

Adrian Fretwell

On 17/08/16 19:25, Adrian Fretwell wrote:
>
> Hello all,  A simple question that I can't seem to work out for myself:
>
> When I execute the MI command :dlg_list_ctx:  I can see lots of useful 
> information about the dialog like caller_contact:: and callee_tag:: etc.
>
> Is there a way of retrieving these values from within the script 
> especially from another dialog?  I have tried get_dialog_info but this 
> will only retrieve values that I have set in the script through 
> store_dlg_value.  Or am I just missing a naming convention for the 
> other dialog values and attributes?
>
> Kind regards,
>
> Adrian Fretwell
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From adrian.fretwell at topgreen.co.uk  Thu Aug 18 11:19:31 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Thu, 18 Aug 2016 10:19:31 +0100
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <686221444.20160818115830@ptl.ru>
References: <686221444.20160818115830@ptl.ru>
Message-ID: <d621801c-f0f3-b014-803b-39304c1852df@topgreen.co.uk>

Denis,

all the examples I have seen for this do the status check in a failure 
route:

t_on_failure("1");

..........

failure_route[1] {
      if (t_was_cancelled()) {
          exit;
      }

      # uncomment the following lines if you want to block client
      # redirect based on 3xx replies.
      ##if (t_check_status("3[0-9][0-9]")) {
      ##t_reply("404","Not found");
      ##    exit;
      ##}

}

Kind regards,

Adrian Fretwell

On 18/08/16 09:58, Denis wrote:
> reject 3xx code Hello
>
> I am using Opensips 2.1.2.
> I want to block sending 3xx codes to caller.
>
> In reply route i wrote such code
> if (status=~"30[1,2]") {
>  drop();
> }
>
> but i see that 302, for example, still process successfully.
>
> Thank you for any help.
>
> /--
> /mailto:denis7979 at mail.ru
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From abalashov at evaristesys.com  Thu Aug 18 11:23:12 2016
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 18 Aug 2016 05:23:12 -0400
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <d621801c-f0f3-b014-803b-39304c1852df@topgreen.co.uk>
References: <686221444.20160818115830@ptl.ru>
	<d621801c-f0f3-b014-803b-39304c1852df@topgreen.co.uk>
Message-ID: <5308725E-751A-4D27-BE95-BF6FF469DC22@evaristesys.com>

It can certainly be done in the onreply_route. 

Conceptually, doing it there makes more sense. The failure_route is not triggered by any particular SIP reply per se, but rather a branch failure event on a transaction. That can be the result of a timeout (e.g. fr_timer) or something else that doesn't gather message form. 

In contrast, onreply_route is for actual reply messages. 


-- Alex

--
Principal, Evariste Systems LLC (www.evaristesys.com)

Sent from my Google Nexus.



From denis7979 at mail.ru  Thu Aug 18 11:34:57 2016
From: denis7979 at mail.ru (Denis)
Date: Thu, 18 Aug 2016 12:34:57 +0300
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <5308725E-751A-4D27-BE95-BF6FF469DC22@evaristesys.com>
References: <686221444.20160818115830@ptl.ru> 
	<d621801c-f0f3-b014-803b-39304c1852df@topgreen.co.uk>
	<5308725E-751A-4D27-BE95-BF6FF469DC22@evaristesys.com>
Message-ID: <1889324395.20160818123457@ptl.ru>

Alex, and how can i do it replay route?
exit, drop, t_relay something else?



-- 
mailto:denis7979 at mail.ru

> It can certainly be done in the onreply_route. 

> Conceptually, doing it there makes more sense. The failure_route is
> not triggered by any particular SIP reply per se, but rather a
> branch failure event on a transaction. That can be the result of a
> timeout (e.g. fr_timer) or something else that doesn't gather message form.

> In contrast, onreply_route is for actual reply messages. 


> -- Alex

> --
> Principal, Evariste Systems LLC (www.evaristesys.com)

> Sent from my Google Nexus.


> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From Ben.Newlin at inin.com  Thu Aug 18 14:49:57 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Thu, 18 Aug 2016 12:49:57 +0000
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <686221444.20160818115830@ptl.ru>
References: <686221444.20160818115830@ptl.ru>
Message-ID: <972118A8-0F8F-4168-A3F6-80C36F97929E@inin.com>

The comma in your regex is unnecessary; it allows a match for ?30,?. It should just be ?30[12]?.

Also, where is the status variable being set from? There are two places to get the return code: $rs and $T_reply_code.

Or you could use t_check_status() like so:

if (t_check_status(?30[12]?)) {
  drop();
}


Ben Newlin

From: <users-bounces at lists.opensips.org> on behalf of Denis <denis7979 at mail.ru>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Thursday, August 18, 2016 at 4:58 AM
To: "users at lists.opensips.org" <users at lists.opensips.org>
Subject: [OpenSIPS-Users] reject 3xx code

Hello

I am using Opensips 2.1.2.
I want to block sending 3xx codes to caller.

In reply route i wrote such code
if (status=~"30[1,2]") {
 drop();
}

but i see that 302, for example, still process successfully.

Thank you for any help.

--
mailto:denis7979 at mail.ru
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From Ben.Newlin at inin.com  Thu Aug 18 14:56:07 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Thu, 18 Aug 2016 12:56:07 +0000
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <972118A8-0F8F-4168-A3F6-80C36F97929E@inin.com>
References: <686221444.20160818115830@ptl.ru>
	<972118A8-0F8F-4168-A3F6-80C36F97929E@inin.com>
Message-ID: <88FB0218-B3A7-497F-BFA6-0C7B25635142@inin.com>

I see now that status is a global parameter that should also be the reply status.

I would still recommend using t_check_status().


Ben Newlin

From: <users-bounces at lists.opensips.org> on behalf of "Newlin, Ben" <Ben.Newlin at inin.com>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Thursday, August 18, 2016 at 8:49 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] reject 3xx code

The comma in your regex is unnecessary; it allows a match for ?30,?. It should just be ?30[12]?.

Also, where is the status variable being set from? There are two places to get the return code: $rs and $T_reply_code.

Or you could use t_check_status() like so:

if (t_check_status(?30[12]?)) {
  drop();
}


Ben Newlin

From: <users-bounces at lists.opensips.org> on behalf of Denis <denis7979 at mail.ru>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Thursday, August 18, 2016 at 4:58 AM
To: "users at lists.opensips.org" <users at lists.opensips.org>
Subject: [OpenSIPS-Users] reject 3xx code

Hello

I am using Opensips 2.1.2.
I want to block sending 3xx codes to caller.

In reply route i wrote such code
if (status=~"30[1,2]") {
 drop();
}

but i see that 302, for example, still process successfully.

Thank you for any help.

--
mailto:denis7979 at mail.ru
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From stasan89 at gmail.com  Thu Aug 18 15:40:09 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Thu, 18 Aug 2016 16:40:09 +0300
Subject: [OpenSIPS-Users] Configure freeswitch and opensips for using tls
	and udp protocols simultaneously.
Message-ID: <CAC89h=LbVaT18nh=cRvutCeJS5evBzK3kyVM9a0h909Y5-xLEA@mail.gmail.com>

I have freeswitch and opensips working with the mobile client in the
conference mode.
When using UDP connection everything works perfectly, but when using tls
connection the call is interrupted in 30 seconds.
Whether to use TLS or UDP connection - it is assigned on the mobile client
before initialization of connection with opensips server.

Originally I assumed that these problems were caused by the NAT settings,
but in that case the problem would be watched irrespective of the
connection used - UDP or TLS.

Generally such scheme works as it should:

+++++++++   udp   ++++++++   udp   +++++++++   udp   +++++++++
+               + ----->  +              +  ----->  +               +
----->  +               +
+   phone  +           +   SIP     +             +    free    +           +
    SIP    +
+               + <-----  +              +  <-----  +   switch  + <-----  +
provider +
+++++++++   udp   ++++++++   udp    +++++++++   udp   +++++++++

And in such scheme a call breaks in 30 seconds:

+++++++++   tls   +++++++++   udp   +++++++++   udp   +++++++++
+               + ----->  +               +  ----->  +               +
----->  +               +
+   phone  +           +   SIP      +             +    free    +
+     SIP    +
+               + <-----  +               +  <-----  +   switch  + <-----
+ provider +
+++++++++   tls   +++++++++   udp    +++++++++   udp   +++++++++

SIP and freeswitch are in one local area network (Amazon EC2). SIP provider
doesn't support tls in principle, they have 5061 closed.

And the BYE packet sends freeswitch, as I understand, from packet headers
as I didn't receive the response to ACK in time. There is the packet:
BYE sip:8 at 85.236.*.*:55194;ob;transport=tls SIP/2.0
Via: SIP/2.0/TLS sip0.*.*:5061;branch=z9hG4bKc7a2.7909e7e1.0;
received=52.58.*.*
Via: SIP/2.0/UDP 172.31.*.*;received=52.58.*.*;rport=5060;branch=
z9hG4bKBK82Zg50c2U0p
Max-Forwards: 69
Contact: <sip:*7906******@52.58.*.*:5060;transport=udp>
To: "8" <sip:8 at sip0.*.*>;tag=59221e6a
From: <sip:*7906******@sip0.*.*>;tag=j4aX21rv83etN
Call-ID: O7E3ktwLPiQWDN2Rism-7g..
CSeq: 95383912 BYE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0

Having looked on logs, I can tell that the INVITE packet from the mobile
client reach freeswitch and provider, but in reverse Trying/Ringing packet
doesn't reach.

I can't understand at what stage there is a problem. Freeswitch can't
respond and transmit the response through opensips, or there is a problem
in something else?
Who faced similar problem, prompt what settings should be analyzed in order
that the above-stated scheme with tls connection start functionning?
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From millennium.bug at gmail.com  Thu Aug 18 15:46:15 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Thu, 18 Aug 2016 18:46:15 +0500
Subject: [OpenSIPS-Users] db_mysql prep stmt took too long
Message-ID: <CAArAGK=SNR674SYbaUkvNDSxtvqLRnsEfxD2o6sTDtsPS8CRtw@mail.gmail.com>

Hi all,

I am Currently using opensips 2.2.0 and have the following query threshold
set:

loadmodule "db_mysql.so"
modparam("db_mysql", "exec_query_threshold", 50000)

I am getting a lot of errors like the following even with a handful of
incoming SIP messages.

WARNING:db_mysql:log_expiry: threshold exceeded : mysql prep stmt took too
 long - 61625 us.Source

The database exists on the same host as opensips itself. The DB engine is
InnoDB and its optimized because all external queries to the same table /
database return instantly.

Has anyone else experienced the same?
Any ideas to have this fixed?

Regards,
Owais
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From denis7979 at mail.ru  Fri Aug 19 16:51:10 2016
From: denis7979 at mail.ru (Denis)
Date: Fri, 19 Aug 2016 17:51:10 +0300
Subject: [OpenSIPS-Users] Opensips 2.1.2 dr_rules is empty
Message-ID: <1788533534.20160819175110@ptl.ru>

Hello!

Recently i begun to get such problem.
Opensips rejects all calls with 404 code. I can solve the problem by making dr_reload.
I tried to analyze a log and what i could found you can see in attachment.
As i could understand last problem began after dr_reload.

Thank you for any help.


mailto:denis7979 at mail.ru
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From bogdan at opensips.org  Fri Aug 19 17:35:40 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 19 Aug 2016 18:35:40 +0300
Subject: [OpenSIPS-Users] Configure freeswitch and opensips for using
 tls and udp protocols simultaneously.
In-Reply-To: <CAC89h=LbVaT18nh=cRvutCeJS5evBzK3kyVM9a0h909Y5-xLEA@mail.gmail.com>
References: <CAC89h=LbVaT18nh=cRvutCeJS5evBzK3kyVM9a0h909Y5-xLEA@mail.gmail.com>
Message-ID: <0fe313f2-3201-03ac-cfd0-c91d98eacccb@opensips.org>

Hi Stas.

A missing ACK may indicate a problem with the contact in the 200 OK 
reply (contact pointing back to callee).
Do you have a SIP capture on the OpenSIPS side ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.08.2016 16:40, ???? ??????? wrote:
> I have freeswitch and opensips working with the mobile client in the 
> conference mode.
> When using UDP connection everything works perfectly, but when using 
> tls connection the call is interrupted in 30 seconds.
> Whether to use TLS or UDP connection - it is assigned on the mobile 
> client before initialization of connection with opensips server.
>
> Originally I assumed that these problems were caused by the NAT 
> settings, but in that case the problem would be watched irrespective 
> of the connection used - UDP or TLS.
>
> Generally such scheme works as it should:
>
> +++++++++   udp   ++++++++   udp   +++++++++   udp   +++++++++
> +               + ----->  +              +  -----> +               + 
> ----->  +               +
> +   phone  +           +   SIP     +             +    free +           
> +     SIP    +
> +               + <-----  +              +  <-----  + switch  + 
> <-----  + provider +
> +++++++++   udp   ++++++++   udp    +++++++++   udp +++++++++
>
> And in such scheme a call breaks in 30 seconds:
>
> +++++++++   tls   +++++++++   udp   +++++++++   udp +++++++++
> +               + ----->  +               +  -----> +               + 
> ----->  +               +
> +   phone  +           +   SIP      +             +    free +        
>    +     SIP    +
> +               + <-----  +               +  <-----  + switch  + 
> <-----  + provider +
> +++++++++   tls   +++++++++   udp    +++++++++   udp +++++++++
>
> SIP and freeswitch are in one local area network (Amazon EC2). SIP 
> provider doesn't support tls in principle, they have 5061 closed.
>
> And the BYE packet sends freeswitch, as I understand, from packet 
> headers as I didn't receive the response to ACK in time. There is the 
> packet:
> BYE sip:8 at 85.236.*.*:55194;ob;transport=tls SIP/2.0
> Via: SIP/2.0/TLS 
> sip0.*.*:5061;branch=z9hG4bKc7a2.7909e7e1.0;received=52.58.*.*
> Via: SIP/2.0/UDP 
> 172.31.*.*;received=52.58.*.*;rport=5060;branch=z9hG4bKBK82Zg50c2U0p
> Max-Forwards: 69
> Contact: <sip:*7906******@52.58.*.*:5060;transport=udp>
> To: "8" <sip:8 at sip0.*.*>;tag=59221e6a
> From: <sip:*7906******@sip0.*.*>;tag=j4aX21rv83etN
> Call-ID: O7E3ktwLPiQWDN2Rism-7g..
> CSeq: 95383912 BYE
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
> REGISTER, REFER, NOTIFY
> Supported: timer, path, replaces
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
> Reason: SIP;cause=408;text="ACK Timeout"
> Content-Length: 0
>
> Having looked on logs, I can tell that the INVITE packet from the 
> mobile client reach freeswitch and provider, but in reverse 
> Trying/Ringing packet doesn't reach.
>
> I can't understand at what stage there is a problem. Freeswitch can't 
> respond and transmit the response through opensips, or there is a 
> problem in something else?
> Who faced similar problem, prompt what settings should be analyzed in 
> order that the above-stated scheme with tls connection start 
> functionning?
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Aug 19 17:40:06 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 19 Aug 2016 18:40:06 +0300
Subject: [OpenSIPS-Users] Dialog information
In-Reply-To: <75d08002-4040-42d0-b9ae-47a615bb7774@topgreen.co.uk>
References: <4e3d7c04-bb4d-bfc7-4e08-cde036b238f3@topgreen.co.uk>
	<75d08002-4040-42d0-b9ae-47a615bb7774@topgreen.co.uk>
Message-ID: <5b9631de-4904-86c4-c35f-213574dfd560@opensips.org>

Hi Adrian,

Unfortunately there is no way to directly access the dialog info 
remotely (from the context of another dialog). Maybe this will be a 
great addition if you consider opening a ticket on the GITHUB tracker.

Now, have you managed to make it work using the double storing in 
conjunction with get_dialog_info() ?


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.08.2016 12:03, Adrian Fretwell wrote:
>
> Ok, To put it another way; It seems like an overhead for me to store 
> say a to_tag so I can retrieve it later when the to_tag is already 
> stored in the dialog, I just can't work out how to get at it...
>
>     if ( t_check_status("200") ) {
>             store_dlg_value("my_callee_tag","$tt");
>     }
> .......
>
> get_dialog_info("my_callee_tag","$var(my_callee_tag)","my_caller_cid","$var(my_srch_call_id)");
>
> Perhaps there is a way I don't need to store something that is already 
> there?
>
> Kind regards,
>
> Adrian Fretwell
>
> On 17/08/16 19:25, Adrian Fretwell wrote:
>>
>> Hello all,  A simple question that I can't seem to work out for myself:
>>
>> When I execute the MI command :dlg_list_ctx: I can see lots of useful 
>> information about the dialog like caller_contact:: and callee_tag:: etc.
>>
>> Is there a way of retrieving these values from within the script 
>> especially from another dialog?  I have tried get_dialog_info but 
>> this will only retrieve values that I have set in the script through 
>> store_dlg_value.  Or am I just missing a naming convention for the 
>> other dialog values and attributes?
>>
>> Kind regards,
>>
>> Adrian Fretwell
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Aug 19 17:44:32 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 19 Aug 2016 18:44:32 +0300
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <686221444.20160818115830@ptl.ru>
References: <686221444.20160818115830@ptl.ru>
Message-ID: <30efb4bf-8b9f-014d-6366-bc86ba8d286a@opensips.org>

Hi Denis,

3xx is a final response and it cannot be simply drop (as will let the 
transaction state machine in an ambiguous state). Not to mention that 
you will de-syncronize the callee side (which rejected the call) and the 
caller side (still waiting for a final answer).

What you can do is to replace the 3xx with another final negative reply. 
Use a failure_route to do that:

failure_route[block_redirect]
{
     if (t_check_status("3[0-9][0-9]))
         t_reply("404","Not Found");
}

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.08.2016 11:58, Denis wrote:
> reject 3xx code Hello
>
> I am using Opensips 2.1.2.
> I want to block sending 3xx codes to caller.
>
> In reply route i wrote such code
> if (status=~"30[1,2]") {
>  drop();
> }
>
> but i see that 302, for example, still process successfully.
>
> Thank you for any help.
>
> /--
> /mailto:denis7979 at mail.ru
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From adrian.fretwell at topgreen.co.uk  Fri Aug 19 17:53:41 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Fri, 19 Aug 2016 16:53:41 +0100
Subject: [OpenSIPS-Users] Dialog information
In-Reply-To: <5b9631de-4904-86c4-c35f-213574dfd560@opensips.org>
References: <4e3d7c04-bb4d-bfc7-4e08-cde036b238f3@topgreen.co.uk>
	<75d08002-4040-42d0-b9ae-47a615bb7774@topgreen.co.uk>
	<5b9631de-4904-86c4-c35f-213574dfd560@opensips.org>
Message-ID: <801d906e-e3e0-03a4-66b7-bb5c363d0f78@topgreen.co.uk>

Bogdan,

Thankyou for the confirmation, yes I have made it work double storing, I 
was just trying to make my script as efficient as possible.

Kind regards,

Adrian Fretwell

On 19/08/16 16:40, Bogdan-Andrei Iancu wrote:
> Hi Adrian,
>
> Unfortunately there is no way to directly access the dialog info 
> remotely (from the context of another dialog). Maybe this will be a 
> great addition if you consider opening a ticket on the GITHUB tracker.
>
> Now, have you managed to make it work using the double storing in 
> conjunction with get_dialog_info() ?
>
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 18.08.2016 12:03, Adrian Fretwell wrote:
>>
>> Ok, To put it another way; It seems like an overhead for me to store 
>> say a to_tag so I can retrieve it later when the to_tag is already 
>> stored in the dialog, I just can't work out how to get at it...
>>
>>     if ( t_check_status("200") ) {
>>             store_dlg_value("my_callee_tag","$tt");
>>     }
>> .......
>>
>> get_dialog_info("my_callee_tag","$var(my_callee_tag)","my_caller_cid","$var(my_srch_call_id)");
>>
>> Perhaps there is a way I don't need to store something that is 
>> already there?
>>
>> Kind regards,
>>
>> Adrian Fretwell
>>
>> On 17/08/16 19:25, Adrian Fretwell wrote:
>>>
>>> Hello all,  A simple question that I can't seem to work out for myself:
>>>
>>> When I execute the MI command :dlg_list_ctx:  I can see lots of 
>>> useful information about the dialog like caller_contact:: and 
>>> callee_tag:: etc.
>>>
>>> Is there a way of retrieving these values from within the script 
>>> especially from another dialog?  I have tried get_dialog_info but 
>>> this will only retrieve values that I have set in the script through 
>>> store_dlg_value.  Or am I just missing a naming convention for the 
>>> other dialog values and attributes?
>>>
>>> Kind regards,
>>>
>>> Adrian Fretwell
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Fri Aug 19 18:06:58 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 19 Aug 2016 19:06:58 +0300
Subject: [OpenSIPS-Users] Dialog information
In-Reply-To: <801d906e-e3e0-03a4-66b7-bb5c363d0f78@topgreen.co.uk>
References: <4e3d7c04-bb4d-bfc7-4e08-cde036b238f3@topgreen.co.uk>
	<75d08002-4040-42d0-b9ae-47a615bb7774@topgreen.co.uk>
	<5b9631de-4904-86c4-c35f-213574dfd560@opensips.org>
	<801d906e-e3e0-03a4-66b7-bb5c363d0f78@topgreen.co.uk>
Message-ID: <6690bf6f-bf61-f831-c54c-9665c580784a@opensips.org>

In that case, you SHOULD consider opening a feature request on the 
tracker ;) It will get your script more efficient.

Thanks and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 19.08.2016 18:53, Adrian Fretwell wrote:
>
> Bogdan,
>
> Thankyou for the confirmation, yes I have made it work double storing, 
> I was just trying to make my script as efficient as possible.
>
> Kind regards,
>
> Adrian Fretwell
>
> On 19/08/16 16:40, Bogdan-Andrei Iancu wrote:
>> Hi Adrian,
>>
>> Unfortunately there is no way to directly access the dialog info 
>> remotely (from the context of another dialog). Maybe this will be a 
>> great addition if you consider opening a ticket on the GITHUB tracker.
>>
>> Now, have you managed to make it work using the double storing in 
>> conjunction with get_dialog_info() ?
>>
>>
>> Regards,
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>> On 18.08.2016 12:03, Adrian Fretwell wrote:
>>>
>>> Ok, To put it another way; It seems like an overhead for me to store 
>>> say a to_tag so I can retrieve it later when the to_tag is already 
>>> stored in the dialog, I just can't work out how to get at it...
>>>
>>>     if ( t_check_status("200") ) {
>>>             store_dlg_value("my_callee_tag","$tt");
>>>     }
>>> .......
>>>
>>> get_dialog_info("my_callee_tag","$var(my_callee_tag)","my_caller_cid","$var(my_srch_call_id)");
>>>
>>> Perhaps there is a way I don't need to store something that is 
>>> already there?
>>>
>>> Kind regards,
>>>
>>> Adrian Fretwell
>>>
>>> On 17/08/16 19:25, Adrian Fretwell wrote:
>>>>
>>>> Hello all,  A simple question that I can't seem to work out for myself:
>>>>
>>>> When I execute the MI command :dlg_list_ctx:  I can see lots of 
>>>> useful information about the dialog like caller_contact:: and 
>>>> callee_tag:: etc.
>>>>
>>>> Is there a way of retrieving these values from within the script 
>>>> especially from another dialog? I have tried get_dialog_info but 
>>>> this will only retrieve values that I have set in the script 
>>>> through store_dlg_value.  Or am I just missing a naming convention 
>>>> for the other dialog values and attributes?
>>>>
>>>> Kind regards,
>>>>
>>>> Adrian Fretwell
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>

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From denis7979 at mail.ru  Fri Aug 19 18:28:54 2016
From: denis7979 at mail.ru (Denis)
Date: Fri, 19 Aug 2016 19:28:54 +0300
Subject: [OpenSIPS-Users] reject 3xx code
In-Reply-To: <30efb4bf-8b9f-014d-6366-bc86ba8d286a@opensips.org>
References: <686221444.20160818115830@ptl.ru>
	<30efb4bf-8b9f-014d-6366-bc86ba8d286a@opensips.org>
Message-ID: <18410173997.20160819192854@ptl.ru>

Hello, Bogdan

I understand, thank you very much.

mailto:denis7979 at mail.ru


Hi Denis,

3xx is a final response and it cannot be simply drop (as will let the transaction state machine in an ambiguous state). Not to mention that you will de-syncronize the callee side (which rejected the call) and the caller side (still waiting for a final answer).

What you can do is to replace the 3xx with another final negative reply. Use a failure_route to do that:

failure_route[block_redirect]
{
    if (t_check_status("3[0-9][0-9]))
        t_reply("404","Not Found");
}

Regards,
 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.08.2016 11:58, Denis wrote:

reject 3xx code Hello

I am using Opensips 2.1.2.
I want to block sending 3xx codes to caller.

In reply route i wrote such code 
if (status=~"30[1,2]") {
 drop();
}

but i see that 302, for example, still process successfully.

Thank you for any help.

-- 
mailto:denis7979 at mail.ru 

_______________________________________________

Users mailing list

Users at lists.opensips.org

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Aug 19 18:32:00 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 19 Aug 2016 19:32:00 +0300
Subject: [OpenSIPS-Users] db_mysql prep stmt took too long
In-Reply-To: <CAArAGK=SNR674SYbaUkvNDSxtvqLRnsEfxD2o6sTDtsPS8CRtw@mail.gmail.com>
References: <CAArAGK=SNR674SYbaUkvNDSxtvqLRnsEfxD2o6sTDtsPS8CRtw@mail.gmail.com>
Message-ID: <ccdf8fc3-1cff-0a3a-c34f-840b72a64bfd@opensips.org>

Hi Owais,

It myght be something specific to your mysql setup - at least there is 
nothing you can do from the OpenSIPS side. Still keep it mind that your 
get the warning when preparing the statement - this is a bit of a 
special DB action (it is not an actual query).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.08.2016 16:46, Owais Ahmad wrote:
> Hi all,
>
> I am Currently using opensips 2.2.0 and have the following query 
> threshold set:
>
> loadmodule "db_mysql.so"
> modparam("db_mysql", "exec_query_threshold", 50000)
>
> I am getting a lot of errors like the following even with a handful of 
> incoming SIP messages.
>
> WARNING:db_mysql:log_expiry: threshold exceeded : mysql prep stmt took too
>  long - 61625 us.Source
>
> The database exists on the same host as opensips itself. The DB engine 
> is InnoDB and its optimized because all external queries to the same 
> table / database return instantly.
>
> Has anyone else experienced the same?
> Any ideas to have this fixed?
>
> Regards,
> Owais
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From linuxvenkey at gmail.com  Sat Aug 20 21:56:07 2016
From: linuxvenkey at gmail.com (Venkatesh Macha)
Date: Sat, 20 Aug 2016 12:56:07 -0700 (MST)
Subject: [OpenSIPS-Users] tls_mgm
Message-ID: <1471722967190-7604091.post@n2.nabble.com>


Hi all,

 I am trying to install OpenSIPS with WSS support. But i am getting
following error on OpenSIPS Startup.

ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips, as this
version is known to be broken; if so, you need to upgrade or downgrade to a
different openssl version !!
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
realloc=0x7f7490dd2624/0x7f749d13cc40 free=0x7f7490dd265c/0x7f749d13ca70
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:init_mod:
failed to initialize module tls_mgm
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:main:
error while initializing modules

My OpenSIPS version:
 opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC,
DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 326a1c6

My OpenSSL Version:
openssl version
OpenSSL 1.0.2g-fips  1 Mar 2016

I am using Ubuntu 16.04 LTS.

Thank you in advance.



--
View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/tls-mgm-tp7604091.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.


From razvan at opensips.org  Sat Aug 20 22:09:52 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Sat, 20 Aug 2016 23:09:52 +0300
Subject: [OpenSIPS-Users] tls_mgm
In-Reply-To: <1471722967190-7604091.post@n2.nabble.com>
References: <1471722967190-7604091.post@n2.nabble.com>
Message-ID: <ac6517cd-c3b6-6c7f-41fb-021f8f3af81a@opensips.org>

Hello!

Is there any chance you could upgrade your openssl library? This version 
has a known bug.
Also, could you tell us what OS you're running, perhaps we can manage to 
replicate this and track it down.

Best regards.

R?zvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/20/2016 10:56 PM, Venkatesh Macha wrote:
>
> Hi all,
>
>  I am trying to install OpenSIPS with WSS support. But i am getting
> following error on OpenSIPS Startup.
>
> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips, as this
> version is known to be broken; if so, you need to upgrade or downgrade to a
> different openssl version !!
> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
> ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
> realloc=0x7f7490dd2624/0x7f749d13cc40 free=0x7f7490dd265c/0x7f749d13ca70
> version=OpenSSL 1.0.2g-fips  1 Mar 2016
> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:init_mod:
> failed to initialize module tls_mgm
> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:main:
> error while initializing modules
>
> My OpenSIPS version:
>  opensips -V
> version: opensips 2.3.0-dev (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC,
> DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> git revision: 326a1c6
>
> My OpenSSL Version:
> openssl version
> OpenSSL 1.0.2g-fips  1 Mar 2016
>
> I am using Ubuntu 16.04 LTS.
>
> Thank you in advance.
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/tls-mgm-tp7604091.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


From stefano.pisani at omnianet.it  Sat Aug 20 22:14:37 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Sat, 20 Aug 2016 22:14:37 +0200
Subject: [OpenSIPS-Users] tls_mgm
In-Reply-To: <ac6517cd-c3b6-6c7f-41fb-021f8f3af81a@opensips.org>
References: <1471722967190-7604091.post@n2.nabble.com>
	<ac6517cd-c3b6-6c7f-41fb-021f8f3af81a@opensips.org>
Message-ID: <f4d4611b-8021-101c-253c-edfb5da48f4a@omnianet.it>

Hello R?zvan,
in my previous email I told you that I have upgraded openssl to 1.0.2h 
then the error was different.
After that I realized that there was a conflict between ubuntu openssl 
package and new openssl.
Finally, after fixed that, tls_mgm module is working properly.

I'm using ubuntu 16.04 LTS

Thanks
Stefano

Il 20/08/2016 22:09, R?zvan Crainea ha scritto:
> Hello!
>
> Is there any chance you could upgrade your openssl library? This 
> version has a known bug.
> Also, could you tell us what OS you're running, perhaps we can manage 
> to replicate this and track it down.
>
> Best regards.
>
> R?zvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 08/20/2016 10:56 PM, Venkatesh Macha wrote:
>>
>> Hi all,
>>
>>  I am trying to install OpenSIPS with WSS support. But i am getting
>> following error on OpenSIPS Startup.
>>
>> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips, 
>> as this
>> version is known to be broken; if so, you need to upgrade or 
>> downgrade to a
>> different openssl version !!
>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>> ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
>> realloc=0x7f7490dd2624/0x7f749d13cc40 free=0x7f7490dd265c/0x7f749d13ca70
>> version=OpenSSL 1.0.2g-fips  1 Mar 2016
>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: 
>> ERROR:core:init_mod:
>> failed to initialize module tls_mgm
>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:main:
>> error while initializing modules
>>
>> My OpenSIPS version:
>>  opensips -V
>> version: opensips 2.3.0-dev (x86_64/linux)
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>> QM_MALLOC,
>> DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>> git revision: 326a1c6
>>
>> My OpenSSL Version:
>> openssl version
>> OpenSSL 1.0.2g-fips  1 Mar 2016
>>
>> I am using Ubuntu 16.04 LTS.
>>
>> Thank you in advance.
>>
>>
>>
>> -- 
>> View this message in context: 
>> http://opensips-open-sip-server.1449251.n2.nabble.com/tls-mgm-tp7604091.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users




From razvan at opensips.org  Sat Aug 20 22:28:27 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Sat, 20 Aug 2016 23:28:27 +0300
Subject: [OpenSIPS-Users] tls_mgm
In-Reply-To: <f4d4611b-8021-101c-253c-edfb5da48f4a@omnianet.it>
References: <1471722967190-7604091.post@n2.nabble.com>
	<ac6517cd-c3b6-6c7f-41fb-021f8f3af81a@opensips.org>
	<f4d4611b-8021-101c-253c-edfb5da48f4a@omnianet.it>
Message-ID: <7a9d17fd-5fae-e218-1fbb-7b17335aead6@opensips.org>

Hi, Stefano!

Sorry, I missed that email. So what was your solution to solve the 
conflict? Deployed a custom deb?

Best regards,

R?zvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/20/2016 11:14 PM, Stefano Pisani wrote:
> Hello R?zvan,
> in my previous email I told you that I have upgraded openssl to 1.0.2h
> then the error was different.
> After that I realized that there was a conflict between ubuntu openssl
> package and new openssl.
> Finally, after fixed that, tls_mgm module is working properly.
>
> I'm using ubuntu 16.04 LTS
>
> Thanks
> Stefano
>
> Il 20/08/2016 22:09, R?zvan Crainea ha scritto:
>> Hello!
>>
>> Is there any chance you could upgrade your openssl library? This
>> version has a known bug.
>> Also, could you tell us what OS you're running, perhaps we can manage
>> to replicate this and track it down.
>>
>> Best regards.
>>
>> R?zvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 08/20/2016 10:56 PM, Venkatesh Macha wrote:
>>>
>>> Hi all,
>>>
>>>  I am trying to install OpenSIPS with WSS support. But i am getting
>>> following error on OpenSIPS Startup.
>>>
>>> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips,
>>> as this
>>> version is known to be broken; if so, you need to upgrade or
>>> downgrade to a
>>> different openssl version !!
>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>> ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
>>> realloc=0x7f7490dd2624/0x7f749d13cc40 free=0x7f7490dd265c/0x7f749d13ca70
>>> version=OpenSSL 1.0.2g-fips  1 Mar 2016
>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>> ERROR:core:init_mod:
>>> failed to initialize module tls_mgm
>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:main:
>>> error while initializing modules
>>>
>>> My OpenSIPS version:
>>>  opensips -V
>>> version: opensips 2.3.0-dev (x86_64/linux)
>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>> QM_MALLOC,
>>> DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>> git revision: 326a1c6
>>>
>>> My OpenSSL Version:
>>> openssl version
>>> OpenSSL 1.0.2g-fips  1 Mar 2016
>>>
>>> I am using Ubuntu 16.04 LTS.
>>>
>>> Thank you in advance.
>>>
>>>
>>>
>>> --
>>> View this message in context:
>>> http://opensips-open-sip-server.1449251.n2.nabble.com/tls-mgm-tp7604091.html
>>>
>>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From stefano.pisani at omnianet.it  Sat Aug 20 22:35:32 2016
From: stefano.pisani at omnianet.it (Stefano Pisani)
Date: Sat, 20 Aug 2016 22:35:32 +0200
Subject: [OpenSIPS-Users] tls_mgm
In-Reply-To: <7a9d17fd-5fae-e218-1fbb-7b17335aead6@opensips.org>
References: <1471722967190-7604091.post@n2.nabble.com>
	<ac6517cd-c3b6-6c7f-41fb-021f8f3af81a@opensips.org>
	<f4d4611b-8021-101c-253c-edfb5da48f4a@omnianet.it>
	<7a9d17fd-5fae-e218-1fbb-7b17335aead6@opensips.org>
Message-ID: <a2c232b9-e9e4-9c25-1d3a-55b86f51c9e2@omnianet.it>

Hi R?zvan,
there is not openssl 1.0.2h available for ubuntu 16.04 so I installed it 
from the tarball and fixed the symlink to the correct libraries libssl 
and libcrypto. That's all.

Thanks.
Stefano

Il 20/08/2016 22:28, R?zvan Crainea ha scritto:
> Hi, Stefano!
>
> Sorry, I missed that email. So what was your solution to solve the 
> conflict? Deployed a custom deb?
>
> Best regards,
>
> R?zvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 08/20/2016 11:14 PM, Stefano Pisani wrote:
>> Hello R?zvan,
>> in my previous email I told you that I have upgraded openssl to 1.0.2h
>> then the error was different.
>> After that I realized that there was a conflict between ubuntu openssl
>> package and new openssl.
>> Finally, after fixed that, tls_mgm module is working properly.
>>
>> I'm using ubuntu 16.04 LTS
>>
>> Thanks
>> Stefano
>>
>> Il 20/08/2016 22:09, R?zvan Crainea ha scritto:
>>> Hello!
>>>
>>> Is there any chance you could upgrade your openssl library? This
>>> version has a known bug.
>>> Also, could you tell us what OS you're running, perhaps we can manage
>>> to replicate this and track it down.
>>>
>>> Best regards.
>>>
>>> R?zvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>>
>>> On 08/20/2016 10:56 PM, Venkatesh Macha wrote:
>>>>
>>>> Hi all,
>>>>
>>>>  I am trying to install OpenSIPS with WSS support. But i am getting
>>>> following error on OpenSIPS Startup.
>>>>
>>>> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips,
>>>> as this
>>>> version is known to be broken; if so, you need to upgrade or
>>>> downgrade to a
>>>> different openssl version !!
>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>> ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
>>>> realloc=0x7f7490dd2624/0x7f749d13cc40 
>>>> free=0x7f7490dd265c/0x7f749d13ca70
>>>> version=OpenSSL 1.0.2g-fips  1 Mar 2016
>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>> ERROR:core:init_mod:
>>>> failed to initialize module tls_mgm
>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: 
>>>> ERROR:core:main:
>>>> error while initializing modules
>>>>
>>>> My OpenSIPS version:
>>>>  opensips -V
>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>>> QM_MALLOC,
>>>> DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>> git revision: 326a1c6
>>>>
>>>> My OpenSSL Version:
>>>> openssl version
>>>> OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>
>>>> I am using Ubuntu 16.04 LTS.
>>>>
>>>> Thank you in advance.
>>>>
>>>>
>>>>
>>>> -- 
>>>> View this message in context:
>>>> http://opensips-open-sip-server.1449251.n2.nabble.com/tls-mgm-tp7604091.html 
>>>>
>>>>
>>>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users




From razvan at opensips.org  Sat Aug 20 22:57:38 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Sat, 20 Aug 2016 23:57:38 +0300
Subject: [OpenSIPS-Users] tls_mgm
In-Reply-To: <a2c232b9-e9e4-9c25-1d3a-55b86f51c9e2@omnianet.it>
References: <1471722967190-7604091.post@n2.nabble.com>
	<ac6517cd-c3b6-6c7f-41fb-021f8f3af81a@opensips.org>
	<f4d4611b-8021-101c-253c-edfb5da48f4a@omnianet.it>
	<7a9d17fd-5fae-e218-1fbb-7b17335aead6@opensips.org>
	<a2c232b9-e9e4-9c25-1d3a-55b86f51c9e2@omnianet.it>
Message-ID: <6a562e1b-f27f-8fd5-dcac-5f79a0cb920c@opensips.org>

Oh, so basically the upgrade fixed it.

Thanks for sharing!

R?zvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/20/2016 11:35 PM, Stefano Pisani wrote:
> Hi R?zvan,
> there is not openssl 1.0.2h available for ubuntu 16.04 so I installed it
> from the tarball and fixed the symlink to the correct libraries libssl
> and libcrypto. That's all.
>
> Thanks.
> Stefano
>
> Il 20/08/2016 22:28, R?zvan Crainea ha scritto:
>> Hi, Stefano!
>>
>> Sorry, I missed that email. So what was your solution to solve the
>> conflict? Deployed a custom deb?
>>
>> Best regards,
>>
>> R?zvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 08/20/2016 11:14 PM, Stefano Pisani wrote:
>>> Hello R?zvan,
>>> in my previous email I told you that I have upgraded openssl to 1.0.2h
>>> then the error was different.
>>> After that I realized that there was a conflict between ubuntu openssl
>>> package and new openssl.
>>> Finally, after fixed that, tls_mgm module is working properly.
>>>
>>> I'm using ubuntu 16.04 LTS
>>>
>>> Thanks
>>> Stefano
>>>
>>> Il 20/08/2016 22:09, R?zvan Crainea ha scritto:
>>>> Hello!
>>>>
>>>> Is there any chance you could upgrade your openssl library? This
>>>> version has a known bug.
>>>> Also, could you tell us what OS you're running, perhaps we can manage
>>>> to replicate this and track it down.
>>>>
>>>> Best regards.
>>>>
>>>> R?zvan Crainea
>>>> OpenSIPS Core Developer
>>>> http://www.opensips-solutions.com
>>>>
>>>> On 08/20/2016 10:56 PM, Venkatesh Macha wrote:
>>>>>
>>>>> Hi all,
>>>>>
>>>>>  I am trying to install OpenSIPS with WSS support. But i am getting
>>>>> following error on OpenSIPS Startup.
>>>>>
>>>>> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
>>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>>> ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips,
>>>>> as this
>>>>> version is known to be broken; if so, you need to upgrade or
>>>>> downgrade to a
>>>>> different openssl version !!
>>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>>> ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
>>>>> realloc=0x7f7490dd2624/0x7f749d13cc40
>>>>> free=0x7f7490dd265c/0x7f749d13ca70
>>>>> version=OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>>> ERROR:core:init_mod:
>>>>> failed to initialize module tls_mgm
>>>>> Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
>>>>> ERROR:core:main:
>>>>> error while initializing modules
>>>>>
>>>>> My OpenSIPS version:
>>>>>  opensips -V
>>>>> version: opensips 2.3.0-dev (x86_64/linux)
>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>>>> QM_MALLOC,
>>>>> DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>>> git revision: 326a1c6
>>>>>
>>>>> My OpenSSL Version:
>>>>> openssl version
>>>>> OpenSSL 1.0.2g-fips  1 Mar 2016
>>>>>
>>>>> I am using Ubuntu 16.04 LTS.
>>>>>
>>>>> Thank you in advance.
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> View this message in context:
>>>>> http://opensips-open-sip-server.1449251.n2.nabble.com/tls-mgm-tp7604091.html
>>>>>
>>>>>
>>>>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From ffshoh at gmail.com  Sun Aug 21 03:19:22 2016
From: ffshoh at gmail.com (Jon Abrams)
Date: Sat, 20 Aug 2016 20:19:22 -0500
Subject: [OpenSIPS-Users] Listening on all interfaces on Opensips 2.2
Message-ID: <CAP9sNzGXe+99FaVmJqobh6ktb0re7U+=vg=bz46-oT69T2Ke6A@mail.gmail.com>

Hello list,

I've been noticing the listen configuration seems to work differently on
the latest 2.2.X versions.

I'm trying to listen on all addresses (essentially 0.0.0.0) without having
to manually add each ip line by line to the script.

In the old days you could just omit the listen and set the port. However I
need to leave a listen directive for HEP.

Mentioned in the 2.2 docs, listen=eth0:5060, gives a vague generic script
error. listen=udp:eth0:5060 works. Are the docs out of date on some of
these? disable_tcp and port seem to give generic script errors despite
being still listed in the docs

Best Regards,

Jon
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From adrian.fretwell at topgreen.co.uk  Sun Aug 21 10:05:40 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Sun, 21 Aug 2016 09:05:40 +0100
Subject: [OpenSIPS-Users] B2BUA RURI with headers
Message-ID: <0f952149-22a2-de92-ac5d-0f077d1be377@topgreen.co.uk>

Hello,

I have put together a very simple scenario that bridges two clients, it 
is called from the MI (datagram) interface. I want the two SIP messages 
that go out to the clients to contain some header values, so I specify 
these in the two parameters sent in the MI call.  Please note, I am 
using the bash shell, /dev/udp... does not actually exist:

/bin/echo -e 
":b2b_trigger_scenario:\nahf1\nsip:alice at abc.net?header1=qaz&header2=wsx\nbob at abc.net?header1=qaz&header2=wsx" 
 > /dev/udp/192.168.126.72/9061

This works but the problem I have is that the B2BUA includes the header 
part of the RURI in both the To and From headers.  When the UAC responds 
with a 200 OK, B2BUA sees the extra information in the To and From 
headers and throws a "unable to parse", "bad header" error.

Is there any way to stop B2BUA from including the RURI header 
information in the To and From headers?  Can headers be specified 
separately in the <destination> xml tag?

Kind regards,

Adrian Fretwell
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From denis7979 at mail.ru  Mon Aug 22 13:20:30 2016
From: denis7979 at mail.ru (Denis)
Date: Mon, 22 Aug 2016 14:20:30 +0300
Subject: [OpenSIPS-Users] Opensips 2.1 crash
Message-ID: <1456730494.20160822142030@ptl.ru>

Hello!

Today i have a big problem with my opensips.
I wanted to insert some modification to opensips.cfg but after restart opensips it has died and i was not able to resuscitate him at all.
I have been saved only that i was planning to migrate from 2.1 to 2.2 and i already have opensips.cfg for 2.2 (db migration not spend much time).
I.e. only with 2.2. i could restore the service!

Changes that i made to configuration of 2.1:
i added 
 if (t_check_status("3[0-9][0-9]")) {
   t_reply("404","Not Found");
   exit;
 }
to the failure route. That is all.

Everything i could gather about the problem is here: https://cloud.mail.ru/public/4Hhj/TcotVSA7w.
In syslog, at each time when i tried to restart Opensips, i saw
[  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]

And one more. I have two servers for running Opensips (active/standby). On both servers i have the same Opensips version. And on both servers, while restarting Opensips, i saw
[  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]

Thank you for any help! I am afraid that there are some identical problems and with 2.2.

mailto:denis7979 at mail.ru 
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From liviu at opensips.org  Mon Aug 22 15:00:10 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 22 Aug 2016 16:00:10 +0300
Subject: [OpenSIPS-Users] Opensips 2.1 crash
In-Reply-To: <1456730494.20160822142030@ptl.ru>
References: <1456730494.20160822142030@ptl.ru>
Message-ID: <b9204ea3-9f77-7f50-1f41-79cafe94628d@opensips.org>

Hi, Denis!

Thank you for reporting the crash! Unfortunately, OpenSIPS was not able 
to finish writing the corefile, which was truncated, as gdb points out:

/opensipscore/core  ... <F4><E0><E9><EB><E0>  >= 2158587904

The size of a corefile is equal to PKG+SHM memory size. Provided that 
you have enough disk space (> 2 GB), could you add a "ulimit -c 
unlimited" command in your startup script, before launching OpenSIPS? We 
should be able to obtain a clear backtrace afterwards.

Cheers,

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 22.08.2016 14:20, Denis wrote:
> Opensips 2.1 crash Hello!
>
> Today i have a big problem with my opensips.
> I wanted to insert some modification to opensips.cfg but after restart 
> opensips it has died and i was not able to resuscitate him at all.
> I have been saved only that i was planning to migrate from 2.1 to 2.2 
> and i already have opensips.cfg for 2.2 (db migration not spend much 
> time).
> I.e. only with 2.2. i could restore the service!
>
> Changes that i made to configuration of 2.1:
> i added
> if (t_check_status("3[0-9][0-9]")) {
>   t_reply("404","Not Found");
>   exit;
> }
> to the failure route. That is all.
>
> Everything i could gather about the problem is here: 
> https://cloud.mail.ru/public/4Hhj/TcotVSA7w.
> In syslog, at each time when i tried to restart Opensips, i saw
> [  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 
> 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]
>
> And one more. I have two servers for running Opensips 
> (active/standby). On both servers i have the same Opensips version. 
> And on both servers, while restarting Opensips, i saw
> [  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 
> 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]
>
> Thank you for any help! I am afraid that there are some identical 
> problems and with 2.2.
>
> mailto:denis7979 at mail.ru
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From denis7979 at mail.ru  Mon Aug 22 15:25:45 2016
From: denis7979 at mail.ru (Denis)
Date: Mon, 22 Aug 2016 16:25:45 +0300
Subject: [OpenSIPS-Users] Opensips 2.1 crash
In-Reply-To: <b9204ea3-9f77-7f50-1f41-79cafe94628d@opensips.org>
References: <1456730494.20160822142030@ptl.ru>
	<b9204ea3-9f77-7f50-1f41-79cafe94628d@opensips.org>
Message-ID: <309313799.20160822162545@ptl.ru>

Hello, Liviu!

Where does this option, i mean "ulimit -c unlimited", used? In opensipsctlrc "startoptions"?

When i tried restart Opensips i just used opensipsctl stop and start.

Without core analyze is there any suggestions about the problem (logs from opensipslog and syslog)?

-- 
mailto:denis7979 at ptl.ru


Hi, Denis!
Thank you for reporting the crash! Unfortunately, OpenSIPS was not able to finish writing the corefile, which was truncated, as gdb points out:
/opensipscore/core  ... <F4><E0><E9><EB><E0>  >= 2158587904
The size of a corefile is equal to PKG+SHM memory size. Provided that you have enough disk space (> 2 GB), could you add a "ulimit -c unlimited" command in your startup script, before launching OpenSIPS? We should be able to obtain a clear backtrace afterwards.
Cheers,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 22.08.2016 14:20, Denis wrote:

Opensips 2.1 crash Hello!

Today i have a big problem with my opensips.
I wanted to insert some modification to opensips.cfg but after restart opensips it has died and i was not able to resuscitate him at all.
I have been saved only that i was planning to migrate from 2.1 to 2.2 and i already have opensips.cfg for 2.2 (db migration not spend much time).
I.e. only with 2.2. i could restore the service!

Changes that i made to configuration of 2.1:
i added 
if (t_check_status("3[0-9][0-9]")) {
  t_reply("404","Not Found");
  exit;
}
to the failure route. That is all.

Everything i could gather about the problem is here: https://cloud.mail.ru/public/4Hhj/TcotVSA7w.
In syslog, at each time when i tried to restart Opensips, i saw
[  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]

And one more. I have two servers for running Opensips (active/standby). On both servers i have the same Opensips version. And on both servers, while restarting Opensips, i saw
[  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]

Thank you for any help! I am afraid that there are some identical problems and with 2.2.

mailto:denis7979 at mail.ru 


_______________________________________________

Users mailing list

Users at lists.opensips.org

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From liviu at opensips.org  Mon Aug 22 15:36:24 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 22 Aug 2016 16:36:24 +0300
Subject: [OpenSIPS-Users] Opensips 2.1 crash
In-Reply-To: <309313799.20160822162545@ptl.ru>
References: <1456730494.20160822142030@ptl.ru>
	<b9204ea3-9f77-7f50-1f41-79cafe94628d@opensips.org>
	<309313799.20160822162545@ptl.ru>
Message-ID: <3137ead1-b5e6-5cf1-6c7b-c20762e167f4@opensips.org>

Edit /usr/local/opensips2.1/sbin/opensipsctl (line ~1878) and add the 
"ulimit" command before OpenSIPS starts, like so:

   ulimit -c unlimited
   if [ $SYSLOG = 1 ] ; then
     $OSIPSBIN -P $PID_FILE $STARTOPTIONS 1>/dev/null 2>/dev/null
   else
     $OSIPSBIN -P $PID_FILE -E $STARTOPTIONS
   fi

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 22.08.2016 16:25, Denis wrote:
> Re: [OpenSIPS-Users] Opensips 2.1 crash Hello, Liviu!
>
> Where does this option, i mean "ulimit -c unlimited", used? In 
> opensipsctlrc "startoptions"?
>
> When i tried restart Opensips i just used opensipsctl stop and start.
>
> Without core analyze is there any suggestions about the problem (logs 
> from opensipslog and syslog)?
>
> /--
> /mailto:denis7979 at ptl.ru <mailto:d.putyato at ptl.ru>
>
>
> 	Hi, Denis!
> Thank you for reporting the crash! Unfortunately, OpenSIPS was not 
> able to finish writing the corefile, which was truncated, as gdb 
> points out:
> /opensipscore/core  ... <F4><E0><E9><EB><E0>  >= 2158587904
> The size of a corefile is equal to PKG+SHM memory size. Provided that 
> you have enough disk space (> 2 GB), could you add a "ulimit -c 
> unlimited" command in your startup script, before launching OpenSIPS? 
> We should be able to obtain a clear backtrace afterwards.
> Cheers,
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 22.08.2016 14:20, Denis wrote:
>
> 	Opensips 2.1 crash Hello!
>
> Today i have a big problem with my opensips.
> I wanted to insert some modification to opensips.cfg but after restart 
> opensips it has died and i was not able to resuscitate him at all.
> I have been saved only that i was planning to migrate from 2.1 to 2.2 
> and i already have opensips.cfg for 2.2 (db migration not spend much 
> time).
> I.e. only with 2.2. i could restore the service!
>
> Changes that i made to configuration of 2.1:
> i added
> if (t_check_status("3[0-9][0-9]")) {
>  t_reply("404","Not Found");
>  exit;
> }
> to the failure route. That is all.
>
> Everything i could gather about the problem is here: 
> https://cloud.mail.ru/public/4Hhj/TcotVSA7w.
> In syslog, at each time when i tried to restart Opensips, i saw
> [  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 
> 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]
>
> And one more. I have two servers for running Opensips 
> (active/standby). On both servers i have the same Opensips version. 
> And on both servers, while restarting Opensips, i saw
> [  140.300956] opensips[2688]: segfault at 0 ip 00007f69f92db144 sp 
> 00007fff6328e7b8 error 4 in drouting.so[7f69f92d4000+46000]
>
> Thank you for any help! I am afraid that there are some identical 
> problems and with 2.2.
>
> mailto:denis7979 at mail.ru
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From jwilkie at usipcom.com  Mon Aug 22 23:56:54 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Mon, 22 Aug 2016 17:56:54 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
Message-ID: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>

New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP 5.0

It appears that when modifying the CP for Dialplan attributes as text, they
do not appear in the edit section once applied.  The attributes do show up
in the view list of dial plans but if you go back to edit the attrs again,
they are not there.  I dot not have the same problem when using attrs as
checkboxes.  Is there more code in the local files that require editing to
display these values?  Also, if an entry in the Dialplan was once a
checkbox value but I've not converted the mode to text, the value does not
show up in the text box to be edited.

# Attention : advanced options !!



 //database tables

 $config->table_dialplan = "dialplan";

 $config->results_per_page = 20;

 $config->results_page_range = 5;


 $config->attrs_cb=array(

                                        // name , description

                                        array("a","Descr a"),

                                        array("b","Descr b"),

                                        array("c","Descr c"),

                                        array("d","Descr d"),

                                        array("e","Descr e"),

                                        array("f","Descr f"),

                                        );

 $config->cb_per_row = 3;

 $talk_to_this_assoc_id = 1 ;


// Dialplan - Add/Edit new Translation Rule - Attributes input type

// 0 - checkboxes

// 1 - text

$dialplan_attributes_mode = 1  ;


?>



Jeff
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From achalkov at ya.ru  Tue Aug 23 16:20:45 2016
From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Tue, 23 Aug 2016 17:20:45 +0300
Subject: [OpenSIPS-Users] accessing to branch data
Message-ID: <727051471962045@web8m.yandex.ru>

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From achalkov at ya.ru  Tue Aug 23 17:31:44 2016
From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Tue, 23 Aug 2016 18:31:44 +0300
Subject: [OpenSIPS-Users] accessing to branch data
In-Reply-To: <702701471963831@web30h.yandex.ru>
Message-ID: <10091471966304@web11m.yandex.ru>

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From liviu at opensips.org  Tue Aug 23 18:04:11 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 23 Aug 2016 19:04:11 +0300
Subject: [OpenSIPS-Users] accessing to branch data
In-Reply-To: <10091471966304@web11m.yandex.ru>
References: <10091471966304@web11m.yandex.ru>
Message-ID: <052c8628-67b8-967e-163a-8b592221c061@opensips.org>

Hi Artem!


Here are a few examples, so you better understand how the branch array 
gets populated after you do a lookup():

lookup();

- with 1 device  behind an AoR: R-URI is updated (branch #1), $branch 
array has 0 elements
- with 2 devices behind an AoR: R-URI is updated (branch #1), $branch 
array has 1 element (index 0, corresponding to branch #2)
- with 3 devices behind an AoR: R-URI is updated (branch #1), $branch 
array has 2 elements (indexes 0 and 1, corresponding to branches #2 and #3)

...

Regarding deleting/adding branches vs. increasing MAX_BRANCHES (in 
config.h), I'd choose the second option. It will only cost you a bit 
more memory for each transaction.

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 23.08.2016 18:31, ?????? ????? wrote:
> Ok, i found my mistake with 2nd part (deleting branches) - it seems 
> like i cant delete branches from failure_route by $T_branch_idx 
> because outside branch_route it always have value 0. Okay, so, can i 
> somehow delete "expired" branches (to not reach branch limit. yes, i 
> know that i can extend the limit, but i think its a bad idea) or its 
> totally impossible?
> Also, first part of my question (about accessing data) is still actual.
>
> 23.08.2016, 17:21, "?????? ?????" <achalkov at ya.ru 
> <mailto:achalkov at ya.ru>>:
>>
>> Hi all!
>> Could you tell me how can i get access to the data of branches, 
>> generated by lookup()?
>>
>> I have something like that:
>>
>> ...
>> route[LOOKUP] {
>> ...
>> lookup("location");
>> t_on_branch("ON_BRANCH");
>> ...
>> }
>>
>> branch_route["ON_BRANCH"] {
>> ...
>> xlog("L_INFO", "[ON_BRANCH] Current branch data: idx = $T_branch_idx, 
>> uri = $(branch(uri)[$T_branch_idx])");
>> ...
>> }
>> ...
>>
>> When lookup found more than 1 contact it appending a new branches, 
>> but i could read any data only from first of them:
>>
>> Aug 23 16:56:03 opensips-test /usr/sbin/opensips[18629]: [ON_BRANCH] 
>> Current branch data: idx = 0, uri = sip:name at domain
>> Aug 23 16:56:03 opensips-test /usr/sbin/opensips[18629]: [ON_BRANCH] 
>> Current branch data: idx = 1, uri = <null>
>> Aug 23 16:56:03 opensips-test /usr/sbin/opensips[18629]: [ON_BRANCH] 
>> Current branch data: idx = 2, uri = <null>
>>
>> So, i can't access to any data of any branch except first one. Even more:
>>
>> I need to delete branches after they fires fr_times (local 408 is 
>> received). To do that, i trying to call remove_branch($T_branch_idx) 
>> from failure_route:
>>
>> ...
>> failure_route[ON_FAIL] {
>> ...
>> if (t_check_status("408")) {
>> ...
>> remove_branch($T_branch_idx);
>> route(LOOKUP);
>> }
>> ...
>> }
>>
>> but after that, all branches still on their places. So, can i somehow 
>> delete these "expired" branches?
>>
>> version: opensips 2.1.4 (x86_64/linux)
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> -------- ?????????? ????????????? ????????? --------
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From achalkov at ya.ru  Tue Aug 23 18:21:56 2016
From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Tue, 23 Aug 2016 19:21:56 +0300
Subject: [OpenSIPS-Users] accessing to branch data
In-Reply-To: <052c8628-67b8-967e-163a-8b592221c061@opensips.org>
References: <10091471966304@web11m.yandex.ru>
	<052c8628-67b8-967e-163a-8b592221c061@opensips.org>
Message-ID: <175171471969316@web4g.yandex.ru>

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From liviu at opensips.org  Tue Aug 23 18:35:18 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 23 Aug 2016 19:35:18 +0300
Subject: [OpenSIPS-Users] Listening on all interfaces on Opensips 2.2
In-Reply-To: <CAP9sNzGXe+99FaVmJqobh6ktb0re7U+=vg=bz46-oT69T2Ke6A@mail.gmail.com>
References: <CAP9sNzGXe+99FaVmJqobh6ktb0re7U+=vg=bz46-oT69T2Ke6A@mail.gmail.com>
Message-ID: <7d14a0aa-a6f8-e1db-f851-b34a6efa7f00@opensips.org>

Hi, Jon!

The "eth0:5060" syntax has been deprecated along with the transport 
layer rework.

In order to achieve the same effect as before, you simply have to add 
one more line to your script for each transport you choose to support:

udp:eth0:5060

tcp:eth0:5060

hep:eth0:9060

...

Docs for the "listen" directive are indeed broken and I'll get them 
fixed asap. Thanks!

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.08.2016 04:19, Jon Abrams wrote:
> Hello list,
>
> I've been noticing the listen configuration seems to work differently 
> on the latest 2.2.X versions.
>
> I'm trying to listen on all addresses (essentially 0.0.0.0) without 
> having to manually add each ip line by line to the script.
>
> In the old days you could just omit the listen and set the port. 
> However I need to leave a listen directive for HEP.
>
> Mentioned in the 2.2 docs, listen=eth0:5060, gives a vague generic 
> script error. listen=udp:eth0:5060 works. Are the docs out of date on 
> some of these? disable_tcp and port seem to give generic script errors 
> despite being still listed in the docs
>
> Best Regards,
>
> Jon
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From adrian.fretwell at topgreen.co.uk  Tue Aug 23 23:50:44 2016
From: adrian.fretwell at topgreen.co.uk (Adrian Fretwell)
Date: Tue, 23 Aug 2016 22:50:44 +0100
Subject: [OpenSIPS-Users] B2BUA parameter parsing
Message-ID: <e6aa655f-7b2e-776f-da9f-09e29f094dc2@topgreen.co.uk>

Can anyone help?  Setting up a call from the middle with B2BUA I pass 
RURI as a parameter via MI interface. RURI has some header values:

param1 = sip:alice at abc.net?header1=qaz&header2=wsx

param2 = sip:bob at abc.net?header1=qaz&header2=wsx

Problem is B2BUA puts the whole parameter string in the From and To 
headers where really only the parameter string up to the ? is required 
and valid for the From and To headers.

Will I need to patch the source code in b2b_entities to solve this?  I'm 
guessing it is when the t_client structure gets populated, can someone 
put me on the right track please?

Kind regards,

Adrian Fretwell
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From millennium.bug at gmail.com  Wed Aug 24 07:16:26 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Wed, 24 Aug 2016 10:16:26 +0500
Subject: [OpenSIPS-Users] dispatcher module call limit
Message-ID: <CAArAGK=ow9zqg0+KnehY2oKcKEpRXENS1Y5YH2fPh=arqQcSGQ@mail.gmail.com>

Hi guys,

Can we use dispatcher module to restrict the number of INVITEs relayed to a
specific destination?

Regards,
Owais
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From bogdan at opensips.org  Wed Aug 24 09:33:47 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Aug 2016 10:33:47 +0300
Subject: [OpenSIPS-Users] B2BUA RURI with headers
In-Reply-To: <0f952149-22a2-de92-ac5d-0f077d1be377@topgreen.co.uk>
References: <0f952149-22a2-de92-ac5d-0f077d1be377@topgreen.co.uk>
Message-ID: <d41c8073-a3ee-b991-ef5d-9df981c7197a@opensips.org>

Hi Adrian,

Do you have a pcap with the SIP traffic ? also the opensips log (with 
the parsing error) will help.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 21.08.2016 11:05, Adrian Fretwell wrote:
>
> Hello,
>
> I have put together a very simple scenario that bridges two clients, 
> it is called from the MI (datagram) interface.  I want the two SIP 
> messages that go out to the clients to contain some header values, so 
> I specify these in the two parameters sent in the MI call.  Please 
> note, I am using the bash shell, /dev/udp... does not actually exist:
>
> /bin/echo -e 
> ":b2b_trigger_scenario:\nahf1\nsip:alice at abc.net?header1=qaz&header2=wsx\nbob at abc.net?header1=qaz&header2=wsx" 
> > /dev/udp/192.168.126.72/9061
>
> This works but the problem I have is that the B2BUA includes the 
> header part of the RURI in both the To and From headers.  When the UAC 
> responds with a 200 OK, B2BUA sees the extra information in the To and 
> From headers and throws a "unable to parse", "bad header" error.
>
> Is there any way to stop B2BUA from including the RURI header 
> information in the To and From headers?  Can headers be specified 
> separately in the <destination> xml tag?
>
> Kind regards,
>
> Adrian Fretwell
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From achalkov at yandex.ru  Tue Aug 23 16:50:31 2016
From: achalkov at yandex.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Tue, 23 Aug 2016 17:50:31 +0300
Subject: [OpenSIPS-Users] accessing to branch data
In-Reply-To: 159596311794961631
References: <727051471962045@web8m.yandex.ru>
Message-ID: <702701471963831@web30h.yandex.ru>

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From bogdan at opensips.org  Wed Aug 24 10:17:19 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Aug 2016 11:17:19 +0300
Subject: [OpenSIPS-Users] B2BUA parameter parsing
In-Reply-To: <e6aa655f-7b2e-776f-da9f-09e29f094dc2@topgreen.co.uk>
References: <e6aa655f-7b2e-776f-da9f-09e29f094dc2@topgreen.co.uk>
Message-ID: <549592a8-8f91-1320-b8e9-25b61a8bfb1d@opensips.org>

Please see my previous reply:
http://lists.opensips.org/pipermail/users/2016-August/035349.html

Regards

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.08.2016 00:50, Adrian Fretwell wrote:
>
> Can anyone help?  Setting up a call from the middle with B2BUA I pass 
> RURI as a parameter via MI interface.  RURI has some header values:
>
> param1 = sip:alice at abc.net?header1=qaz&header2=wsx
>
> param2 = sip:bob at abc.net?header1=qaz&header2=wsx
>
> Problem is B2BUA puts the whole parameter string in the From and To 
> headers where really only the parameter string up to the ? is required 
> and valid for the From and To headers.
>
> Will I need to patch the source code in b2b_entities to solve this?  
> I'm guessing it is when the t_client structure gets populated, can 
> someone put me on the right track please?
>
> Kind regards,
>
> Adrian Fretwell
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Aug 24 11:20:00 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Aug 2016 12:20:00 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
Message-ID: <77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>

Hi Jeff,

You mean you do not get the Attributes text area at all in the Edit 
section ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 23.08.2016 00:56, Jeff Wilkie wrote:
> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP 5.0
>
> It appears that when modifying the CP for Dialplan attributes as text, 
> they do not appear in the edit section once applied.  The attributes 
> do show up in the view list of dial plans but if you go back to edit 
> the attrs again, they are not there.  I dot not have the same problem 
> when using attrs as checkboxes.  Is there more code in the local files 
> that require editing to display these values?  Also, if an entry in 
> the Dialplan was once a checkbox value but I've not converted the mode 
> to text, the value does not show up in the text box to be edited.
>
> # Attention : advanced options !!
>
>
>
>  //database tables
>
>  $config->table_dialplan = "dialplan";
>
>  $config->results_per_page = 20;
>
>  $config->results_page_range = 5;
>
>
>  $config->attrs_cb=array(
>
>                                       // name , description
>
>                                       array("a","Descr a"),
>
>                                       array("b","Descr b"),
>
>                                       array("c","Descr c"),
>
>                                       array("d","Descr d"),
>
>                                       array("e","Descr e"),
>
>                                       array("f","Descr f"),
>
>                                       );
>
>  $config->cb_per_row = 3;
>
>  $talk_to_this_assoc_id = 1 ;
>
>
> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>
> // 0 - checkboxes
>
> // 1 - text
>
> $dialplan_attributes_mode = 1  ;
>
>
> ?>
>
>
>
>
> Jeff
>
>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Aug 24 11:22:01 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Aug 2016 12:22:01 +0300
Subject: [OpenSIPS-Users] dispatcher module call limit
In-Reply-To: <CAArAGK=ow9zqg0+KnehY2oKcKEpRXENS1Y5YH2fPh=arqQcSGQ@mail.gmail.com>
References: <CAArAGK=ow9zqg0+KnehY2oKcKEpRXENS1Y5YH2fPh=arqQcSGQ@mail.gmail.com>
Message-ID: <f3ee808d-5a2a-9e27-bf2a-98e4647e9491@opensips.org>

Hi Owais,

The dispatcher module is just dispatching traffic between multiple 
destinations. The restrict (CPS limitation) traffic use the ratelimit 
module:
     http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.08.2016 08:16, Owais Ahmad wrote:
> Hi guys,
>
> Can we use dispatcher module to restrict the number of INVITEs relayed 
> to a specific destination?
>
> Regards,
> Owais
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From millennium.bug at gmail.com  Wed Aug 24 12:11:54 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Wed, 24 Aug 2016 15:11:54 +0500
Subject: [OpenSIPS-Users] dispatcher module call limit
In-Reply-To: <f3ee808d-5a2a-9e27-bf2a-98e4647e9491@opensips.org>
References: <CAArAGK=ow9zqg0+KnehY2oKcKEpRXENS1Y5YH2fPh=arqQcSGQ@mail.gmail.com>
	<f3ee808d-5a2a-9e27-bf2a-98e4647e9491@opensips.org>
Message-ID: <CAArAGKm7=jdWXd1jTUgE-bNPbq4CAyu=HdJNu1d-_Bp+1Ww01w@mail.gmail.com>

Hello Bogdan,

I should have specified earlier that I want to limit the *concurrent *calls
per destination.
I don't think ratelimit can be used for that.

I want to have calls get dropped with a 503 at the dispatcher server if a
defined concurrent call limit is met; not traversing any nodes deeper down
the network.



On Wed, Aug 24, 2016 at 2:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Owais,
>
> The dispatcher module is just dispatching traffic between multiple
> destinations. The restrict (CPS limitation) traffic use the ratelimit
> module:
>     http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 24.08.2016 08:16, Owais Ahmad wrote:
>
> Hi guys,
>
> Can we use dispatcher module to restrict the number of INVITEs relayed to
> a specific destination?
>
> Regards,
> Owais
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From bogdan at opensips.org  Wed Aug 24 12:40:31 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Aug 2016 13:40:31 +0300
Subject: [OpenSIPS-Users] dispatcher module call limit
In-Reply-To: <CAArAGKm7=jdWXd1jTUgE-bNPbq4CAyu=HdJNu1d-_Bp+1Ww01w@mail.gmail.com>
References: <CAArAGK=ow9zqg0+KnehY2oKcKEpRXENS1Y5YH2fPh=arqQcSGQ@mail.gmail.com>
	<f3ee808d-5a2a-9e27-bf2a-98e4647e9491@opensips.org>
	<CAArAGKm7=jdWXd1jTUgE-bNPbq4CAyu=HdJNu1d-_Bp+1Ww01w@mail.gmail.com>
Message-ID: <be902fa8-8c20-5c8c-3dfd-facb69f48fb3@opensips.org>

OK, in this case you should use the dialog module with call profile support.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.08.2016 13:11, Owais Ahmad wrote:
> Hello Bogdan,
>
> I should have specified earlier that I want to limit the *concurrent 
> *calls per destination.
> I don't think ratelimit can be used for that.
>
> I want to have calls get dropped with a 503 at the dispatcher server 
> if a defined concurrent call limit is met; not traversing any nodes 
> deeper down the network.
>
>
>
> On Wed, Aug 24, 2016 at 2:22 PM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Owais,
>
>     The dispatcher module is just dispatching traffic between multiple
>     destinations. The restrict (CPS limitation) traffic use the
>     ratelimit module:
>     http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html
>     <http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html>
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://www.opensips-solutions.com>
>
>     On 24.08.2016 08:16, Owais Ahmad wrote:
>>     Hi guys,
>>     Can we use dispatcher module to restrict the number of INVITEs
>>     relayed to a specific destination?
>>     Regards,
>>     Owais
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
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From jwilkie at usipcom.com  Wed Aug 24 16:46:38 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Wed, 24 Aug 2016 10:46:38 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
Message-ID: <CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>

Yes, The text box is there but when you place a value in a preexisting dial
plan, the result in the DB is just the value in the text attribute field
with no dial plan information.  By editing or adding an attribute it kills
the rest of the dial plan that you were editing and you cannot add the dial
plan information back in via the CP as long as an attribute in text form is
added.  The same does not happen if using check box.  I can provide pics if
you'd like.  The only var changed is the

$dialplan_attributes_mode = 1

Jeff Wilkie
Chief Technology Officer
US IP Communications
919.297.1057


*"This e-mail communication and any attachments may contain confidential
and privileged information and is for use by the designated addressee(s)
named above only. Any files transmitted with it are confidential and
intended solely for the use of the individual to whom it is addressed. Any
views or opinions presented are solely those of the author and do not
necessarily represent those of USIPCOM, LLC. If you are not the intended
addressee, you are hereby notified that you have received this
communication in error and that any use or reproduction of this email or
its contents is strictly prohibited and may be unlawful. If you have
received this communication in error, please notify us immediately by
replying to this message and deleting it from your computer. Thank you". *

On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jeff,
>
> You mean you do not get the Attributes text area at all in the Edit
> section ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 23.08.2016 00:56, Jeff Wilkie wrote:
>
> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP 5.0
>
> It appears that when modifying the CP for Dialplan attributes as text,
> they do not appear in the edit section once applied.  The attributes do
> show up in the view list of dial plans but if you go back to edit the attrs
> again, they are not there.  I dot not have the same problem when using
> attrs as checkboxes.  Is there more code in the local files that require
> editing to display these values?  Also, if an entry in the Dialplan was
> once a checkbox value but I've not converted the mode to text, the value
> does not show up in the text box to be edited.
>
> # Attention : advanced options !!
>
>
>
>  //database tables
>
>  $config->table_dialplan = "dialplan";
>
>  $config->results_per_page = 20;
>
>  $config->results_page_range = 5;
>
>
>  $config->attrs_cb=array(
>
>                                         // name , description
>
>                                         array("a","Descr a"),
>
>                                         array("b","Descr b"),
>
>                                         array("c","Descr c"),
>
>                                         array("d","Descr d"),
>
>                                         array("e","Descr e"),
>
>                                         array("f","Descr f"),
>
>                                         );
>
>  $config->cb_per_row = 3;
>
>  $talk_to_this_assoc_id = 1 ;
>
>
> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>
> // 0 - checkboxes
>
> // 1 - text
>
> $dialplan_attributes_mode = 1  ;
>
>
> ?>
>
>
>
> Jeff
>
>
>
>
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From Agalya_Ramachandran at comcast.com  Wed Aug 24 17:31:56 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 24 Aug 2016 15:31:56 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
Message-ID: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>

Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya
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From bogdan at opensips.org  Wed Aug 24 17:35:03 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Aug 2016 18:35:03 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
Message-ID: <20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>

Hi Jeff,

I spotted the issue and fixed it on GIT repo. Please give it a try and 
let me know if it works for you too.

Thanks and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.08.2016 17:46, Jeff Wilkie wrote:
> Yes, The text box is there but when you place a value in a preexisting 
> dial plan, the result in the DB is just the value in the text 
> attribute field with no dial plan information.  By editing or adding 
> an attribute it kills the rest of the dial plan that you were editing 
> and you cannot add the dial plan information back in via the CP as 
> long as an attribute in text form is added.  The same does not happen 
> if using check box.  I can provide pics if you'd like.  The only var 
> changed is the
>>
>> $dialplan_attributes_mode = 1
>>
> Jeff Wilkie
> Chief Technology Officer
> US IP Communications
> 919.297.1057
>
>
> /"This e-mail communication and any attachments may contain 
> confidential and privileged information and is for use by the 
> designated addressee(s) named above only. Any files transmitted with 
> it are confidential and intended solely for the use of the individual 
> to whom it is addressed. Any views or opinions presented are solely 
> those of the author and do not necessarily represent those of USIPCOM, 
> LLC. If you are not the intended addressee, you are hereby notified 
> that you have received this communication in error and that any use or 
> reproduction of this email or its contents is strictly prohibited and 
> may be unlawful. If you have received this communication in error, 
> please notify us immediately by replying to this message and deleting 
> it from your computer. Thank you". /
>
> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Jeff,
>
>     You mean you do not get the Attributes text area at all in the
>     Edit section ?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://www.opensips-solutions.com>
>
>     On 23.08.2016 00:56, Jeff Wilkie wrote:
>>     New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with
>>     CP 5.0
>>     It appears that when modifying the CP for Dialplan attributes as
>>     text, they do not appear in the edit section once applied.  The
>>     attributes do show up in the view list of dial plans but if you
>>     go back to edit the attrs again, they are not there.  I dot not
>>     have the same problem when using attrs as checkboxes.  Is there
>>     more code in the local files that require editing to display
>>     these values?  Also, if an entry in the Dialplan was once a
>>     checkbox value but I've not converted the mode to text, the value
>>     does not show up in the text box to be edited.
>>
>>     # Attention : advanced options !!
>>
>>      //database tables
>>
>>      $config->table_dialplan = "dialplan";
>>
>>      $config->results_per_page = 20;
>>
>>      $config->results_page_range = 5;
>>
>>      $config->attrs_cb=array(
>>
>>                                           // name , description
>>
>>                                           array("a","Descr a"),
>>
>>                                           array("b","Descr b"),
>>
>>                                           array("c","Descr c"),
>>
>>                                           array("d","Descr d"),
>>
>>                                           array("e","Descr e"),
>>
>>                                           array("f","Descr f"),
>>
>>                                           );
>>
>>      $config->cb_per_row = 3;
>>
>>      $talk_to_this_assoc_id = 1 ;
>>
>>     // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>
>>     // 0 - checkboxes
>>
>>     // 1 - text
>>
>>     $dialplan_attributes_mode = 1  ;
>>
>>     ?>
>>
>>     Jeff
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
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From jwilkie at usipcom.com  Wed Aug 24 22:25:39 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Wed, 24 Aug 2016 16:25:39 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
Message-ID: <CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>

Ok.  The value now displays but if you modify the value and hit save, it
kills the regex value upon saving.  Also, you cannot add the regex value
back into the row.  All other values appear to be kept.

Thanks

Jeff

On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jeff,
>
> I spotted the issue and fixed it on GIT repo. Please give it a try and let
> me know if it works for you too.
>
> Thanks and regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 24.08.2016 17:46, Jeff Wilkie wrote:
>
> Yes, The text box is there but when you place a value in a preexisting
> dial plan, the result in the DB is just the value in the text attribute
> field with no dial plan information.  By editing or adding an attribute it
> kills the rest of the dial plan that you were editing and you cannot add
> the dial plan information back in via the CP as long as an attribute in
> text form is added.  The same does not happen if using check box.  I can
> provide pics if you'd like.  The only var changed is the
>
> $dialplan_attributes_mode = 1
>
> Jeff Wilkie
> Chief Technology Officer
> US IP Communications
> 919.297.1057
>
>
> *"This e-mail communication and any attachments may contain confidential
> and privileged information and is for use by the designated addressee(s)
> named above only. Any files transmitted with it are confidential and
> intended solely for the use of the individual to whom it is addressed. Any
> views or opinions presented are solely those of the author and do not
> necessarily represent those of USIPCOM, LLC. If you are not the intended
> addressee, you are hereby notified that you have received this
> communication in error and that any use or reproduction of this email or
> its contents is strictly prohibited and may be unlawful. If you have
> received this communication in error, please notify us immediately by
> replying to this message and deleting it from your computer. Thank you". *
>
> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi Jeff,
>>
>> You mean you do not get the Attributes text area at all in the Edit
>> section ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>
>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP 5.0
>> It appears that when modifying the CP for Dialplan attributes as text,
>> they do not appear in the edit section once applied.  The attributes do
>> show up in the view list of dial plans but if you go back to edit the attrs
>> again, they are not there.  I dot not have the same problem when using
>> attrs as checkboxes.  Is there more code in the local files that require
>> editing to display these values?  Also, if an entry in the Dialplan was
>> once a checkbox value but I've not converted the mode to text, the value
>> does not show up in the text box to be edited.
>>
>> # Attention : advanced options !!
>>
>>  //database tables
>>
>>  $config->table_dialplan = "dialplan";
>>
>>  $config->results_per_page = 20;
>>
>>  $config->results_page_range = 5;
>>
>>  $config->attrs_cb=array(
>>
>>                                         // name , description
>>
>>                                         array("a","Descr a"),
>>
>>                                         array("b","Descr b"),
>>
>>                                         array("c","Descr c"),
>>
>>                                         array("d","Descr d"),
>>
>>                                         array("e","Descr e"),
>>
>>                                         array("f","Descr f"),
>>
>>                                         );
>>
>>  $config->cb_per_row = 3;
>>
>>  $talk_to_this_assoc_id = 1 ;
>>
>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>
>> // 0 - checkboxes
>>
>> // 1 - text
>>
>> $dialplan_attributes_mode = 1  ;
>>
>> ?>
>> Jeff
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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From bogdan at opensips.org  Wed Aug 24 23:15:53 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 25 Aug 2016 00:15:53 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
Message-ID: <fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>

Jeff,

You mean "Matching Regular Expression", right ? upon performing an edit 
+ save where you only changed the attrs, the "Matching Regular 
Expression" gets to empty ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.08.2016 23:25, Jeff Wilkie wrote:
> Ok.  The value now displays but if you modify the value and hit save, 
> it kills the regex value upon saving.  Also, you cannot add the regex 
> value back into the row.  All other values appear to be kept.
>
> Thanks
>
> Jeff
>
> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Jeff,
>
>     I spotted the issue and fixed it on GIT repo. Please give it a try
>     and let me know if it works for you too.
>
>     Thanks and regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://www.opensips-solutions.com>
>
>     On 24.08.2016 17:46, Jeff Wilkie wrote:
>>     Yes, The text box is there but when you place a value in a
>>     preexisting dial plan, the result in the DB is just the value in
>>     the text attribute field with no dial plan information.  By
>>     editing or adding an attribute it kills the rest of the dial plan
>>     that you were editing and you cannot add the dial plan
>>     information back in via the CP as long as an attribute in text
>>     form is added.  The same does not happen if using check box.  I
>>     can provide pics if you'd like.  The only var changed is the
>>>
>>>     $dialplan_attributes_mode = 1
>>>
>>     Jeff Wilkie
>>     Chief Technology Officer
>>     US IP Communications 919.297.1057 <tel:919.297.1057> /"This
>>     e-mail communication and any attachments may contain confidential
>>     and privileged information and is for use by the designated
>>     addressee(s) named above only. Any files transmitted with it are
>>     confidential and intended solely for the use of the individual to
>>     whom it is addressed. Any views or opinions presented are solely
>>     those of the author and do not necessarily represent those of
>>     USIPCOM, LLC. If you are not the intended addressee, you are
>>     hereby notified that you have received this communication in
>>     error and that any use or reproduction of this email or its
>>     contents is strictly prohibited and may be unlawful. If you have
>>     received this communication in error, please notify us
>>     immediately by replying to this message and deleting it from your
>>     computer. Thank you". /
>>     On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu
>>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>         Hi Jeff, You mean you do not get the Attributes text area at
>>         all in the Edit section ? Regards,
>>
>>         Bogdan-Andrei Iancu
>>         OpenSIPS Founder and Developer
>>         http://www.opensips-solutions.com
>>         <http://www.opensips-solutions.com>
>>
>>         On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>         New small issue with DIALPLAN in the CP.  Running OSIPS 1.10
>>>         with CP 5.0
>>>         It appears that when modifying the CP for Dialplan
>>>         attributes as text, they do not appear in the edit section
>>>         once applied.  The attributes do show up in the view list of
>>>         dial plans but if you go back to edit the attrs again, they
>>>         are not there.  I dot not have the same problem when using
>>>         attrs as checkboxes.  Is there more code in the local files
>>>         that require editing to display these values?  Also, if an
>>>         entry in the Dialplan was once a checkbox value but I've not
>>>         converted the mode to text, the value does not show up in
>>>         the text box to be edited.
>>>
>>>         # Attention : advanced options !!
>>>
>>>          //database tables
>>>
>>>          $config->table_dialplan = "dialplan";
>>>
>>>          $config->results_per_page = 20;
>>>
>>>          $config->results_page_range = 5;
>>>
>>>          $config->attrs_cb=array(
>>>
>>>                                               // name , description
>>>
>>>                                               array("a","Descr a"),
>>>
>>>                                               array("b","Descr b"),
>>>
>>>                                               array("c","Descr c"),
>>>
>>>                                               array("d","Descr d"),
>>>
>>>                                               array("e","Descr e"),
>>>
>>>                                               array("f","Descr f"),
>>>
>>>                                               );
>>>
>>>          $config->cb_per_row = 3;
>>>
>>>          $talk_to_this_assoc_id = 1 ;
>>>
>>>         // Dialplan - Add/Edit new Translation Rule - Attributes
>>>         input type
>>>
>>>         // 0 - checkboxes
>>>
>>>         // 1 - text
>>>
>>>         $dialplan_attributes_mode = 1  ;
>>>
>>>         ?>
>>>
>>>         Jeff
>>>
>>>         _______________________________________________
>>>         Users mailing list
>>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>         <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
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From jwilkie at usipcom.com  Wed Aug 24 23:30:36 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Wed, 24 Aug 2016 17:30:36 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
Message-ID: <CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>

Yes.  After any edits followed by a save with the attribute field active as
text causes "Matching Regular Expression" to be deleted.

Thanks

Jeff

On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Jeff,
>
> You mean "Matching Regular Expression", right ? upon performing an edit +
> save where you only changed the attrs, the "Matching Regular Expression"
> gets to empty ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 24.08.2016 23:25, Jeff Wilkie wrote:
>
> Ok.  The value now displays but if you modify the value and hit save, it
> kills the regex value upon saving.  Also, you cannot add the regex value
> back into the row.  All other values appear to be kept.
>
> Thanks
>
> Jeff
>
> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <bogdan at opensips.org
> > wrote:
>
>> Hi Jeff,
>>
>> I spotted the issue and fixed it on GIT repo. Please give it a try and
>> let me know if it works for you too.
>>
>> Thanks and regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 24.08.2016 17:46, Jeff Wilkie wrote:
>>
>> Yes, The text box is there but when you place a value in a preexisting
>> dial plan, the result in the DB is just the value in the text attribute
>> field with no dial plan information.  By editing or adding an attribute it
>> kills the rest of the dial plan that you were editing and you cannot add
>> the dial plan information back in via the CP as long as an attribute in
>> text form is added.  The same does not happen if using check box.  I can
>> provide pics if you'd like.  The only var changed is the
>>
>> $dialplan_attributes_mode = 1
>>
>> Jeff Wilkie
>> Chief Technology Officer
>> US IP Communications 919.297.1057 *"This e-mail communication and any
>> attachments may contain confidential and privileged information and is for
>> use by the designated addressee(s) named above only. Any files transmitted
>> with it are confidential and intended solely for the use of the individual
>> to whom it is addressed. Any views or opinions presented are solely those
>> of the author and do not necessarily represent those of USIPCOM, LLC. If
>> you are not the intended addressee, you are hereby notified that you have
>> received this communication in error and that any use or reproduction of
>> this email or its contents is strictly prohibited and may be unlawful. If
>> you have received this communication in error, please notify us immediately
>> by replying to this message and deleting it from your computer. Thank you".
>> *
>> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <bogdan at opensips.org
>> > wrote:
>>>
>>> Hi Jeff, You mean you do not get the Attributes text area at all in the
>>> Edit section ? Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>
>>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP 5.0
>>> It appears that when modifying the CP for Dialplan attributes as text,
>>> they do not appear in the edit section once applied.  The attributes do
>>> show up in the view list of dial plans but if you go back to edit the attrs
>>> again, they are not there.  I dot not have the same problem when using
>>> attrs as checkboxes.  Is there more code in the local files that require
>>> editing to display these values?  Also, if an entry in the Dialplan was
>>> once a checkbox value but I've not converted the mode to text, the value
>>> does not show up in the text box to be edited.
>>>
>>> # Attention : advanced options !!
>>>
>>>  //database tables
>>>
>>>  $config->table_dialplan = "dialplan";
>>>
>>>  $config->results_per_page = 20;
>>>
>>>  $config->results_page_range = 5;
>>>
>>>  $config->attrs_cb=array(
>>>
>>>                                         // name , description
>>>
>>>                                         array("a","Descr a"),
>>>
>>>                                         array("b","Descr b"),
>>>
>>>                                         array("c","Descr c"),
>>>
>>>                                         array("d","Descr d"),
>>>
>>>                                         array("e","Descr e"),
>>>
>>>                                         array("f","Descr f"),
>>>
>>>                                         );
>>>
>>>  $config->cb_per_row = 3;
>>>
>>>  $talk_to_this_assoc_id = 1 ;
>>>
>>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>>
>>> // 0 - checkboxes
>>>
>>> // 1 - text
>>>
>>> $dialplan_attributes_mode = 1  ;
>>>
>>> ?>
>>> Jeff
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
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From jwilkie at usipcom.com  Thu Aug 25 00:22:38 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Wed, 24 Aug 2016 18:22:38 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
Message-ID: <CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>

Here is what it looks like to my db when I create a DialPlan

  20 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='877'

  20 Query PREPARE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
FROM 'INSERT INTO dialplan\n (dpid, pr, match_op, match_exp, match_flags,
subst_exp, \n repl_exp, attrs) VALUES \n (36, 0,1,
\'877\',0,\'\',\'\',\'.30\')'

  20 Prepare INSERT INTO dialplan

(dpid, pr, match_op, match_exp, match_flags, subst_exp,

repl_exp, attrs) VALUES

(36, 0,1, '877',0,'','','.30')

  20 Query EXECUTE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74

  20 Execute INSERT INTO dialplan

(dpid, pr, match_op, match_exp, match_flags, subst_exp,

repl_exp, attrs) VALUES

(36, 0,1, '877',0,'','','.30')

This is what happens when I edit that same DialPlan

  23 Query select * from dialplan where id='9030'

  23 Quit

  24 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='' AND
id!=9030

  24 Query PREPARE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
FROM 'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp =\'\',
match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs= \'.30\' WHERE
id=9030'

  24 Prepare UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030

  24 Query EXECUTE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2

  24 Execute UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030

  24 Query DEALLOCATE PREPARE
MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2

  24 Quit


You can see that the container for match_ex is not getting passed to the
'24' SELECT statement which ends in a 0 results query apparently.


Thanks

Jeff

On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie <jwilkie at usipcom.com> wrote:

> Yes.  After any edits followed by a save with the attribute field active
> as text causes "Matching Regular Expression" to be deleted.
>
> Thanks
>
> Jeff
>
> On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Jeff,
>>
>> You mean "Matching Regular Expression", right ? upon performing an edit +
>> save where you only changed the attrs, the "Matching Regular Expression"
>> gets to empty ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 24.08.2016 23:25, Jeff Wilkie wrote:
>>
>> Ok.  The value now displays but if you modify the value and hit save, it
>> kills the regex value upon saving.  Also, you cannot add the regex value
>> back into the row.  All other values appear to be kept.
>>
>> Thanks
>>
>> Jeff
>>
>> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <
>> bogdan at opensips.org> wrote:
>>
>>> Hi Jeff,
>>>
>>> I spotted the issue and fixed it on GIT repo. Please give it a try and
>>> let me know if it works for you too.
>>>
>>> Thanks and regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>
>>> Yes, The text box is there but when you place a value in a preexisting
>>> dial plan, the result in the DB is just the value in the text attribute
>>> field with no dial plan information.  By editing or adding an attribute it
>>> kills the rest of the dial plan that you were editing and you cannot add
>>> the dial plan information back in via the CP as long as an attribute in
>>> text form is added.  The same does not happen if using check box.  I can
>>> provide pics if you'd like.  The only var changed is the
>>>
>>> $dialplan_attributes_mode = 1
>>>
>>> Jeff Wilkie
>>> Chief Technology Officer
>>> US IP Communications 919.297.1057 *"This e-mail communication and any
>>> attachments may contain confidential and privileged information and is for
>>> use by the designated addressee(s) named above only. Any files transmitted
>>> with it are confidential and intended solely for the use of the individual
>>> to whom it is addressed. Any views or opinions presented are solely those
>>> of the author and do not necessarily represent those of USIPCOM, LLC. If
>>> you are not the intended addressee, you are hereby notified that you have
>>> received this communication in error and that any use or reproduction of
>>> this email or its contents is strictly prohibited and may be unlawful. If
>>> you have received this communication in error, please notify us immediately
>>> by replying to this message and deleting it from your computer. Thank you".
>>> *
>>> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <
>>> bogdan at opensips.org> wrote:
>>>>
>>>> Hi Jeff, You mean you do not get the Attributes text area at all in the
>>>> Edit section ? Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>
>>>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP 5.0
>>>> It appears that when modifying the CP for Dialplan attributes as text,
>>>> they do not appear in the edit section once applied.  The attributes do
>>>> show up in the view list of dial plans but if you go back to edit the attrs
>>>> again, they are not there.  I dot not have the same problem when using
>>>> attrs as checkboxes.  Is there more code in the local files that require
>>>> editing to display these values?  Also, if an entry in the Dialplan was
>>>> once a checkbox value but I've not converted the mode to text, the value
>>>> does not show up in the text box to be edited.
>>>>
>>>> # Attention : advanced options !!
>>>>
>>>>  //database tables
>>>>
>>>>  $config->table_dialplan = "dialplan";
>>>>
>>>>  $config->results_per_page = 20;
>>>>
>>>>  $config->results_page_range = 5;
>>>>
>>>>  $config->attrs_cb=array(
>>>>
>>>>                                         // name , description
>>>>
>>>>                                         array("a","Descr a"),
>>>>
>>>>                                         array("b","Descr b"),
>>>>
>>>>                                         array("c","Descr c"),
>>>>
>>>>                                         array("d","Descr d"),
>>>>
>>>>                                         array("e","Descr e"),
>>>>
>>>>                                         array("f","Descr f"),
>>>>
>>>>                                         );
>>>>
>>>>  $config->cb_per_row = 3;
>>>>
>>>>  $talk_to_this_assoc_id = 1 ;
>>>>
>>>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>>>
>>>> // 0 - checkboxes
>>>>
>>>> // 1 - text
>>>>
>>>> $dialplan_attributes_mode = 1  ;
>>>>
>>>> ?>
>>>> Jeff
>>>>
>>>> _______________________________________________
>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>
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From jwilkie at usipcom.com  Thu Aug 25 01:06:56 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Wed, 24 Aug 2016 19:06:56 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
Message-ID: <CANFkrT3uk82=b5=dF+ezmGPX0R09G16yDSqNVsHskKPx+oHHgg@mail.gmail.com>

Disregard the statement I made about the select statement. This is where
you're checking for duplicates. But I do notice the regex is not present
here either.

Thanks

Jeff Wilkie
USIP Communications

On Aug 24, 2016 6:22 PM, "Jeff Wilkie" <jwilkie at usipcom.com> wrote:

> Here is what it looks like to my db when I create a DialPlan
>
>   20 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='877'
>
>   20 Query PREPARE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
> FROM 'INSERT INTO dialplan\n (dpid, pr, match_op, match_exp, match_flags,
> subst_exp, \n repl_exp, attrs) VALUES \n (36, 0,1,
> \'877\',0,\'\',\'\',\'.30\')'
>
>   20 Prepare INSERT INTO dialplan
>
> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>
> repl_exp, attrs) VALUES
>
> (36, 0,1, '877',0,'','','.30')
>
>   20 Query EXECUTE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>
>   20 Execute INSERT INTO dialplan
>
> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>
> repl_exp, attrs) VALUES
>
> (36, 0,1, '877',0,'','','.30')
>
> This is what happens when I edit that same DialPlan
>
>   23 Query select * from dialplan where id='9030'
>
>   23 Quit
>
>   24 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='' AND
> id!=9030
>
>   24 Query PREPARE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
> FROM 'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp =\'\',
> match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs= \'.30\' WHERE
> id=9030'
>
>   24 Prepare UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
> ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030
>
>   24 Query EXECUTE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>
>   24 Execute UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
> ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030
>
>   24 Query DEALLOCATE PREPARE MDB2_STATEMENT_mysql_
> 880efa45d78ebff7cbf5f617428df4d2
>
>   24 Quit
>
>
> You can see that the container for match_ex is not getting passed to the
> '24' SELECT statement which ends in a 0 results query apparently.
>
>
> Thanks
>
> Jeff
>
> On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie <jwilkie at usipcom.com> wrote:
>
>> Yes.  After any edits followed by a save with the attribute field active
>> as text causes "Matching Regular Expression" to be deleted.
>>
>> Thanks
>>
>> Jeff
>>
>> On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu <bogdan at opensips.org
>> > wrote:
>>
>>> Jeff,
>>>
>>> You mean "Matching Regular Expression", right ? upon performing an edit
>>> + save where you only changed the attrs, the "Matching Regular
>>> Expression" gets to empty ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 24.08.2016 23:25, Jeff Wilkie wrote:
>>>
>>> Ok.  The value now displays but if you modify the value and hit save, it
>>> kills the regex value upon saving.  Also, you cannot add the regex value
>>> back into the row.  All other values appear to be kept.
>>>
>>> Thanks
>>>
>>> Jeff
>>>
>>> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <
>>> bogdan at opensips.org> wrote:
>>>
>>>> Hi Jeff,
>>>>
>>>> I spotted the issue and fixed it on GIT repo. Please give it a try and
>>>> let me know if it works for you too.
>>>>
>>>> Thanks and regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>> On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>>
>>>> Yes, The text box is there but when you place a value in a preexisting
>>>> dial plan, the result in the DB is just the value in the text attribute
>>>> field with no dial plan information.  By editing or adding an attribute it
>>>> kills the rest of the dial plan that you were editing and you cannot add
>>>> the dial plan information back in via the CP as long as an attribute in
>>>> text form is added.  The same does not happen if using check box.  I can
>>>> provide pics if you'd like.  The only var changed is the
>>>>
>>>> $dialplan_attributes_mode = 1
>>>>
>>>> Jeff Wilkie
>>>> Chief Technology Officer
>>>> US IP Communications 919.297.1057 *"This e-mail communication and any
>>>> attachments may contain confidential and privileged information and is for
>>>> use by the designated addressee(s) named above only. Any files transmitted
>>>> with it are confidential and intended solely for the use of the individual
>>>> to whom it is addressed. Any views or opinions presented are solely those
>>>> of the author and do not necessarily represent those of USIPCOM, LLC. If
>>>> you are not the intended addressee, you are hereby notified that you have
>>>> received this communication in error and that any use or reproduction of
>>>> this email or its contents is strictly prohibited and may be unlawful. If
>>>> you have received this communication in error, please notify us immediately
>>>> by replying to this message and deleting it from your computer. Thank you".
>>>> *
>>>> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <
>>>> bogdan at opensips.org> wrote:
>>>>>
>>>>> Hi Jeff, You mean you do not get the Attributes text area at all in
>>>>> the Edit section ? Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>>
>>>>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP
>>>>> 5.0
>>>>> It appears that when modifying the CP for Dialplan attributes as text,
>>>>> they do not appear in the edit section once applied.  The attributes do
>>>>> show up in the view list of dial plans but if you go back to edit the attrs
>>>>> again, they are not there.  I dot not have the same problem when using
>>>>> attrs as checkboxes.  Is there more code in the local files that require
>>>>> editing to display these values?  Also, if an entry in the Dialplan was
>>>>> once a checkbox value but I've not converted the mode to text, the value
>>>>> does not show up in the text box to be edited.
>>>>>
>>>>> # Attention : advanced options !!
>>>>>
>>>>>  //database tables
>>>>>
>>>>>  $config->table_dialplan = "dialplan";
>>>>>
>>>>>  $config->results_per_page = 20;
>>>>>
>>>>>  $config->results_page_range = 5;
>>>>>
>>>>>  $config->attrs_cb=array(
>>>>>
>>>>>                                         // name , description
>>>>>
>>>>>                                         array("a","Descr a"),
>>>>>
>>>>>                                         array("b","Descr b"),
>>>>>
>>>>>                                         array("c","Descr c"),
>>>>>
>>>>>                                         array("d","Descr d"),
>>>>>
>>>>>                                         array("e","Descr e"),
>>>>>
>>>>>                                         array("f","Descr f"),
>>>>>
>>>>>                                         );
>>>>>
>>>>>  $config->cb_per_row = 3;
>>>>>
>>>>>  $talk_to_this_assoc_id = 1 ;
>>>>>
>>>>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>>>>
>>>>> // 0 - checkboxes
>>>>>
>>>>> // 1 - text
>>>>>
>>>>> $dialplan_attributes_mode = 1  ;
>>>>>
>>>>> ?>
>>>>> Jeff
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>
>
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From jwilkie at usipcom.com  Thu Aug 25 04:33:15 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Wed, 24 Aug 2016 22:33:15 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
Message-ID: <CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>

If I modify the file
/var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
and comment out the following:

        //              if(get_magic_quotes_gpc()==0){


        //                      $match_exp =
mysql_real_escape_string($match_exp, *$link*);

        //              }

The updates work fine.  I think noticed that the ref to *$link* sits below
this line in the "modify" section so I decided to remove the $link ref in
the above lines and then uncommented the lines leaving the modify section
like this

                if ($errors=="") {

                        if(get_magic_quotes_gpc()==0){


                                $match_exp =
mysql_real_escape_string($match_exp);

                        }


                        $sql = "SELECT * FROM ".$table.

                        " WHERE dpid=" .$dpid. " AND match_exp='"
.$match_exp. "'".

                        " AND id!=".$id;

                        $resultset = *$link*->queryAll($sql);


Let me know if I've broken something as a result but this appears to have
corrected the problem I was having.  Unknown of any side effects currently
:)  Take a look and let me know your thoughts.

Jeff



On Wed, Aug 24, 2016 at 6:22 PM, Jeff Wilkie <jwilkie at usipcom.com> wrote:

> Here is what it looks like to my db when I create a DialPlan
>
>   20 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='877'
>
>   20 Query PREPARE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
> FROM 'INSERT INTO dialplan\n (dpid, pr, match_op, match_exp, match_flags,
> subst_exp, \n repl_exp, attrs) VALUES \n (36, 0,1,
> \'877\',0,\'\',\'\',\'.30\')'
>
>   20 Prepare INSERT INTO dialplan
>
> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>
> repl_exp, attrs) VALUES
>
> (36, 0,1, '877',0,'','','.30')
>
>   20 Query EXECUTE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>
>   20 Execute INSERT INTO dialplan
>
> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>
> repl_exp, attrs) VALUES
>
> (36, 0,1, '877',0,'','','.30')
>
> This is what happens when I edit that same DialPlan
>
>   23 Query select * from dialplan where id='9030'
>
>   23 Quit
>
>   24 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='' AND
> id!=9030
>
>   24 Query PREPARE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
> FROM 'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp =\'\',
> match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs= \'.30\' WHERE
> id=9030'
>
>   24 Prepare UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
> ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030
>
>   24 Query EXECUTE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>
>   24 Execute UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
> ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030
>
>   24 Query DEALLOCATE PREPARE MDB2_STATEMENT_mysql_
> 880efa45d78ebff7cbf5f617428df4d2
>
>   24 Quit
>
>
> You can see that the container for match_ex is not getting passed to the
> '24' SELECT statement which ends in a 0 results query apparently.
>
>
> Thanks
>
> Jeff
>
> On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie <jwilkie at usipcom.com> wrote:
>
>> Yes.  After any edits followed by a save with the attribute field active
>> as text causes "Matching Regular Expression" to be deleted.
>>
>> Thanks
>>
>> Jeff
>>
>> On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu <bogdan at opensips.org
>> > wrote:
>>
>>> Jeff,
>>>
>>> You mean "Matching Regular Expression", right ? upon performing an edit
>>> + save where you only changed the attrs, the "Matching Regular
>>> Expression" gets to empty ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 24.08.2016 23:25, Jeff Wilkie wrote:
>>>
>>> Ok.  The value now displays but if you modify the value and hit save, it
>>> kills the regex value upon saving.  Also, you cannot add the regex value
>>> back into the row.  All other values appear to be kept.
>>>
>>> Thanks
>>>
>>> Jeff
>>>
>>> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <
>>> bogdan at opensips.org> wrote:
>>>
>>>> Hi Jeff,
>>>>
>>>> I spotted the issue and fixed it on GIT repo. Please give it a try and
>>>> let me know if it works for you too.
>>>>
>>>> Thanks and regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>> On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>>
>>>> Yes, The text box is there but when you place a value in a preexisting
>>>> dial plan, the result in the DB is just the value in the text attribute
>>>> field with no dial plan information.  By editing or adding an attribute it
>>>> kills the rest of the dial plan that you were editing and you cannot add
>>>> the dial plan information back in via the CP as long as an attribute in
>>>> text form is added.  The same does not happen if using check box.  I can
>>>> provide pics if you'd like.  The only var changed is the
>>>>
>>>> $dialplan_attributes_mode = 1
>>>>
>>>> Jeff Wilkie
>>>> Chief Technology Officer
>>>> US IP Communications 919.297.1057 *"This e-mail communication and any
>>>> attachments may contain confidential and privileged information and is for
>>>> use by the designated addressee(s) named above only. Any files transmitted
>>>> with it are confidential and intended solely for the use of the individual
>>>> to whom it is addressed. Any views or opinions presented are solely those
>>>> of the author and do not necessarily represent those of USIPCOM, LLC. If
>>>> you are not the intended addressee, you are hereby notified that you have
>>>> received this communication in error and that any use or reproduction of
>>>> this email or its contents is strictly prohibited and may be unlawful. If
>>>> you have received this communication in error, please notify us immediately
>>>> by replying to this message and deleting it from your computer. Thank you".
>>>> *
>>>> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <
>>>> bogdan at opensips.org> wrote:
>>>>>
>>>>> Hi Jeff, You mean you do not get the Attributes text area at all in
>>>>> the Edit section ? Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>>
>>>>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP
>>>>> 5.0
>>>>> It appears that when modifying the CP for Dialplan attributes as text,
>>>>> they do not appear in the edit section once applied.  The attributes do
>>>>> show up in the view list of dial plans but if you go back to edit the attrs
>>>>> again, they are not there.  I dot not have the same problem when using
>>>>> attrs as checkboxes.  Is there more code in the local files that require
>>>>> editing to display these values?  Also, if an entry in the Dialplan was
>>>>> once a checkbox value but I've not converted the mode to text, the value
>>>>> does not show up in the text box to be edited.
>>>>>
>>>>> # Attention : advanced options !!
>>>>>
>>>>>  //database tables
>>>>>
>>>>>  $config->table_dialplan = "dialplan";
>>>>>
>>>>>  $config->results_per_page = 20;
>>>>>
>>>>>  $config->results_page_range = 5;
>>>>>
>>>>>  $config->attrs_cb=array(
>>>>>
>>>>>                                         // name , description
>>>>>
>>>>>                                         array("a","Descr a"),
>>>>>
>>>>>                                         array("b","Descr b"),
>>>>>
>>>>>                                         array("c","Descr c"),
>>>>>
>>>>>                                         array("d","Descr d"),
>>>>>
>>>>>                                         array("e","Descr e"),
>>>>>
>>>>>                                         array("f","Descr f"),
>>>>>
>>>>>                                         );
>>>>>
>>>>>  $config->cb_per_row = 3;
>>>>>
>>>>>  $talk_to_this_assoc_id = 1 ;
>>>>>
>>>>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>>>>
>>>>> // 0 - checkboxes
>>>>>
>>>>> // 1 - text
>>>>>
>>>>> $dialplan_attributes_mode = 1  ;
>>>>>
>>>>> ?>
>>>>> Jeff
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>
>
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From spanda at 3clogic.com  Thu Aug 25 12:35:43 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Thu, 25 Aug 2016 16:05:43 +0530
Subject: [OpenSIPS-Users] Username in location table of opensips-1.11 .
Message-ID: <CALf-Q44ZU-r=-DZX3SCbH8US6W2++mCyQspsvNbVRQas4+q7Fw@mail.gmail.com>

Hi All ,

     I am using opensips-1.11 . I am facing some problem while doing lookup
.

There is two agent . SIP and SIP1
Both are register in with the above name .
Now SIP1 is calling SIP , and the request uri in INVITE is like bellow .

*INVITE sip : sip at sip-aws.i3clogic.com:5507
<http://sip at sip-aws.i3clogic.com:5507>*

For this request SER is sending 404 not found . This is working fine with
 opensips-1.6  . But in opensips-1.11 its not working .

  Is there any changes in 1.11 . As for RFC username should not be case
sensitive . So why is the Server sending 404 for the request .


Please help me I am block with this .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From bogdan at opensips.org  Thu Aug 25 16:35:09 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 25 Aug 2016 17:35:09 +0300
Subject: [OpenSIPS-Users] Username in location table of opensips-1.11 .
In-Reply-To: <CALf-Q44ZU-r=-DZX3SCbH8US6W2++mCyQspsvNbVRQas4+q7Fw@mail.gmail.com>
References: <CALf-Q44ZU-r=-DZX3SCbH8US6W2++mCyQspsvNbVRQas4+q7Fw@mail.gmail.com>
Message-ID: <359745b2-8646-320d-f648-8f7af9b8d9eb@opensips.org>

Hi Panda,

Actually RFC3261 says that the AOR matching is case sensitive. So, 1.11 
is acting like that.

Please see:
http://www.opensips.org/html/docs/modules/1.11.x/registrar.html#id293632

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 25.08.2016 13:35, Sasmita Panda wrote:
> Hi All ,
>
>      I am using opensips-1.11 . I am facing some problem while doing 
> lookup .
>
> There is two agent . SIP and SIP1
> Both are register in with the above name .
> Now SIP1 is calling SIP , and the request uri in INVITE is like bellow .
>
> *INVITE sip : sip at sip-aws.i3clogic.com:5507 
> <http://sip at sip-aws.i3clogic.com:5507>*
> *
> *
> For this request SER is sending 404 not found . This is working fine with
>  opensips-1.6  . But in opensips-1.11 its not working .
>
>   Is there any changes in 1.11 . As for RFC username should not be 
> case sensitive . So why is the Server sending 404 for the request .
>
>
> Please help me I am block with this .
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Thu Aug 25 16:53:41 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 25 Aug 2016 17:53:41 +0300
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
Message-ID: <f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>

Hi Agalya,

Question : after you get these logs, you mentioned the inability of 
handling new calls. Does OpenSIPS recover from this ? or you have to 
restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
>
> Hi team,
>
> Am using opensips 2.1.4 version and trying to do load test. When 
> running the test cases, am getting too many warnings in OpenSIPS log.
>
> These warnings are coming continuously and its more than 2 page in log 
> file.
>
> Below are the warnings.
>
> Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: 
> WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 
> us delay in execution
>
> Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: 
> WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 
> us delay in execution
>
> Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: 
> WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 
> us delay in execution
>
> Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: 
> WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 
> us delay in execution
>
> Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: 
> WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 
> us delay in execution
>
> Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: 
> WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 
> us delay in execution
>
> When I get this Warning, calls are getting failed.
>
> Why am getting this warning and what could be done to avoid it?
>
> Regards,
> Agalya
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Agalya_Ramachandran at comcast.com  Thu Aug 25 17:08:45 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 25 Aug 2016 15:08:45 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
Message-ID: <778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>

Hi Bogdan,

Does OpenSIPS recover from this ?
Answer: It depends upon the volume of the call. If am sending too many calls (may be more than 50 calls), OpenSIPS is not recovering. And am restarting
But if I send less than 10 calls then it is OK.

Have you seen other job reported, besides "tm-utimer" ?
Answer: I also have "blcore-expire" in logs, like once every 10 times of "tm-utimer"

I also observed that increasing the children process in config file, is solving this issue.

P.S:

e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to keep children as 12.
     For even 10 calls, if I keep children as 4, 4 to 6 calls are passing and other calls are failing.
Why do we need to increase the children process?

Just to confirm, is opensips 2.1.4 version  a stable release?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 10:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Question : after you get these logs, you mentioned the inability of handling new calls. Does OpenSIPS recover from this ? or you have to restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya




_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From saioa.perurena at enigmedia.es  Thu Aug 25 18:27:47 2016
From: saioa.perurena at enigmedia.es (Saioa Perurena)
Date: Thu, 25 Aug 2016 18:27:47 +0200
Subject: [OpenSIPS-Users] HEP protocol on opensips 2.1 siptrace module
Message-ID: <75036af3-cd46-8e61-5969-6c7954805fd6@enigmedia.es>

Hi,

I'm trying to use sip trace module on opensips 2.1 to send all sip
messages to a homer server, but it does not send the messages with hep
protocol to the sipcapture server. Any idea of what I'm missing?

I have the following configuration on opensips script:

#### Siptrace module
loadmodule "siptrace.so"
modparam("siptrace", "duplicate_uri", "sip:x.x.x.x:9060")
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_flag", "TRACE_FLAG")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "db_url","")
modparam("siptrace", "enable_ack_trace", 1)
modparam("siptrace", "traced_user_avp", "$avp(trace_id)")
modparam("siptrace", "duplicate_with_hep", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "hep_capture_id", 1)

And at the beginning of the route:
	setflag(TRACE_FLAG);
	$avp(trace_id) = $fu;
	sip_trace();

Thanks in advance!

Saioa.


From Agalya_Ramachandran at comcast.com  Thu Aug 25 20:58:32 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 25 Aug 2016 18:58:32 +0000
Subject: [OpenSIPS-Users] HEP protocol on opensips 2.1 siptrace module
In-Reply-To: <75036af3-cd46-8e61-5969-6c7954805fd6@enigmedia.es>
References: <75036af3-cd46-8e61-5969-6c7954805fd6@enigmedia.es>
Message-ID: <18531c19a34e461caf8588ebfa03e9f4@COPDCEX28.cable.comcast.com>

Hi Saioa,

I have been using sip trace module to forward packets.

I don't have the below added in config.
modparam("siptrace", "trace_flag", "TRACE_FLAG")
modparam("siptrace", "enable_ack_trace", 1)
modparam("siptrace", "hep_capture_id", 1)

Remove, setflag(TRACE_FLAG); as we are removing trace_flag modparam.
And I have $avp(trace_id) = 1;

This works for me. Try it .

Regards,
Agalya

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Saioa Perurena
Sent: Thursday, August 25, 2016 12:28 PM
To: users at lists.opensips.org
Subject: [OpenSIPS-Users] HEP protocol on opensips 2.1 siptrace module

Hi,

I'm trying to use sip trace module on opensips 2.1 to send all sip messages to a homer server, but it does not send the messages with hep protocol to the sipcapture server. Any idea of what I'm missing?

I have the following configuration on opensips script:

#### Siptrace module
loadmodule "siptrace.so"
modparam("siptrace", "duplicate_uri", "sip:x.x.x.x:9060") modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", "TRACE_FLAG") modparam("siptrace", "trace_on", 1) modparam("siptrace", "db_url","") modparam("siptrace", "enable_ack_trace", 1) modparam("siptrace", "traced_user_avp", "$avp(trace_id)") modparam("siptrace", "duplicate_with_hep", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_capture_id", 1)

And at the beginning of the route:
        setflag(TRACE_FLAG);
        $avp(trace_id) = $fu;
        sip_trace();

Thanks in advance!

Saioa.

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From bogdan at opensips.org  Thu Aug 25 23:11:49 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 26 Aug 2016 00:11:49 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
Message-ID: <d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>

Hi Jeff,

I run some test with the exact code on GITHUB (branch 5.0) and the edit 
actually worked well (without any further changes).

The $link is a variable that is initialized in the included file 
lib/db_connect.php, so it valid (anyhow, it is used later for the actual 
query, which works ;) ).

Could you check if the return of mysql_real_escape_string() is actually 
something valid or is it FALSE ?
http://stackoverflow.com/questions/3005135/mysql-real-escape-string-just-makes-an-empty-string

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 25.08.2016 05:33, Jeff Wilkie wrote:
> If I modify the file 
> /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php and 
> comment out the following:
>
>         // if(get_magic_quotes_gpc()==0){
>
>
>       //                      $match_exp = 
> mysql_real_escape_string($match_exp, *$link*);
>
>         //              }
>
>
> The updates work fine.  I think noticed that the ref to *$link* sits 
> below this line in the "modify" section so I decided to remove the 
> $link ref in the above lines and then uncommented the lines leaving 
> the modify section like this
>
>                 if ($errors=="") {
>
> if(get_magic_quotes_gpc()==0){
>
>
>                                 $match_exp = 
> mysql_real_escape_string($match_exp);
>
>                         }
>
>
>                         $sql = "SELECT * FROM ".$table.
>
>                         " WHERE dpid=" .$dpid. " AND match_exp='" 
> .$match_exp. "'".
>
>                         " AND id!=".$id;
>
>                       $resultset = *$link*->queryAll($sql);
>
>
>
> Let me know if I've broken something as a result but this appears to 
> have corrected the problem I was having.  Unknown of any side effects 
> currently :)  Take a look and let me know your thoughts.
>
> Jeff
>
>
>
> On Wed, Aug 24, 2016 at 6:22 PM, Jeff Wilkie <jwilkie at usipcom.com 
> <mailto:jwilkie at usipcom.com>> wrote:
>
>     Here is what it looks like to my db when I create a DialPlan
>
>       20 QuerySELECT * FROM dialplan WHERE dpid=36 AND match_exp='877'
>
>       20 QueryPREPARE
>     MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74 FROM 'INSERT
>     INTO dialplan\n(dpid, pr, match_op, match_exp, match_flags,
>     subst_exp, \nrepl_exp, attrs) VALUES \n(36, 0,1,
>     \'877\',0,\'\',\'\',\'.30\')'
>
>       20 PrepareINSERT INTO dialplan
>
>     (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>
>     repl_exp, attrs) VALUES
>
>     (36, 0,1, '877',0,'','','.30')
>
>       20 QueryEXECUTE
>     MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>
>       20 ExecuteINSERT INTO dialplan
>
>     (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>
>     repl_exp, attrs) VALUES
>
>     (36, 0,1, '877',0,'','','.30')
>
>
>     This is what happens when I edit that same DialPlan
>
>       23 Queryselect * from dialplan where id='9030'
>
>       23 Quit
>
>       24 QuerySELECT * FROM dialplan WHERE dpid=36 AND match_exp=''
>     AND id!=9030
>
>       24 QueryPREPARE
>     MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2 FROM 'UPDATE
>     dialplan SET dpid=36, pr = 0, match_op= 1, match_exp =\'\',
>     match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs= \'.30\'
>     WHERE id=9030'
>
>       24 PrepareUPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>     match_exp ='', match_flags=0, subst_exp = '', repl_exp='', attrs=
>     '.30' WHERE id=9030
>
>       24 QueryEXECUTE
>     MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>
>       24 ExecuteUPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>     match_exp ='', match_flags=0, subst_exp = '', repl_exp='', attrs=
>     '.30' WHERE id=9030
>
>       24 QueryDEALLOCATE PREPARE
>     MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>
>       24 Quit
>
>
>
>     You can see that the container for match_ex is not getting passed
>     to the '24' SELECT statement which ends in a 0 results query
>     apparently.
>
>
>     Thanks
>
>     Jeff
>
>     On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie <jwilkie at usipcom.com
>     <mailto:jwilkie at usipcom.com>> wrote:
>
>         Yes.  After any edits followed by a save with the attribute
>         field active as text causes "Matching Regular Expression" to
>         be deleted.
>
>         Thanks
>
>         Jeff
>
>         On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu
>         <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>             Jeff,
>
>             You mean "Matching Regular Expression", right ? upon
>             performing an edit + save where you only changed the
>             attrs, the "Matching Regular Expression" gets to empty ?
>
>             Regards,
>
>             Bogdan-Andrei Iancu
>             OpenSIPS Founder and Developer
>             http://www.opensips-solutions.com
>             <http://www.opensips-solutions.com>
>
>             On 24.08.2016 23:25, Jeff Wilkie wrote:
>>             Ok.  The value now displays but if you modify the value
>>             and hit save, it kills the regex value upon saving. 
>>             Also, you cannot add the regex value back into the row. 
>>             All other values appear to be kept.
>>             Thanks
>>             Jeff
>>             On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu
>>             <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>                 Hi Jeff, I spotted the issue and fixed it on GIT
>>                 repo. Please give it a try and let me know if it
>>                 works for you too. Thanks and regards,
>>
>>                 Bogdan-Andrei Iancu
>>                 OpenSIPS Founder and Developer
>>                 http://www.opensips-solutions.com
>>                 <http://www.opensips-solutions.com>
>>
>>                 On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>                 Yes, The text box is there but when you place a
>>>                 value in a preexisting dial plan, the result in the
>>>                 DB is just the value in the text attribute field
>>>                 with no dial plan information.  By editing or adding
>>>                 an attribute it kills the rest of the dial plan that
>>>                 you were editing and you cannot add the dial plan
>>>                 information back in via the CP as long as an
>>>                 attribute in text form is added.  The same does not
>>>                 happen if using check box.  I can provide pics if
>>>                 you'd like.  The only var changed is the
>>>>
>>>>                 $dialplan_attributes_mode = 1
>>>>
>>>                 Jeff Wilkie
>>>                 Chief Technology Officer
>>>                 US IP Communications 919.297.1057 <tel:919.297.1057>
>>>                 /"This e-mail communication and any attachments may
>>>                 contain confidential and privileged information and
>>>                 is for use by the designated addressee(s) named
>>>                 above only. Any files transmitted with it are
>>>                 confidential and intended solely for the use of the
>>>                 individual to whom it is addressed. Any views or
>>>                 opinions presented are solely those of the author
>>>                 and do not necessarily represent those of USIPCOM,
>>>                 LLC. If you are not the intended addressee, you are
>>>                 hereby notified that you have received this
>>>                 communication in error and that any use or
>>>                 reproduction of this email or its contents is
>>>                 strictly prohibited and may be unlawful. If you have
>>>                 received this communication in error, please notify
>>>                 us immediately by replying to this message and
>>>                 deleting it from your computer. Thank you". /
>>>                 On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu
>>>                 <bogdan at opensips.org <mailto:bogdan at opensips.org>>
>>>                 wrote:
>>>
>>>                     Hi Jeff, You mean you do not get the Attributes
>>>                     text area at all in the Edit section ? Regards,
>>>
>>>                     Bogdan-Andrei Iancu
>>>                     OpenSIPS Founder and Developer
>>>                     http://www.opensips-solutions.com
>>>                     <http://www.opensips-solutions.com>
>>>
>>>                     On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>                     New small issue with DIALPLAN in the CP. 
>>>>                     Running OSIPS 1.10 with CP 5.0
>>>>                     It appears that when modifying the CP for
>>>>                     Dialplan attributes as text, they do not appear
>>>>                     in the edit section once applied.  The
>>>>                     attributes do show up in the view list of dial
>>>>                     plans but if you go back to edit the attrs
>>>>                     again, they are not there.  I dot not have the
>>>>                     same problem when using attrs as checkboxes. 
>>>>                     Is there more code in the local files that
>>>>                     require editing to display these values?  Also,
>>>>                     if an entry in the Dialplan was once a checkbox
>>>>                     value but I've not converted the mode to text,
>>>>                     the value does not show up in the text box to
>>>>                     be edited.
>>>>
>>>>                     # Attention : advanced options !!
>>>>
>>>>                      //database tables
>>>>
>>>>                      $config->table_dialplan = "dialplan";
>>>>
>>>>                      $config->results_per_page = 20;
>>>>
>>>>                      $config->results_page_range = 5;
>>>>
>>>>                      $config->attrs_cb=array(
>>>>
>>>>                                                           // name ,
>>>>                     description
>>>>
>>>>                                                          
>>>>                     array("a","Descr a"),
>>>>
>>>>                                                          
>>>>                     array("b","Descr b"),
>>>>
>>>>                                                          
>>>>                     array("c","Descr c"),
>>>>
>>>>                                                          
>>>>                     array("d","Descr d"),
>>>>
>>>>                                                          
>>>>                     array("e","Descr e"),
>>>>
>>>>                                                          
>>>>                     array("f","Descr f"),
>>>>
>>>>                                                           );
>>>>
>>>>                      $config->cb_per_row = 3;
>>>>
>>>>                      $talk_to_this_assoc_id = 1 ;
>>>>
>>>>                     // Dialplan - Add/Edit new Translation Rule -
>>>>                     Attributes input type
>>>>
>>>>                     // 0 - checkboxes
>>>>
>>>>                     // 1 - text
>>>>
>>>>                     $dialplan_attributes_mode = 1  ;
>>>>
>>>>                     ?>
>>>>
>>>>                     Jeff
>>>>
>>>>                     _______________________________________________
>>>>                     Users mailing list
>>>>                     Users at lists.opensips.org
>>>>                     <mailto:Users at lists.opensips.org>
>>>>                     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>                     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>
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From bogdan at opensips.org  Thu Aug 25 23:19:30 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 26 Aug 2016 00:19:30 +0300
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
Message-ID: <b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>

Hi Agalya,

Before those warnings, have you got an warning like:
         WARNING:core:utimer_ticker: utimer task <tm-utimer> already 
scheduled for....

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> *Does OpenSIPS recover from this ?*
>
> Answer: It depends upon the volume of the call. If am sending too many 
> calls (may be more than 50 calls), OpenSIPS is not recovering. And am 
> restarting
>
> But if I send less than 10 calls then it is OK.
>
> *Have you seen other job reported, besides "tm-utimer" ?*
>
> Answer: I also have ?blcore-expire? in logs, like once every 10 times 
> of "tm-utimer"
>
> I also observed that increasing the children process in config file, 
> is solving this issue.
>
> P.S:
>
> e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to 
> keep children as 12.
>
>      For even 10 calls, if I keep children as 4, 4 to 6 calls are 
> passing and other calls are failing.
>
> Why do we need to increase the children process?
>
> Just to confirm, is opensips 2.1.4 version  a stable release?
>
> Regards,
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Thursday, August 25, 2016 10:54 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; 
> Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
> *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
> Hi Agalya,
>
> Question : after you get these logs, you mentioned the inability of 
> handling new calls. Does OpenSIPS recover from this ? or you have to 
> restart after such an event ?
>
> Have you seen other job reported, besides "tm-utimer" ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi team,
>
>     Am using opensips 2.1.4 version and trying to do load test. When
>     running the test cases, am getting too many warnings in OpenSIPS log.
>
>     These warnings are coming continuously and its more than 2 page in
>     log file.
>
>     Below are the warnings.
>
>     Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>     39450000 us delay in execution
>
>     Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>     39450000 us delay in execution
>
>     Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>     39450000 us delay in execution
>
>     Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>     39450000 us delay in execution
>
>     Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>     39450000 us delay in execution
>
>     Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>     39450000 us delay in execution
>
>     When I get this Warning, calls are getting failed.
>
>     Why am getting this warning and what could be done to avoid it?
>
>     Regards,
>     Agalya
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From jwilkie at usipcom.com  Fri Aug 26 05:05:22 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Thu, 25 Aug 2016 23:05:22 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
Message-ID: <CANFkrT25RE+6fP5O-6wJt_muV1r6kjaj_Rpfna8P=LCsxx8xzg@mail.gmail.com>

When using

$match_exp = mysql_real_escape_string($match_exp, *$link*);

I get NULL with var_dump

When using

$match_exp = mysql_real_escape_string($match_exp);

I get with a var_dump

string(10) "^\\+1907.*"

Thanks
Jeff
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From jwilkie at usipcom.com  Fri Aug 26 07:33:19 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Fri, 26 Aug 2016 01:33:19 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
Message-ID: <CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>

I just loaded all the web directory content from git to test but the
results are the same.  The config dir for dialplan are the same other than
the toggle for 1 on text attributes.  Are there other files/dirs that would
affect the dial plan section of CP?  Also, the git download does not
include the changes from expression string length to expression flags (5.0
cp)

Jeff


On Thu, Aug 25, 2016 at 5:11 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jeff,
>
> I run some test with the exact code on GITHUB (branch 5.0) and the edit
> actually worked well (without any further changes).
>
> The $link is a variable that is initialized in the included file
> lib/db_connect.php, so it valid (anyhow, it is used later for the actual
> query, which works ;) ).
>
> Could you check if the return of mysql_real_escape_string() is actually
> something valid or is it FALSE ?
>     http://stackoverflow.com/questions/3005135/mysql-real-
> escape-string-just-makes-an-empty-string
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 25.08.2016 05:33, Jeff Wilkie wrote:
>
> If I modify the file /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
> and comment out the following:
>
>         //              if(get_magic_quotes_gpc()==0){
>
>
>         //                      $match_exp = mysql_real_escape_string($match_exp,
> *$link*);
>
>         //              }
>
> The updates work fine.  I think noticed that the ref to *$link* sits
> below this line in the "modify" section so I decided to remove the $link
> ref in the above lines and then uncommented the lines leaving the modify
> section like this
>
>                 if ($errors=="") {
>
>                         if(get_magic_quotes_gpc()==0){
>
>
>                                 $match_exp = mysql_real_escape_string($
> match_exp);
>
>                         }
>
>
>                         $sql = "SELECT * FROM ".$table.
>
>                         " WHERE dpid=" .$dpid. " AND match_exp='"
> .$match_exp. "'".
>
>                         " AND id!=".$id;
>
>                         $resultset = *$link*->queryAll($sql);
>
>
> Let me know if I've broken something as a result but this appears to have
> corrected the problem I was having.  Unknown of any side effects currently
> :)  Take a look and let me know your thoughts.
>
> Jeff
>
>
>
> On Wed, Aug 24, 2016 at 6:22 PM, Jeff Wilkie <jwilkie at usipcom.com> wrote:
>
>> Here is what it looks like to my db when I create a DialPlan
>>
>>   20 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='877'
>>
>>   20 Query PREPARE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>> FROM 'INSERT INTO dialplan\n (dpid, pr, match_op, match_exp,
>> match_flags, subst_exp, \n repl_exp, attrs) VALUES \n (36, 0,1,
>> \'877\',0,\'\',\'\',\'.30\')'
>>
>>   20 Prepare INSERT INTO dialplan
>>
>> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>
>> repl_exp, attrs) VALUES
>>
>> (36, 0,1, '877',0,'','','.30')
>>
>>   20 Query EXECUTE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>>
>>   20 Execute INSERT INTO dialplan
>>
>> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>
>> repl_exp, attrs) VALUES
>>
>> (36, 0,1, '877',0,'','','.30')
>>
>> This is what happens when I edit that same DialPlan
>>
>>   23 Query select * from dialplan where id='9030'
>>
>>   23 Quit
>>
>>   24 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='' AND
>> id!=9030
>>
>>   24 Query PREPARE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>> FROM 'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp =\'\',
>> match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs= \'.30\' WHERE
>> id=9030'
>>
>>   24 Prepare UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
>> ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030
>>
>>   24 Query EXECUTE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>
>>   24 Execute UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
>> ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30' WHERE id=9030
>>
>>   24 Query DEALLOCATE PREPARE MDB2_STATEMENT_mysql_880efa45d
>> 78ebff7cbf5f617428df4d2
>>
>>   24 Quit
>>
>>
>> You can see that the container for match_ex is not getting passed to the
>> '24' SELECT statement which ends in a 0 results query apparently.
>>
>>
>> Thanks
>>
>> Jeff
>>
>> On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie <jwilkie at usipcom.com> wrote:
>>
>>> Yes.  After any edits followed by a save with the attribute field active
>>> as text causes "Matching Regular Expression" to be deleted.
>>>
>>> Thanks
>>>
>>> Jeff
>>>
>>> On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu <
>>> bogdan at opensips.org> wrote:
>>>
>>>> Jeff,
>>>>
>>>> You mean "Matching Regular Expression", right ? upon performing an edit
>>>> + save where you only changed the attrs, the "Matching Regular
>>>> Expression" gets to empty ?
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>> On 24.08.2016 23:25, Jeff Wilkie wrote:
>>>>
>>>> Ok.  The value now displays but if you modify the value and hit save,
>>>> it kills the regex value upon saving.  Also, you cannot add the regex value
>>>> back into the row.  All other values appear to be kept.
>>>> Thanks
>>>> Jeff
>>>> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <
>>>> bogdan at opensips.org> wrote:
>>>>>
>>>>> Hi Jeff, I spotted the issue and fixed it on GIT repo. Please give it
>>>>> a try and let me know if it works for you too. Thanks and regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>> On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>>>
>>>>> Yes, The text box is there but when you place a value in a preexisting
>>>>> dial plan, the result in the DB is just the value in the text attribute
>>>>> field with no dial plan information.  By editing or adding an attribute it
>>>>> kills the rest of the dial plan that you were editing and you cannot add
>>>>> the dial plan information back in via the CP as long as an attribute in
>>>>> text form is added.  The same does not happen if using check box.  I can
>>>>> provide pics if you'd like.  The only var changed is the
>>>>>
>>>>> $dialplan_attributes_mode = 1
>>>>>
>>>>> Jeff Wilkie
>>>>> Chief Technology Officer
>>>>> US IP Communications 919.297.1057 *"This e-mail communication and any
>>>>> attachments may contain confidential and privileged information and is for
>>>>> use by the designated addressee(s) named above only. Any files transmitted
>>>>> with it are confidential and intended solely for the use of the individual
>>>>> to whom it is addressed. Any views or opinions presented are solely those
>>>>> of the author and do not necessarily represent those of USIPCOM, LLC. If
>>>>> you are not the intended addressee, you are hereby notified that you have
>>>>> received this communication in error and that any use or reproduction of
>>>>> this email or its contents is strictly prohibited and may be unlawful. If
>>>>> you have received this communication in error, please notify us immediately
>>>>> by replying to this message and deleting it from your computer. Thank you".
>>>>> *
>>>>> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <
>>>>> bogdan at opensips.org> wrote:
>>>>>>
>>>>>> Hi Jeff, You mean you do not get the Attributes text area at all in
>>>>>> the Edit section ? Regards,
>>>>>>
>>>>>> Bogdan-Andrei Iancu
>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>
>>>>>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>>>
>>>>>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP
>>>>>> 5.0
>>>>>> It appears that when modifying the CP for Dialplan attributes as
>>>>>> text, they do not appear in the edit section once applied.  The attributes
>>>>>> do show up in the view list of dial plans but if you go back to edit the
>>>>>> attrs again, they are not there.  I dot not have the same problem when
>>>>>> using attrs as checkboxes.  Is there more code in the local files that
>>>>>> require editing to display these values?  Also, if an entry in the Dialplan
>>>>>> was once a checkbox value but I've not converted the mode to text, the
>>>>>> value does not show up in the text box to be edited.
>>>>>>
>>>>>> # Attention : advanced options !!
>>>>>>
>>>>>>  //database tables
>>>>>>
>>>>>>  $config->table_dialplan = "dialplan";
>>>>>>
>>>>>>  $config->results_per_page = 20;
>>>>>>
>>>>>>  $config->results_page_range = 5;
>>>>>>
>>>>>>  $config->attrs_cb=array(
>>>>>>
>>>>>>                                         // name , description
>>>>>>
>>>>>>                                         array("a","Descr a"),
>>>>>>
>>>>>>                                         array("b","Descr b"),
>>>>>>
>>>>>>                                         array("c","Descr c"),
>>>>>>
>>>>>>                                         array("d","Descr d"),
>>>>>>
>>>>>>                                         array("e","Descr e"),
>>>>>>
>>>>>>                                         array("f","Descr f"),
>>>>>>
>>>>>>                                         );
>>>>>>
>>>>>>  $config->cb_per_row = 3;
>>>>>>
>>>>>>  $talk_to_this_assoc_id = 1 ;
>>>>>>
>>>>>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>>>>>
>>>>>> // 0 - checkboxes
>>>>>>
>>>>>> // 1 - text
>>>>>>
>>>>>> $dialplan_attributes_mode = 1  ;
>>>>>>
>>>>>> ?>
>>>>>> Jeff
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
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From saioa.perurena at enigmedia.es  Fri Aug 26 11:16:03 2016
From: saioa.perurena at enigmedia.es (Saioa Perurena)
Date: Fri, 26 Aug 2016 11:16:03 +0200
Subject: [OpenSIPS-Users] HEP protocol on opensips 2.1 siptrace module
In-Reply-To: <18531c19a34e461caf8588ebfa03e9f4@COPDCEX28.cable.comcast.com>
References: <75036af3-cd46-8e61-5969-6c7954805fd6@enigmedia.es>
	<18531c19a34e461caf8588ebfa03e9f4@COPDCEX28.cable.comcast.com>
Message-ID: <4bc7840b-a169-e151-5915-611e6ae1c247@enigmedia.es>

Hi,

Thanks for your reply, but neither it has worked for me.

My problem is that messages are send to homer with sip protocol, no with
hep, so i can't visualize the publish rtcp-xr statistics well because of
the protocol type.
I've tryed putting hep instead of sip on duplicate_uri, but opensips 2.1
don't like it:

modparam("siptrace", "duplicate_uri", "hep:x.x.x.x:9060")

Regards,
Saioa.


On 25/08/16 20:58, Ramachandran, Agalya (Contractor) wrote:
> Hi Saioa,
> 
> I have been using sip trace module to forward packets.
> 
> I don't have the below added in config.
> modparam("siptrace", "trace_flag", "TRACE_FLAG")
> modparam("siptrace", "enable_ack_trace", 1)
> modparam("siptrace", "hep_capture_id", 1)
> 
> Remove, setflag(TRACE_FLAG); as we are removing trace_flag modparam.
> And I have $avp(trace_id) = 1;
> 
> This works for me. Try it .
> 
> Regards,
> Agalya
> 
> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Saioa Perurena
> Sent: Thursday, August 25, 2016 12:28 PM
> To: users at lists.opensips.org
> Subject: [OpenSIPS-Users] HEP protocol on opensips 2.1 siptrace module
> 
> Hi,
> 
> I'm trying to use sip trace module on opensips 2.1 to send all sip messages to a homer server, but it does not send the messages with hep protocol to the sipcapture server. Any idea of what I'm missing?
> 
> I have the following configuration on opensips script:
> 
> #### Siptrace module
> loadmodule "siptrace.so"
> modparam("siptrace", "duplicate_uri", "sip:x.x.x.x:9060") modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", "TRACE_FLAG") modparam("siptrace", "trace_on", 1) modparam("siptrace", "db_url","") modparam("siptrace", "enable_ack_trace", 1) modparam("siptrace", "traced_user_avp", "$avp(trace_id)") modparam("siptrace", "duplicate_with_hep", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_capture_id", 1)
> 
> And at the beginning of the route:
>         setflag(TRACE_FLAG);
>         $avp(trace_id) = $fu;
>         sip_trace();
> 
> Thanks in advance!
> 
> Saioa.
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org<mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


From bogdan at opensips.org  Fri Aug 26 11:16:20 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 26 Aug 2016 12:16:20 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT25RE+6fP5O-6wJt_muV1r6kjaj_Rpfna8P=LCsxx8xzg@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT25RE+6fP5O-6wJt_muV1r6kjaj_Rpfna8P=LCsxx8xzg@mail.gmail.com>
Message-ID: <0aec6176-e0b4-8e53-3528-85046b09dee2@opensips.org>

Hi Jeff,

That is really weird as the $link is valid (you can print it to 
doublecheck) as it is later used in the code for running the sql code.

I'm not a php expert :P, but just googling, I found this:
http://stackoverflow.com/questions/19888928/why-mysql-real-escape-string-returns-boolean

Could you add the error_reporting(E_ALL) as suggested in the first 
response, maybe we get a clue on what is going wrong.

Thanks,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.08.2016 06:05, Jeff Wilkie wrote:
> When using
>
> $match_exp = mysql_real_escape_string($match_exp, *$link*);
>
> I get NULL with var_dump
>
> When using
>
> $match_exp = mysql_real_escape_string($match_exp);
>
> I get with a var_dump
>
> string(10) "^\\+1907.*"
>
> Thanks
> Jeff

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From bogdan at opensips.org  Fri Aug 26 11:24:12 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 26 Aug 2016 12:24:12 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
Message-ID: <f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>

Hi Jeff,

Maybe the version of MDB2 or MDB2::mysql php pear ?

The CP versions are correlated with the OpenSIPS versions. The len field 
was moved to a flag field (in dialplan table) in opensips 1.11, 
corresponding to CP 6.0 (6.1 matches opensips 2.1 and CP trunk matches 
the 2.2)

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.08.2016 08:33, Jeff Wilkie wrote:
> I just loaded all the web directory content from git to test but the 
> results are the same.  The config dir for dialplan are the same other 
> than the toggle for 1 on text attributes.  Are there other files/dirs 
> that would affect the dial plan section of CP?  Also, the git download 
> does not include the changes from expression string length to 
> expression flags (5.0 cp)
>
> Jeff
>
>
> On Thu, Aug 25, 2016 at 5:11 PM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Jeff,
>
>     I run some test with the exact code on GITHUB (branch 5.0) and the
>     edit actually worked well (without any further changes).
>
>     The $link is a variable that is initialized in the included file
>     lib/db_connect.php, so it valid (anyhow, it is used later for the
>     actual query, which works ;) ).
>
>     Could you check if the return of mysql_real_escape_string() is
>     actually something valid or is it FALSE ?
>     http://stackoverflow.com/questions/3005135/mysql-real-escape-string-just-makes-an-empty-string
>     <http://stackoverflow.com/questions/3005135/mysql-real-escape-string-just-makes-an-empty-string>
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://www.opensips-solutions.com>
>
>     On 25.08.2016 05:33, Jeff Wilkie wrote:
>>     If I modify the file
>>     /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php and
>>     comment out the following:
>>
>>             // if(get_magic_quotes_gpc()==0){
>>
>>           //                      $match_exp =
>>     mysql_real_escape_string($match_exp, *$link*);
>>
>>             //              }
>>
>>     The updates work fine.  I think noticed that the ref to *$link*
>>     sits below this line in the "modify" section so I decided to
>>     remove the $link ref in the above lines and then uncommented the
>>     lines leaving the modify section like this
>>
>>                     if ($errors=="") {
>>
>>     if(get_magic_quotes_gpc()==0){
>>
>>                                     $match_exp =
>>     mysql_real_escape_string($match_exp);
>>
>>                             }
>>
>>                             $sql = "SELECT * FROM ".$table.
>>
>>                             " WHERE dpid=" .$dpid. " AND match_exp='"
>>     .$match_exp. "'".
>>
>>                             " AND id!=".$id;
>>
>>                           $resultset = *$link*->queryAll($sql);
>>
>>     Let me know if I've broken something as a result but this appears
>>     to have corrected the problem I was having.  Unknown of any side
>>     effects currently :)  Take a look and let me know your thoughts.
>>     Jeff
>>     On Wed, Aug 24, 2016 at 6:22 PM, Jeff Wilkie <jwilkie at usipcom.com
>>     <mailto:jwilkie at usipcom.com>> wrote:
>>
>>         Here is what it looks like to my db when I create a DialPlan
>>
>>           20 QuerySELECT * FROM dialplan WHERE dpid=36 AND
>>         match_exp='877'
>>
>>           20 QueryPREPARE
>>         MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74 FROM
>>         'INSERT INTO dialplan\n(dpid, pr, match_op, match_exp,
>>         match_flags, subst_exp, \nrepl_exp, attrs) VALUES \n(36, 0,1,
>>         \'877\',0,\'\',\'\',\'.30\')'
>>
>>           20 PrepareINSERT INTO dialplan
>>
>>         (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>
>>         repl_exp, attrs) VALUES
>>
>>         (36, 0,1, '877',0,'','','.30')
>>
>>           20 QueryEXECUTE
>>         MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>>
>>           20 ExecuteINSERT INTO dialplan
>>
>>         (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>
>>         repl_exp, attrs) VALUES
>>
>>         (36, 0,1, '877',0,'','','.30')
>>
>>         This is what happens when I edit that same DialPlan
>>
>>           23 Queryselect * from dialplan where id='9030'
>>
>>           23 Quit
>>
>>           24 QuerySELECT * FROM dialplan WHERE dpid=36 AND
>>         match_exp='' AND id!=9030
>>
>>           24 QueryPREPARE
>>         MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2 FROM
>>         'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp
>>         =\'\', match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs=
>>         \'.30\' WHERE id=9030'
>>
>>           24 PrepareUPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>>         match_exp ='', match_flags=0, subst_exp = '', repl_exp='',
>>         attrs= '.30' WHERE id=9030
>>
>>           24 QueryEXECUTE
>>         MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>
>>           24 ExecuteUPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>>         match_exp ='', match_flags=0, subst_exp = '', repl_exp='',
>>         attrs= '.30' WHERE id=9030
>>
>>           24 QueryDEALLOCATE PREPARE
>>         MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>
>>           24 Quit
>>
>>         You can see that the container for match_ex is not getting
>>         passed to the '24' SELECT statement which ends in a 0 results
>>         query apparently.
>>         Thanks
>>         Jeff
>>         On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie
>>         <jwilkie at usipcom.com <mailto:jwilkie at usipcom.com>> wrote:
>>
>>             Yes.  After any edits followed by a save with the
>>             attribute field active as text causes "Matching Regular
>>             Expression" to be deleted.
>>             Thanks
>>             Jeff
>>             On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu
>>             <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>                 Jeff, You mean "Matching Regular Expression", right ?
>>                 upon performing an edit + save where you only changed
>>                 the attrs, the "Matching Regular Expression" gets to
>>                 empty ? Regards,
>>
>>                 Bogdan-Andrei Iancu
>>                 OpenSIPS Founder and Developer
>>                 http://www.opensips-solutions.com
>>                 <http://www.opensips-solutions.com>
>>
>>                 On 24.08.2016 23:25, Jeff Wilkie wrote:
>>>                 Ok.  The value now displays but if you modify the
>>>                 value and hit save, it kills the regex value upon
>>>                 saving.  Also, you cannot add the regex value back
>>>                 into the row.  All other values appear to be kept.
>>>                 Thanks
>>>                 Jeff
>>>                 On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei
>>>                 Iancu <bogdan at opensips.org
>>>                 <mailto:bogdan at opensips.org>> wrote:
>>>
>>>                     Hi Jeff, I spotted the issue and fixed it on GIT
>>>                     repo. Please give it a try and let me know if it
>>>                     works for you too. Thanks and regards,
>>>
>>>                     Bogdan-Andrei Iancu
>>>                     OpenSIPS Founder and Developer
>>>                     http://www.opensips-solutions.com
>>>                     <http://www.opensips-solutions.com>
>>>
>>>                     On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>>                     Yes, The text box is there but when you place a
>>>>                     value in a preexisting dial plan, the result in
>>>>                     the DB is just the value in the text attribute
>>>>                     field with no dial plan information.  By
>>>>                     editing or adding an attribute it kills the
>>>>                     rest of the dial plan that you were editing and
>>>>                     you cannot add the dial plan information back
>>>>                     in via the CP as long as an attribute in text
>>>>                     form is added.  The same does not happen if
>>>>                     using check box.  I can provide pics if you'd
>>>>                     like.  The only var changed is the
>>>>>
>>>>>                     $dialplan_attributes_mode = 1
>>>>>
>>>>                     Jeff Wilkie
>>>>                     Chief Technology Officer
>>>>                     US IP Communications 919.297.1057
>>>>                     <tel:919.297.1057> /"This e-mail communication
>>>>                     and any attachments may contain confidential
>>>>                     and privileged information and is for use by
>>>>                     the designated addressee(s) named above only.
>>>>                     Any files transmitted with it are confidential
>>>>                     and intended solely for the use of the
>>>>                     individual to whom it is addressed. Any views
>>>>                     or opinions presented are solely those of the
>>>>                     author and do not necessarily represent those
>>>>                     of USIPCOM, LLC. If you are not the intended
>>>>                     addressee, you are hereby notified that you
>>>>                     have received this communication in error and
>>>>                     that any use or reproduction of this email or
>>>>                     its contents is strictly prohibited and may be
>>>>                     unlawful. If you have received this
>>>>                     communication in error, please notify us
>>>>                     immediately by replying to this message and
>>>>                     deleting it from your computer. Thank you". /
>>>>                     On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei
>>>>                     Iancu <bogdan at opensips.org
>>>>                     <mailto:bogdan at opensips.org>> wrote:
>>>>
>>>>                         Hi Jeff, You mean you do not get the
>>>>                         Attributes text area at all in the Edit
>>>>                         section ? Regards,
>>>>
>>>>                         Bogdan-Andrei Iancu
>>>>                         OpenSIPS Founder and Developer
>>>>                         http://www.opensips-solutions.com
>>>>                         <http://www.opensips-solutions.com>
>>>>
>>>>                         On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>>                         New small issue with DIALPLAN in the CP. 
>>>>>                         Running OSIPS 1.10 with CP 5.0
>>>>>                         It appears that when modifying the CP for
>>>>>                         Dialplan attributes as text, they do not
>>>>>                         appear in the edit section once applied. 
>>>>>                         The attributes do show up in the view list
>>>>>                         of dial plans but if you go back to edit
>>>>>                         the attrs again, they are not there.  I
>>>>>                         dot not have the same problem when using
>>>>>                         attrs as checkboxes.  Is there more code
>>>>>                         in the local files that require editing to
>>>>>                         display these values?  Also, if an entry
>>>>>                         in the Dialplan was once a checkbox value
>>>>>                         but I've not converted the mode to text,
>>>>>                         the value does not show up in the text box
>>>>>                         to be edited.
>>>>>
>>>>>                         # Attention : advanced options !!
>>>>>
>>>>>                          //database tables
>>>>>
>>>>>                          $config->table_dialplan = "dialplan";
>>>>>
>>>>>                          $config->results_per_page = 20;
>>>>>
>>>>>                          $config->results_page_range = 5;
>>>>>
>>>>>                          $config->attrs_cb=array(
>>>>>
>>>>>                                                               //
>>>>>                         name , description
>>>>>
>>>>>                                                              
>>>>>                         array("a","Descr a"),
>>>>>
>>>>>                                                              
>>>>>                         array("b","Descr b"),
>>>>>
>>>>>                                                              
>>>>>                         array("c","Descr c"),
>>>>>
>>>>>                                                              
>>>>>                         array("d","Descr d"),
>>>>>
>>>>>                                                              
>>>>>                         array("e","Descr e"),
>>>>>
>>>>>                                                              
>>>>>                         array("f","Descr f"),
>>>>>
>>>>>                                                               );
>>>>>
>>>>>                          $config->cb_per_row = 3;
>>>>>
>>>>>                          $talk_to_this_assoc_id = 1 ;
>>>>>
>>>>>                         // Dialplan - Add/Edit new Translation
>>>>>                         Rule - Attributes input type
>>>>>
>>>>>                         // 0 - checkboxes
>>>>>
>>>>>                         // 1 - text
>>>>>
>>>>>                         $dialplan_attributes_mode = 1  ;
>>>>>
>>>>>                         ?>
>>>>>
>>>>>                         Jeff
>>>>>
>>>>>                         _______________________________________________
>>>>>                         Users mailing list
>>>>>                         Users at lists.opensips.org
>>>>>                         <mailto:Users at lists.opensips.org>
>>>>>                         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>                         <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>>
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From Agalya_Ramachandran at comcast.com  Fri Aug 26 15:17:57 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Fri, 26 Aug 2016 13:17:57 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
Message-ID: <da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>

Yes Bogdan , I do get those warnings.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 5:20 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before those warnings, have you got an warning like:
        WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled for....

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Does OpenSIPS recover from this ?
Answer: It depends upon the volume of the call. If am sending too many calls (may be more than 50 calls), OpenSIPS is not recovering. And am restarting
But if I send less than 10 calls then it is OK.

Have you seen other job reported, besides "tm-utimer" ?
Answer: I also have "blcore-expire" in logs, like once every 10 times of "tm-utimer"

I also observed that increasing the children process in config file, is solving this issue.

P.S:

e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to keep children as 12.
     For even 10 calls, if I keep children as 4, 4 to 6 calls are passing and other calls are failing.
Why do we need to increase the children process?

Just to confirm, is opensips 2.1.4 version  a stable release?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 10:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Question : after you get these logs, you mentioned the inability of handling new calls. Does OpenSIPS recover from this ? or you have to restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya





_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From jwilkie at usipcom.com  Fri Aug 26 19:05:20 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Fri, 26 Aug 2016 13:05:20 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
Message-ID: <CANFkrT1LmR8nZc8g4oXjhBeGRySa+=Bo9U959dDf-Za_T1w6qQ@mail.gmail.com>

I'll check the versions.  As for the flag field in dialplan,  I guess the
documents are wrong?  I show it as flag field in 1.10.x which I am
currently running

http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html#id293827

Thanks

Jeff
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From Agalya_Ramachandran at comcast.com  Fri Aug 26 20:29:03 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Fri, 26 Aug 2016 18:29:03 +0000
Subject: [OpenSIPS-Users] FW:  Asynchronous operation for REST queries
References: <105bbed280a54ae6bf96fc28ec5a7076@COPDCEX28.cable.comcast.com>
	<d7de7fc9-c336-77d2-0aef-d7ee7fc9a8b8@opensips.org>
	<3cd6edaf3dde4140851f87b7411ac805@COPDCEX28.cable.comcast.com>
	<74027c08-f423-f519-a55b-3b72695fa581@opensips.org> 
Message-ID: <8e9bb6cd812546eaa373203c73e2bd46@COPDCEX28.cable.comcast.com>

Hi Liviu,

Here is the dump from the time when the socket is opened and when the request sent out.
Highlighted in RED color. Any clue for this 10 sec delay?

17:29:56.430379 IP6 sturn-test.novalocal.44797 > destination.rtc.sys.comcast.net.webcache: Flags [S], seq 3356294440, win 28800, options [mss 1440,sackOK,TS val 3588300781 ecr 0,nop,wscale 7], length 0
        0x0000:  6000 0000 0028 0640 2001 0558 fc0b 0100  `....(. at ...X<mailto:. at ...X>....
        0x0010:  f816 3eff fed5 0bfe 2001 0558 fc11 0000  ..>........X....
        0x0020:  f816 3eff fe11 d5aa aefd 1f90 c80c fd28  ..>............(
        0x0030:  0000 0000 a002 7080 90bb 0000 0204 05a0  ......p.........
        0x0040:  0402 080a d5e1 1fed 0000 0000 0103 0307  ................
17:29:56.436526 IP6 destination.rtc.sys.comcast.net.webcache > sturn-test.novalocal.44797: Flags [S.], seq 4065018495, ack 3356294441, win 28560, options [mss 1440,sackOK,TS val 1771678349 ecr 3588300781,nop,wscale 9], length 0
        0x0000:  6000 0000 0028 063a 2001 0558 fc11 0000  `....(.:...X....
        0x0010:  f816 3eff fe11 d5aa 2001 0558 fc0b 0100  ..>........X....
        0x0020:  f816 3eff fed5 0bfe 1f90 aefd f24b 427f  ..>..........KB.
        0x0030:  c80c fd29 a012 6f90 755f 0000 0204 05a0  ...)..o.u_......
        0x0040:  0402 080a 6999 aa8d d5e1 1fed 0103 0309  ....i...........
17:29:56.436575 IP6 sturn-test.novalocal.44797 > destination.rtc.sys.comcast.net.webcache: Flags [.], ack 1, win 225, options [nop,nop,TS val 3588300787 ecr 1771678349], length 0
        0x0000:  6000 0000 0020 0640 2001 0558 fc0b 0100  `...... at ...X....
        0x0010:  f816 3eff fed5 0bfe 2001 0558 fc11 0000  ..>........X....
        0x0020:  f816 3eff fe11 d5aa aefd 1f90 c80c fd29  ..>............)
        0x0030:  f24b 4280 8010 00e1 90b3 0000 0101 080a  .KB.............
        0x0040:  d5e1 1ff3 6999 aa8d                      ....i...

17:30:06.667279 IP6 sturn-test.novalocal.44797 > destination.rtc.sys.comcast.net.webcache: Flags [P.], seq 1:681, ack 1, win 225, options [nop,nop,TS val 3588311018 ecr 1771678349], length 680
PUT./RTCGSessionManager/rest/tel/session/createroom?uid=pp&custguid=1-32750 at XX.YYY.231.239..HTTP/1.1..Host:.tr-sm-as-a-001.rtc.sys.comcast.net:8080..Accept:.*/*..Conten<mailto:PUT./RTCGSessionManager/rest/tel/session/createroom?uid=pp&custguid=1-32750 at XX.YYY.231.239..HTTP/1.1..Host:.tr-sm-as-a-001.rtc.sys.comcast.net:8080..Accept:.*/*..Conten> t-Type:.application/json..Uid:.pp..serviceId:.tel..custguid:.1-32750 at 96.119.231. 239...x-tracking-id:.820..X-Trace:.trace-id=1-32750 at 96.119.231.239...x-server-name:.RTCGSM..x-client-name:.IMS
x-source-id:.OpenSIP..user-agent:.OpenSIPv1.11.3..device-id:.1-32750 at XX.YYY.231. 239...Content-Length:.211....{"participantsInfo" :["2155551212"],"callType":"audio","instanceId": "1-32750 at XX.YYY. 231.239.<mailto:1-32750 at XX.YYY.%20231.239.>","fromUID":"pp","fromTN":"pp","isOpenSipRequest":true," displayName":"Call.From.IMS","deviceType":"Android/iOS/STB/IMS"}

17:30:06.672819 IP6 destination.rtc.sys.comcast.net.webcache > sturn-test.novalocal.44797: Flags [.], ack 681, win 59, options [nop,nop,TS val 1771688585 ecr 3588311018], length 0
        0x0000:  6000 0000 0020 063a 2001 0558 fc11 0000  `......:...X....
        0x0010:  f816 3eff fe11 d5aa 2001 0558 fc0b 0100  ..>........X....
        0x0020:  f816 3eff fed5 0bfe 1f90 aefd f24b 4280  ..>..........KB.
        0x0030:  c80c ffd1 8010 003b c0cd 0000 0101 080a  .......;........
        0x0040:  6999 d289 d5e1 47ea                      i.....G.
17:30:06.778978 IP6 destination.rtc.sys.comcast.net.webcache > sturn-test.novalocal.44797: Flags [P.], seq 1:358, ack 681, win 59, options [nop,nop,TS val 1771688691 ecr 3588311018], length 357
        0x0000:  6000 0000 0185 063a 2001 0558 fc11 0000  `......:...X....
        0x0010:  f816 3eff fe11 d5aa 2001 0558 fc0b 0100  ..>........X....
        0x0020:  f816 3eff fed5 0bfe 1f90 aefd f24b 4280  ..>..........KB.
        0x0030:  c80c ffd1 8018 003b 8583 0000 0101 080a  .......;........
        0x0040:  6999 d2f3 d5e1 47ea 4854 5450 2f31 2e31  i.....G.HTTP/1.1
        0x0050:  2032 3030 204f 4b0d 0a53 6572 7665 723a  .200.OK..Server:

...

Regards,
Agalya
From: Ramachandran, Agalya (Contractor)
Sent: Friday, August 26, 2016 12:01 PM
To: 'Liviu Chircu' <liviu at opensips.org>
Subject: RE: FW: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Liviu,

As I already emailed you, to contribute the source code there is some approval process in our organization.
If that goes though I can contribute the REST PUT code.

But I have a question in async operation in opensips.
When a async mode REST API is called either POST/PUT, I see that opensips waits for 10 sec of time and then only it sends out the HTTP request out to the network.
Why there is a huge wait time of 10sec?

When I tried to debug, I found it is looping inside the below for loop for long time in rest_methods.c  file, (but all that looping are taking same seconds, later 10 sec am getting HTTP response)
for (i = 0; i < busy_wait; i += check_time) {.....}

Wondering and interested to know about this delay time.
Please bear some of your time:). Thanks in advance.

Regards,
Agalya

From: Liviu Chircu [mailto:liviu at opensips.org]
Sent: Thursday, August 18, 2016 10:14 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com<mailto:Agalya_Ramachandran at comcast.com>>
Subject: Re: FW: [OpenSIPS-Users] Asynchronous operation for REST queries


Hi Agalya,

I am very busy these days and cannot afford to troubleshoot random problems with libcurl usage. Do you intend to open source the PUT function by making a Pull Request, after we fix it? If yes, it will definitely go up in my priority queue.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 18.08.2016 16:13, Ramachandran, Agalya (Contractor) wrote:
Hi Liviu,

Please help me by giving some clue that what am missing on the below topic.
Your guidance here would be much helpful for me.

Regards,
Agalya

From: Ramachandran, Agalya (Contractor)
Sent: Tuesday, August 16, 2016 10:54 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: RE: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Bogdan/Liviu,

I have downloaded 2.1.4 version if opensips and trying to use rest_put call in async mode.

I have added the necessary code for rest_put () similar to rest_post() in rest_methods.c, rest_methods.h and rest_client.c
If I just call, rest_put from config file, without async it works fine. I could see rest_put() is called and do the necessary steps.

And am trying to call async from config file as,
 async(rest_put("http://***url*** <http://***url***%20> ", "$fU,$tU,$ci ", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume_put);

Further debugging I got some info that I can share with you so that you can help me out easily.
Instead of calling rest_put asynchronously I have tried to call rest_post async . I could able to see that HTTP request went out(from Wireshark traces) and I got 4xx response for the same.

With my code changes in place and when I call rest_put, am not seeing HTTP request sent out.
Assuming rest_put only is not working, there should be some libcurl stuffs not handled properly in 'start_async_http_req' for 'REST_CLIENT_PUT' case.

Am attaching the files that  contains code changes for rest_put only. I have removed other code in order to reduce the file size. Can you please take a look guide me here?.
Please let me know if you need more information on this.

Regards,
Agalya


From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Friday, August 05, 2016 4:47 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Asynchronous operation for REST queries

Hi Agalya,

1.11 has no async support at core level. If you need that, you will have to use 2.1+

Async rest_put() is currently not implemented, but its logic would be fairly similar to the GET and POST counterparts.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 04.08.2016 23:14, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips1.11.3 version for our project.
I have a question here. Right now we are using a REST API call from config file for our project requirement and it is in synchronous mode currently.

From jim at devito.cc  Fri Aug 26 20:32:30 2016
From: jim at devito.cc (Jim DeVito)
Date: Fri, 26 Aug 2016 11:32:30 -0700
Subject: [OpenSIPS-Users] Help responding to 401 in reply route.
Message-ID: <304596bb9fd582365fe69330b78d4378@mail.devito.cc>

Hi All,

I've been able to REGISTER a UAC aginst the subscriber tables and save 
it's info to the location tables. I am then able to use the aliasdb 
module to to match an incoming DID to a subscriber and direct it to the 
UAC. My problem comes when the UAC challenges OpenSIPS for auth on that 
initial INVITE. I'm sure I need to load an on reply route to handle the 
401 from the UAC but I'm unsure what I need to do in that route to get 
the proper credentials from the subscriber table and send a new INVITE 
to the UAC with the proper auth headers.

If some one could point me in the right direction that would be helpful.

Thanks!!

-- 
Jim DeVito


From jwilkie at usipcom.com  Fri Aug 26 22:25:10 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Fri, 26 Aug 2016 16:25:10 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <0aec6176-e0b4-8e53-3528-85046b09dee2@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT25RE+6fP5O-6wJt_muV1r6kjaj_Rpfna8P=LCsxx8xzg@mail.gmail.com>
	<0aec6176-e0b4-8e53-3528-85046b09dee2@opensips.org>
Message-ID: <CANFkrT0oqvBbwNnG_R04oz-7zD4nWqhk7-KAMpPckuKAUCNkxQ@mail.gmail.com>

Attempting to edit any of these fields produces these warnings and null
entries.

Warning: mysql_real_escape_string() expects parameter 2 to be resource,
object given in /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
on line 292 Warning: mysql_real_escape_string() expects parameter 2 to be
resource, object given in
/var/www/opensips_cp/web/tools/system/dialplan/dialplan.php on line 320
Warning: mysql_real_escape_string() expects parameter 2 to be resource,
object given in /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
on line 322

All three of these errors reference the $link resource pulled from the lib
dir.

"pear/MDB2", version 2.4.1

I just updated all the others but still get the same error.

Jeff
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From jwilkie at usipcom.com  Mon Aug 29 23:41:32 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Mon, 29 Aug 2016 17:41:32 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
Message-ID: <CANFkrT2j38nm0U+Ea2go-+ULD3nKtYwrojwFhdnrvknOUmw6tA@mail.gmail.com>

Got this from a var dump on $link when i process if !is_resource.  Not sure
if this actually provides any information that may help.  I'm digging
online to see why the $link is not connecting as a resource in this portion
for whatever reason:

Warning: mysql_real_escape_string() expects parameter 2 to be resource,
object given in /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
on line 288

array(0) { }
object(MDB2_Driver_mysql)#2 (36) {
   ["string_quoting"]=> array(4) {
        ["start"]=> string(1) "'"
        ["end"]=> string(1) "'"
        ["escape"]=> string(1) "\"
        ["escape_pattern"]=> string(1) "\" }
   ["identifier_quoting"]=> array(3) {
        ["start"]=> string(1) "`"
        ["end"]=> string(1) "`"
        ["escape"]=> string(1) "`" }
   ["sql_comments"]=> array(3) {
        [0]=> array(3) {
            ["start"]=> string(3) "-- "
            ["end"]=> string(1) " "
            ["escape"]=> bool(false) }
        [1]=> array(3) {
            ["start"]=> string(1) "#"
            ["end"]=> string(1) " "
            ["escape"]=> bool(false) }
        [2]=> array(3) {
            ["start"]=> string(2) "/*"
            ["end"]=> string(2) "*/"
            ["escape"]=> bool(false) } }
    ["start_transaction"]=> bool(true)
    ["varchar_max_length"]=> int(65532)
    ["db_index"]=> int(1)
    ["dsn"]=> array(10) {
         ["phptype"]=> string(5) "mysql"
         ["dbsyntax"]=> string(5) "mysql"
         ["username"]=> string(10) "xxxxxxxx"
         ["password"]=> string(12) "xxxxxxxx"
         ["protocol"]=> string(3) "tcp"
         ["hostspec"]=> string(9) "localhost"
         ["port"]=> bool(false)
         ["socket"]=> bool(false)
         ["database"]=> bool(false)
         ["mode"]=> bool(false) }
    ["connected_dsn"]=> array(10) {
         ["phptype"]=> string(5) "mysql"
         ["dbsyntax"]=> string(5) "mysql"
         ["username"]=> string(10) "xxxxxx"
         ["password"]=> string(12) "xxxxxx"
         ["protocol"]=> string(3) "tcp"
         ["hostspec"]=> string(9) "localhost"
         ["port"]=> bool(false)
         ["socket"]=> bool(false)
         ["database"]=> bool(false)
         ["mode"]=> bool(false) }
    ["connection"]=> resource(16) of type (mysql link)
    ["opened_persistent"]=> bool(false)
    ["database_name"]=> string(10) "xxxxxxxxx"
    ["connected_database_name"]=> string(10) "xxxxxxxxx"
["connected_server_info"]=> string(10) "5.1.73-log"
["supported"]=> array(19) {
["sequences"]=> string(8) "emulated"
["indexes"]=> bool(true)
["affected_rows"]=> bool(true)
["summary_functions"]=> bool(true)
["order_by_text"]=> bool(true)
["transactions"]=> bool(true)
["savepoints"]=> bool(true)
["current_id"]=> string(8) "emulated"
["limit_queries"]=> bool(true)
 ["LOBs"]=> bool(true)
["replace"]=> bool(true)
["sub_selects"]=> bool(true)
["auto_increment"]=> bool(true)
["primary_key"]=> bool(true)
["result_introspection"]=> bool(true)
["prepared_statements"]=> bool(true)
["identifier_quoting"]=> bool(true)
["pattern_escaping"]=> bool(true)
["new_link"]=> bool(true) }
["options"]=> array(30) {
["ssl"]=> bool(false)
["field_case"]=> int(0)
["disable_query"]=> bool(false)
 ["result_class"]=> string(14) "MDB2_Result_%s"
["buffered_result_class"]=> string(22) "MDB2_BufferedResult_%s"
["result_wrap_class"]=> bool(false)
["result_buffering"]=> bool(true)
["fetch_class"]=> string(8) "stdClass"
["persistent"]=> bool(false)
["debug"]=> int(0)
["debug_handler"]=> string(23) "MDB2_defaultDebugOutput"
["debug_expanded_output"]=> bool(false)
["default_text_field_length"]=> int(4096)
["lob_buffer_length"]=> int(8192)
["log_line_break"]=> string(1) " "
["idxname_format"]=> string(6) "%s_idx"
["seqname_format"]=> string(6) "%s_seq"
 ["savepoint_format"]=> string(17) "MDB2_SAVEPOINT_%s"
["statement_format"]=> string(24) "MDB2_STATEMENT_%1$s_%2$s"
["seqcol_name"]=> string(8) "sequence"
["quote_identifier"]=> bool(false)
["use_transactions"]=> bool(true)
["decimal_places"]=> int(2)
["portability"]=> int(127)
["modules"]=> array(6) {
["ex"]=> string(8) "Extended"
["dt"]=> string(8) "Datatype"
 ["mg"]=> string(7) "Manager"
["rv"]=> string(7) "Reverse"
["na"]=> string(6) "Native"
["fc"]=> string(8) "Function" }
["emulate_prepared"]=> bool(false)
["datatype_map"]=> array(0) { }
["datatype_map_callback"]=> array(0) { }
["nativetype_map_callback"]=> array(0) { }
["default_table_type"]=> string(0) "" }
["wildcards"]=> array(2) {
[0]=> string(1) "%"
[1]=> string(1) "_" }
["as_keyword"]=> string(4) " AS "
["warnings"]=> array(0) { }
["debug_output"]=> string(0) ""
["in_transaction"]=> NULL
["nested_transaction_counter"]=> NULL
["has_transaction_error"]=> bool(false)
["offset"]=> int(0) ["limit"]=> int(0)
["phptype"]=> string(5) "mysql"
["dbsyntax"]=> string(5) "mysql"
["last_query"]=> NULL
["fetchmode"]=> int(2)
["modules"]=> array(0) { }
["destructor_registered"]=> bool(true)
["_debug"]=> bool(false)
["_default_error_mode"]=> NULL
["_default_error_options"]=> NULL
["_default_error_handler"]=> string(0) ""
["_error_class"]=> string(10) "PEAR_Error"
["_expected_errors"]=> array(0) { } }

Jeff
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From jwilkie at usipcom.com  Fri Aug 26 16:38:43 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Fri, 26 Aug 2016 10:38:43 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
Message-ID: <CANFkrT1MsW=LvBvBYroMR_-zHLAKDP2c7Okq0-wan=O9TALD+A@mail.gmail.com>

I'll check the versions.  As for the flag field in dialplan,  I guess the
documents are wrong?  I show it as flag field in 1.10.x which I am
currently running

http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html#id293827


Jeff

On Fri, Aug 26, 2016 at 5:24 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jeff,
>
> Maybe the version of MDB2 or MDB2::mysql php pear ?
>
> The CP versions are correlated with the OpenSIPS versions. The len field
> was moved to a flag field (in dialplan table) in opensips 1.11,
> corresponding to CP 6.0 (6.1 matches opensips 2.1 and CP trunk matches the
> 2.2)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 26.08.2016 08:33, Jeff Wilkie wrote:
>
> I just loaded all the web directory content from git to test but the
> results are the same.  The config dir for dialplan are the same other than
> the toggle for 1 on text attributes.  Are there other files/dirs that would
> affect the dial plan section of CP?  Also, the git download does not
> include the changes from expression string length to expression flags (5.0
> cp)
>
> Jeff
>
>
> On Thu, Aug 25, 2016 at 5:11 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi Jeff,
>>
>> I run some test with the exact code on GITHUB (branch 5.0) and the edit
>> actually worked well (without any further changes).
>>
>> The $link is a variable that is initialized in the included file
>> lib/db_connect.php, so it valid (anyhow, it is used later for the actual
>> query, which works ;) ).
>>
>> Could you check if the return of mysql_real_escape_string() is actually
>> something valid or is it FALSE ?
>>     http://stackoverflow.com/questions/3005135/mysql-real-escape
>> -string-just-makes-an-empty-string
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 25.08.2016 05:33, Jeff Wilkie wrote:
>>
>> If I modify the file /var/www/opensips_cp/web/
>> tools/system/dialplan/dialplan.php and comment out the following:
>>
>>         //              if(get_magic_quotes_gpc()==0){
>>
>>         //                      $match_exp =
>> mysql_real_escape_string($match_exp, *$link*);
>>
>>         //              }
>> The updates work fine.  I think noticed that the ref to *$link* sits
>> below this line in the "modify" section so I decided to remove the $link
>> ref in the above lines and then uncommented the lines leaving the modify
>> section like this
>>
>>                 if ($errors=="") {
>>
>>                         if(get_magic_quotes_gpc()==0){
>>
>>                                 $match_exp =
>> mysql_real_escape_string($match_exp);
>>
>>                         }
>>
>>                         $sql = "SELECT * FROM ".$table.
>>
>>                         " WHERE dpid=" .$dpid. " AND match_exp='"
>> .$match_exp. "'".
>>
>>                         " AND id!=".$id;
>>
>>                         $resultset = *$link*->queryAll($sql);
>> Let me know if I've broken something as a result but this appears to have
>> corrected the problem I was having.  Unknown of any side effects currently
>> :)  Take a look and let me know your thoughts.
>> Jeff
>> On Wed, Aug 24, 2016 at 6:22 PM, Jeff Wilkie <jwilkie at usipcom.com>
>> wrote:
>>>
>>> Here is what it looks like to my db when I create a DialPlan
>>>
>>>   20 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='877'
>>>
>>>   20 Query PREPARE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>>> FROM 'INSERT INTO dialplan\n (dpid, pr, match_op, match_exp,
>>> match_flags, subst_exp, \n repl_exp, attrs) VALUES \n (36, 0,1,
>>> \'877\',0,\'\',\'\',\'.30\')'
>>>
>>>   20 Prepare INSERT INTO dialplan
>>>
>>> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>>
>>> repl_exp, attrs) VALUES
>>>
>>> (36, 0,1, '877',0,'','','.30')
>>>
>>>   20 Query EXECUTE MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>>>
>>>   20 Execute INSERT INTO dialplan
>>>
>>> (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>>
>>> repl_exp, attrs) VALUES
>>>
>>> (36, 0,1, '877',0,'','','.30')
>>> This is what happens when I edit that same DialPlan
>>>
>>>   23 Query select * from dialplan where id='9030'
>>>
>>>   23 Quit
>>>
>>>   24 Query SELECT * FROM dialplan WHERE dpid=36 AND match_exp='' AND
>>> id!=9030
>>>
>>>   24 Query PREPARE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>> FROM 'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1, match_exp =\'\',
>>> match_flags=0, subst_exp = \'\', repl_exp=\'\', attrs= \'.30\' WHERE
>>> id=9030'
>>>
>>>   24 Prepare UPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>>> match_exp ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30'
>>> WHERE id=9030
>>>
>>>   24 Query EXECUTE MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>>
>>>   24 Execute UPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>>> match_exp ='', match_flags=0, subst_exp = '', repl_exp='', attrs= '.30'
>>> WHERE id=9030
>>>
>>>   24 Query DEALLOCATE PREPARE MDB2_STATEMENT_mysql_880efa45d
>>> 78ebff7cbf5f617428df4d2
>>>
>>>   24 Quit
>>> You can see that the container for match_ex is not getting passed to the
>>> '24' SELECT statement which ends in a 0 results query apparently.
>>> Thanks
>>> Jeff
>>> On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie <jwilkie at usipcom.com>
>>> wrote:
>>>>
>>>> Yes.  After any edits followed by a save with the attribute field
>>>> active as text causes "Matching Regular Expression" to be deleted.
>>>> Thanks
>>>> Jeff
>>>> On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu <
>>>> bogdan at opensips.org> wrote:
>>>>
>>>>> Jeff, You mean "Matching Regular Expression", right ? upon performing
>>>>> an edit + save where you only changed the attrs, the "Matching
>>>>> Regular Expression" gets to empty ? Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>> On 24.08.2016 23:25, Jeff Wilkie wrote:
>>>>>
>>>>> Ok.  The value now displays but if you modify the value and hit save,
>>>>> it kills the regex value upon saving.  Also, you cannot add the regex value
>>>>> back into the row.  All other values appear to be kept.
>>>>> Thanks
>>>>> Jeff
>>>>> On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei Iancu <
>>>>> bogdan at opensips.org> wrote:
>>>>>>
>>>>>> Hi Jeff, I spotted the issue and fixed it on GIT repo. Please give it
>>>>>> a try and let me know if it works for you too. Thanks and regards,
>>>>>>
>>>>>> Bogdan-Andrei Iancu
>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>
>>>>>> On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>>>>
>>>>>> Yes, The text box is there but when you place a value in a
>>>>>> preexisting dial plan, the result in the DB is just the value in the text
>>>>>> attribute field with no dial plan information.  By editing or adding an
>>>>>> attribute it kills the rest of the dial plan that you were editing and you
>>>>>> cannot add the dial plan information back in via the CP as long as an
>>>>>> attribute in text form is added.  The same does not happen if using check
>>>>>> box.  I can provide pics if you'd like.  The only var changed is the
>>>>>>
>>>>>> $dialplan_attributes_mode = 1
>>>>>>
>>>>>> Jeff Wilkie
>>>>>> Chief Technology Officer
>>>>>> US IP Communications 919.297.1057 *"This e-mail communication and
>>>>>> any attachments may contain confidential and privileged information and is
>>>>>> for use by the designated addressee(s) named above only. Any files
>>>>>> transmitted with it are confidential and intended solely for the use of the
>>>>>> individual to whom it is addressed. Any views or opinions presented are
>>>>>> solely those of the author and do not necessarily represent those of
>>>>>> USIPCOM, LLC. If you are not the intended addressee, you are hereby
>>>>>> notified that you have received this communication in error and that any
>>>>>> use or reproduction of this email or its contents is strictly prohibited
>>>>>> and may be unlawful. If you have received this communication in error,
>>>>>> please notify us immediately by replying to this message and deleting it
>>>>>> from your computer. Thank you". *
>>>>>> On Wed, Aug 24, 2016 at 5:20 AM, Bogdan-Andrei Iancu <
>>>>>> bogdan at opensips.org> wrote:
>>>>>>>
>>>>>>> Hi Jeff, You mean you do not get the Attributes text area at all in
>>>>>>> the Edit section ? Regards,
>>>>>>>
>>>>>>> Bogdan-Andrei Iancu
>>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>>
>>>>>>> On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>>>>
>>>>>>> New small issue with DIALPLAN in the CP.  Running OSIPS 1.10 with CP
>>>>>>> 5.0
>>>>>>> It appears that when modifying the CP for Dialplan attributes as
>>>>>>> text, they do not appear in the edit section once applied.  The attributes
>>>>>>> do show up in the view list of dial plans but if you go back to edit the
>>>>>>> attrs again, they are not there.  I dot not have the same problem when
>>>>>>> using attrs as checkboxes.  Is there more code in the local files that
>>>>>>> require editing to display these values?  Also, if an entry in the Dialplan
>>>>>>> was once a checkbox value but I've not converted the mode to text, the
>>>>>>> value does not show up in the text box to be edited.
>>>>>>>
>>>>>>> # Attention : advanced options !!
>>>>>>>
>>>>>>>  //database tables
>>>>>>>
>>>>>>>  $config->table_dialplan = "dialplan";
>>>>>>>
>>>>>>>  $config->results_per_page = 20;
>>>>>>>
>>>>>>>  $config->results_page_range = 5;
>>>>>>>
>>>>>>>  $config->attrs_cb=array(
>>>>>>>
>>>>>>>                                         // name , description
>>>>>>>
>>>>>>>                                         array("a","Descr a"),
>>>>>>>
>>>>>>>                                         array("b","Descr b"),
>>>>>>>
>>>>>>>                                         array("c","Descr c"),
>>>>>>>
>>>>>>>                                         array("d","Descr d"),
>>>>>>>
>>>>>>>                                         array("e","Descr e"),
>>>>>>>
>>>>>>>                                         array("f","Descr f"),
>>>>>>>
>>>>>>>                                         );
>>>>>>>
>>>>>>>  $config->cb_per_row = 3;
>>>>>>>
>>>>>>>  $talk_to_this_assoc_id = 1 ;
>>>>>>>
>>>>>>> // Dialplan - Add/Edit new Translation Rule - Attributes input type
>>>>>>>
>>>>>>> // 0 - checkboxes
>>>>>>>
>>>>>>> // 1 - text
>>>>>>>
>>>>>>> $dialplan_attributes_mode = 1  ;
>>>>>>>
>>>>>>> ?>
>>>>>>> Jeff
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
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From bogdan at opensips.org  Mon Aug 29 09:56:52 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 29 Aug 2016 10:56:52 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT1MsW=LvBvBYroMR_-zHLAKDP2c7Okq0-wan=O9TALD+A@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
	<CANFkrT1MsW=LvBvBYroMR_-zHLAKDP2c7Okq0-wan=O9TALD+A@mail.gmail.com>
Message-ID: <c5ae91e5-2c48-56bd-88fc-f80db93dd5e4@opensips.org>

My bad, the len was replaced with flags between 1.8 and 1.9 :)

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.08.2016 17:38, Jeff Wilkie wrote:
> I'll check the versions.  As for the flag field in dialplan,  I guess 
> the documents are wrong?  I show it as flag field in 1.10.x which I am 
> currently running
>
> http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html#id293827
>
>
> Jeff
>
> On Fri, Aug 26, 2016 at 5:24 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Jeff,
>
>     Maybe the version of MDB2 or MDB2::mysql php pear ?
>
>     The CP versions are correlated with the OpenSIPS versions. The len
>     field was moved to a flag field (in dialplan table) in opensips
>     1.11, corresponding to CP 6.0 (6.1 matches opensips 2.1 and CP
>     trunk matches the 2.2)
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://www.opensips-solutions.com>
>
>     On 26.08.2016 08:33, Jeff Wilkie wrote:
>>     I just loaded all the web directory content from git to test but
>>     the results are the same.  The config dir for dialplan are the
>>     same other than the toggle for 1 on text attributes.  Are there
>>     other files/dirs that would affect the dial plan section of CP? 
>>     Also, the git download does not include the changes from
>>     expression string length to expression flags (5.0 cp)
>>     Jeff
>>     On Thu, Aug 25, 2016 at 5:11 PM, Bogdan-Andrei Iancu
>>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>         Hi Jeff, I run some test with the exact code on GITHUB
>>         (branch 5.0) and the edit actually worked well (without any
>>         further changes). The $link is a variable that is initialized
>>         in the included file lib/db_connect.php, so it valid (anyhow,
>>         it is used later for the actual query, which works ;) ).
>>         Could you check if the return of mysql_real_escape_string()
>>         is actually something valid or is it FALSE ?
>>         http://stackoverflow.com/questions/3005135/mysql-real-escape-string-just-makes-an-empty-string
>>         <http://stackoverflow.com/questions/3005135/mysql-real-escape-string-just-makes-an-empty-string>
>>         Regards,
>>
>>         Bogdan-Andrei Iancu
>>         OpenSIPS Founder and Developer
>>         http://www.opensips-solutions.com
>>         <http://www.opensips-solutions.com>
>>
>>         On 25.08.2016 05:33, Jeff Wilkie wrote:
>>>         If I modify the file
>>>         /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
>>>         and comment out the following:
>>>
>>>                 // if(get_magic_quotes_gpc()==0){
>>>
>>>               //                      $match_exp =
>>>         mysql_real_escape_string($match_exp, *$link*);
>>>
>>>                 //              }
>>>
>>>         The updates work fine.  I think noticed that the ref to
>>>         *$link* sits below this line in the "modify" section so I
>>>         decided to remove the $link ref in the above lines and then
>>>         uncommented the lines leaving the modify section like this
>>>
>>>                         if ($errors=="") {
>>>
>>>         if(get_magic_quotes_gpc()==0){
>>>
>>>                                         $match_exp =
>>>         mysql_real_escape_string($match_exp);
>>>
>>>                                 }
>>>
>>>                                 $sql = "SELECT * FROM ".$table.
>>>
>>>                                 " WHERE dpid=" .$dpid. " AND
>>>         match_exp='" .$match_exp. "'".
>>>
>>>                                 " AND id!=".$id;
>>>
>>>                               $resultset = *$link*->queryAll($sql);
>>>
>>>         Let me know if I've broken something as a result but this
>>>         appears to have corrected the problem I was having.  Unknown
>>>         of any side effects currently :)  Take a look and let me
>>>         know your thoughts.
>>>         Jeff
>>>         On Wed, Aug 24, 2016 at 6:22 PM, Jeff Wilkie
>>>         <jwilkie at usipcom.com <mailto:jwilkie at usipcom.com>> wrote:
>>>
>>>             Here is what it looks like to my db when I create a
>>>             DialPlan
>>>
>>>               20 QuerySELECT * FROM dialplan WHERE dpid=36 AND
>>>             match_exp='877'
>>>
>>>               20 QueryPREPARE
>>>             MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>>>             FROM 'INSERT INTO dialplan\n(dpid, pr, match_op,
>>>             match_exp, match_flags, subst_exp, \nrepl_exp, attrs)
>>>             VALUES \n(36, 0,1, \'877\',0,\'\',\'\',\'.30\')'
>>>
>>>               20 PrepareINSERT INTO dialplan
>>>
>>>             (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>>
>>>             repl_exp, attrs) VALUES
>>>
>>>             (36, 0,1, '877',0,'','','.30')
>>>
>>>               20 QueryEXECUTE
>>>             MDB2_STATEMENT_mysql_a9e5e113b009b855c4d690637965bc74
>>>
>>>               20 ExecuteINSERT INTO dialplan
>>>
>>>             (dpid, pr, match_op, match_exp, match_flags, subst_exp,
>>>
>>>             repl_exp, attrs) VALUES
>>>
>>>             (36, 0,1, '877',0,'','','.30')
>>>
>>>             This is what happens when I edit that same DialPlan
>>>
>>>               23 Queryselect * from dialplan where id='9030'
>>>
>>>               23 Quit
>>>
>>>               24 QuerySELECT * FROM dialplan WHERE dpid=36 AND
>>>             match_exp='' AND id!=9030
>>>
>>>               24 QueryPREPARE
>>>             MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>>             FROM 'UPDATE dialplan SET dpid=36, pr = 0, match_op= 1,
>>>             match_exp =\'\', match_flags=0, subst_exp = \'\',
>>>             repl_exp=\'\', attrs= \'.30\' WHERE id=9030'
>>>
>>>               24 PrepareUPDATE dialplan SET dpid=36, pr = 0,
>>>             match_op= 1, match_exp ='', match_flags=0, subst_exp =
>>>             '', repl_exp='', attrs= '.30' WHERE id=9030
>>>
>>>               24 QueryEXECUTE
>>>             MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>>
>>>               24 ExecuteUPDATE dialplan SET dpid=36, pr = 0,
>>>             match_op= 1, match_exp ='', match_flags=0, subst_exp =
>>>             '', repl_exp='', attrs= '.30' WHERE id=9030
>>>
>>>               24 QueryDEALLOCATE PREPARE
>>>             MDB2_STATEMENT_mysql_880efa45d78ebff7cbf5f617428df4d2
>>>
>>>               24 Quit
>>>
>>>             You can see that the container for match_ex is not
>>>             getting passed to the '24' SELECT statement which ends
>>>             in a 0 results query apparently.
>>>             Thanks
>>>             Jeff
>>>             On Wed, Aug 24, 2016 at 5:30 PM, Jeff Wilkie
>>>             <jwilkie at usipcom.com <mailto:jwilkie at usipcom.com>> wrote:
>>>
>>>                 Yes.  After any edits followed by a save with the
>>>                 attribute field active as text causes "Matching
>>>                 Regular Expression" to be deleted.
>>>                 Thanks
>>>                 Jeff
>>>                 On Wed, Aug 24, 2016 at 5:15 PM, Bogdan-Andrei Iancu
>>>                 <bogdan at opensips.org <mailto:bogdan at opensips.org>>
>>>                 wrote:
>>>
>>>                     Jeff, You mean "Matching Regular Expression",
>>>                     right ? upon performing an edit + save where you
>>>                     only changed the attrs, the "Matching Regular
>>>                     Expression" gets to empty ? Regards,
>>>
>>>                     Bogdan-Andrei Iancu
>>>                     OpenSIPS Founder and Developer
>>>                     http://www.opensips-solutions.com
>>>                     <http://www.opensips-solutions.com>
>>>
>>>                     On 24.08.2016 23:25, Jeff Wilkie wrote:
>>>>                     Ok.  The value now displays but if you modify
>>>>                     the value and hit save, it kills the regex
>>>>                     value upon saving.  Also, you cannot add the
>>>>                     regex value back into the row.  All other
>>>>                     values appear to be kept.
>>>>                     Thanks
>>>>                     Jeff
>>>>                     On Wed, Aug 24, 2016 at 11:35 AM, Bogdan-Andrei
>>>>                     Iancu <bogdan at opensips.org
>>>>                     <mailto:bogdan at opensips.org>> wrote:
>>>>
>>>>                         Hi Jeff, I spotted the issue and fixed it
>>>>                         on GIT repo. Please give it a try and let
>>>>                         me know if it works for you too. Thanks and
>>>>                         regards,
>>>>
>>>>                         Bogdan-Andrei Iancu
>>>>                         OpenSIPS Founder and Developer
>>>>                         http://www.opensips-solutions.com
>>>>                         <http://www.opensips-solutions.com>
>>>>
>>>>                         On 24.08.2016 17:46, Jeff Wilkie wrote:
>>>>>                         Yes, The text box is there but when you
>>>>>                         place a value in a preexisting dial plan,
>>>>>                         the result in the DB is just the value in
>>>>>                         the text attribute field with no dial plan
>>>>>                         information.  By editing or adding an
>>>>>                         attribute it kills the rest of the dial
>>>>>                         plan that you were editing and you cannot
>>>>>                         add the dial plan information back in via
>>>>>                         the CP as long as an attribute in text
>>>>>                         form is added.  The same does not happen
>>>>>                         if using check box.  I can provide pics if
>>>>>                         you'd like.  The only var changed is the
>>>>>>
>>>>>>                         $dialplan_attributes_mode = 1
>>>>>>
>>>>>                         Jeff Wilkie
>>>>>                         Chief Technology Officer
>>>>>                         US IP Communications 919.297.1057
>>>>>                         <tel:919.297.1057> /"This e-mail
>>>>>                         communication and any attachments may
>>>>>                         contain confidential and privileged
>>>>>                         information and is for use by the
>>>>>                         designated addressee(s) named above only.
>>>>>                         Any files transmitted with it are
>>>>>                         confidential and intended solely for the
>>>>>                         use of the individual to whom it is
>>>>>                         addressed. Any views or opinions presented
>>>>>                         are solely those of the author and do not
>>>>>                         necessarily represent those of USIPCOM,
>>>>>                         LLC. If you are not the intended
>>>>>                         addressee, you are hereby notified that
>>>>>                         you have received this communication in
>>>>>                         error and that any use or reproduction of
>>>>>                         this email or its contents is strictly
>>>>>                         prohibited and may be unlawful. If you
>>>>>                         have received this communication in error,
>>>>>                         please notify us immediately by replying
>>>>>                         to this message and deleting it from your
>>>>>                         computer. Thank you". /
>>>>>                         On Wed, Aug 24, 2016 at 5:20 AM,
>>>>>                         Bogdan-Andrei Iancu <bogdan at opensips.org
>>>>>                         <mailto:bogdan at opensips.org>> wrote:
>>>>>
>>>>>                             Hi Jeff, You mean you do not get the
>>>>>                             Attributes text area at all in the
>>>>>                             Edit section ? Regards,
>>>>>
>>>>>                             Bogdan-Andrei Iancu
>>>>>                             OpenSIPS Founder and Developer
>>>>>                             http://www.opensips-solutions.com
>>>>>                             <http://www.opensips-solutions.com>
>>>>>
>>>>>                             On 23.08.2016 00:56, Jeff Wilkie wrote:
>>>>>>                             New small issue with DIALPLAN in the
>>>>>>                             CP.  Running OSIPS 1.10 with CP 5.0
>>>>>>                             It appears that when modifying the CP
>>>>>>                             for Dialplan attributes as text, they
>>>>>>                             do not appear in the edit section
>>>>>>                             once applied.  The attributes do show
>>>>>>                             up in the view list of dial plans but
>>>>>>                             if you go back to edit the attrs
>>>>>>                             again, they are not there.  I dot not
>>>>>>                             have the same problem when using
>>>>>>                             attrs as checkboxes.  Is there more
>>>>>>                             code in the local files that require
>>>>>>                             editing to display these values? 
>>>>>>                             Also, if an entry in the Dialplan was
>>>>>>                             once a checkbox value but I've not
>>>>>>                             converted the mode to text, the value
>>>>>>                             does not show up in the text box to
>>>>>>                             be edited.
>>>>>>
>>>>>>                             # Attention : advanced options !!
>>>>>>
>>>>>>                              //database tables
>>>>>>
>>>>>>                              $config->table_dialplan = "dialplan";
>>>>>>
>>>>>>                              $config->results_per_page = 20;
>>>>>>
>>>>>>                              $config->results_page_range = 5;
>>>>>>
>>>>>>                              $config->attrs_cb=array(
>>>>>>
>>>>>>                                                                  
>>>>>>                             // name , description
>>>>>>
>>>>>>                                                                  
>>>>>>                             array("a","Descr a"),
>>>>>>
>>>>>>                                                                  
>>>>>>                             array("b","Descr b"),
>>>>>>
>>>>>>                                                                  
>>>>>>                             array("c","Descr c"),
>>>>>>
>>>>>>                                                                  
>>>>>>                             array("d","Descr d"),
>>>>>>
>>>>>>                                                                  
>>>>>>                             array("e","Descr e"),
>>>>>>
>>>>>>                                                                  
>>>>>>                             array("f","Descr f"),
>>>>>>
>>>>>>                                                                   );
>>>>>>
>>>>>>                              $config->cb_per_row = 3;
>>>>>>
>>>>>>                              $talk_to_this_assoc_id = 1 ;
>>>>>>
>>>>>>                             // Dialplan - Add/Edit new
>>>>>>                             Translation Rule - Attributes input type
>>>>>>
>>>>>>                             // 0 - checkboxes
>>>>>>
>>>>>>                             // 1 - text
>>>>>>
>>>>>>                             $dialplan_attributes_mode = 1  ;
>>>>>>
>>>>>>                             ?>
>>>>>>
>>>>>>                             Jeff
>>>>>>
>>>>>>                             _______________________________________________
>>>>>>                             Users mailing list
>>>>>>                             Users at lists.opensips.org
>>>>>>                             <mailto:Users at lists.opensips.org>
>>>>>>                             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>                             <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>>>
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From bogdan at opensips.org  Tue Aug 30 09:43:43 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Aug 2016 10:43:43 +0300
Subject: [OpenSIPS-Users] Help responding to 401 in reply route.
In-Reply-To: <304596bb9fd582365fe69330b78d4378@mail.devito.cc>
References: <304596bb9fd582365fe69330b78d4378@mail.devito.cc>
Message-ID: <06e5d970-b57a-af82-7aaf-3ffd7b791d3c@opensips.org>

Hi Jim

It is unusual for an UAC to challenge a proxy, still, to get access to 
the 401 and to generate a new INVITE, use a failure_route:
     http://www.opensips.org/Documentation/Script-Routes-2-2#toc3
in combination with the uac and uac_auth modules:
http://www.opensips.org/html/docs/modules/2.2.x/uac.html#id293810

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.08.2016 21:32, Jim DeVito wrote:
> Hi All,
>
> I've been able to REGISTER a UAC aginst the subscriber tables and save 
> it's info to the location tables. I am then able to use the aliasdb 
> module to to match an incoming DID to a subscriber and direct it to 
> the UAC. My problem comes when the UAC challenges OpenSIPS for auth on 
> that initial INVITE. I'm sure I need to load an on reply route to 
> handle the 401 from the UAC but I'm unsure what I need to do in that 
> route to get the proper credentials from the subscriber table and send 
> a new INVITE to the UAC with the proper auth headers.
>
> If some one could point me in the right direction that would be helpful.
>
> Thanks!!
>



From bogdan at opensips.org  Tue Aug 30 10:12:38 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Aug 2016 11:12:38 +0300
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT2j38nm0U+Ea2go-+ULD3nKtYwrojwFhdnrvknOUmw6tA@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
	<CANFkrT2j38nm0U+Ea2go-+ULD3nKtYwrojwFhdnrvknOUmw6tA@mail.gmail.com>
Message-ID: <5b66860a-9256-1552-4a8c-200bb2d0b4bd@opensips.org>

Hi Jeff,

We are getting closer :).

I found this 
http://stackoverflow.com/questions/9014877/mysql-real-escape-string-error-despite-connecting-to-database-what-am-i-doing

where the problem goes into mysql versus mysqli. I guess your MDB2 uses 
mysqli ?? Could you list all the php packages you have on your system 
along with all the php pears (pear list) ?

Also, for testing purposes, try to change |mysql_real_escape_string() 
into ||mysqli_real_escape_string|||()

|Thanks, |

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 30.08.2016 00:41, Jeff Wilkie wrote:
> Got this from a var dump on $link when i process if !is_resource.  Not 
> sure if this actually provides any information that may help.  I'm 
> digging online to see why the $link is not connecting as a resource in 
> this portion for whatever reason:
>
> Warning: mysql_real_escape_string() expects parameter 2 to be 
> resource, object given in 
> /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php on line 288
>
>

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From bogdan at opensips.org  Tue Aug 30 12:20:17 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Aug 2016 13:20:17 +0300
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
Message-ID: <7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>

Hi Agalya,

The message that usually pops up first :
     WARNING:core:utimer task <tm-utimer> already scheduled for....

tells that the given timer routing is slow (tm-utimer in our case)and 
the previous execution is not completed when the next triggering 
happens. This TM routine is responsible for SIP retransmissions.

The second message
     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx 
us delay in execution
tells how long the timer job spent between being scheduled (triggered) 
and being lunched in execution (the 2 events happen in different processes).

In your case, you have 4 processes, one UDP listener and you do 2cps ? 
and you get one of the warnings ? do you use the default script ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
>
> Yes Bogdan , I do get those warnings.
>
> Regards,
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Thursday, August 25, 2016 5:20 PM
> *To:* Ramachandran, Agalya (Contractor) 
> <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list 
> <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
> Hi Agalya,
>
> Before those warnings, have you got an warning like:
>         WARNING:core:utimer_ticker: utimer task <tm-utimer> already 
> scheduled for....
>
> Regards,
>
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi Bogdan,
>
>     *Does OpenSIPS recover from this ?*
>
>     Answer: It depends upon the volume of the call. If am sending too
>     many calls (may be more than 50 calls), OpenSIPS is not
>     recovering. And am restarting
>
>     But if I send less than 10 calls then it is OK.
>
>     *Have you seen other job reported, besides "tm-utimer" ?*
>
>     Answer: I also have ?blcore-expire? in logs, like once every 10
>     times of "tm-utimer"
>
>     I also observed that increasing the children process in config
>     file, is solving this issue.
>
>     P.S:
>
>     e.g : Sending 10 calls to opensips, at 2 calls per sec, then I
>     need to keep children as 12.
>
>          For even 10 calls, if I keep children as 4, 4 to 6 calls are
>     passing and other calls are failing.
>
>     Why do we need to increase the children process?
>
>     Just to confirm, is opensips 2.1.4 version  a stable release?
>
>     Regards,
>     Agalya
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Thursday, August 25, 2016 10:54 AM
>     *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>     <mailto:users at lists.opensips.org>; Ramachandran, Agalya
>     (Contractor) <Agalya_Ramachandran at comcast.com>
>     <mailto:Agalya_Ramachandran at comcast.com>
>     *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>     Hi Agalya,
>
>     Question : after you get these logs, you mentioned the inability
>     of handling new calls. Does OpenSIPS recover from this ? or you
>     have to restart after such an event ?
>
>     Have you seen other job reported, besides "tm-utimer" ?
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
>
>         Hi team,
>
>         Am using opensips 2.1.4 version and trying to do load test.
>         When running the test cases, am getting too many warnings in
>         OpenSIPS log.
>
>         These warnings are coming continuously and its more than 2
>         page in log file.
>
>         Below are the warnings.
>
>         Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>         WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>         39450000 us delay in execution
>
>         Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>         WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>         39450000 us delay in execution
>
>         Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>         WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>         39450000 us delay in execution
>
>         Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>         WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>         39450000 us delay in execution
>
>         Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>         WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>         39450000 us delay in execution
>
>         Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]:
>         WARNING:core:handle_timer_job: utimer job <tm-utimer> has a
>         39450000 us delay in execution
>
>         When I get this Warning, calls are getting failed.
>
>         Why am getting this warning and what could be done to avoid it?
>
>         Regards,
>         Agalya
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Tue Aug 30 12:24:37 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Aug 2016 13:24:37 +0300
Subject: [OpenSIPS-Users] Busy Signal When Using Voicemail
In-Reply-To: <CA+vx6KK6gtzZ4Ts36aOAa4WNOWZzB6A2TYWX7bypto1wZ5gxqg@mail.gmail.com>
References: <CA+vx6KLtHTdRmuO_tuDAdNiG58JPbVCi5wQVbzuGHAj_WuoOFQ@mail.gmail.com>
	<698f4f81-e8f2-ff10-6acb-253d18255079@opensips.org>
	<CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com>
	<507034b8-575f-b8da-35db-e9b36f2977ec@opensips.org>
	<CA+vx6KK6gtzZ4Ts36aOAa4WNOWZzB6A2TYWX7bypto1wZ5gxqg@mail.gmail.com>
Message-ID: <4eae4ca1-9285-97e3-8aa7-7bfddb9597fd@opensips.org>

I do not have to script anything as the end-user (already in a call) is 
generating the 486 reply, not OpenSIPS itself. Make a sip capture and 
check which party is generating the 486.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.08.2016 21:58, Nabeel wrote:
> On 10 August 2016 at 19:38, Bogdan-Andrei Iancu <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Nabeel,
>
>     OpenSIPS does not assume anything by default. If you want to have
>     any new calls to user A rejected (if A already in a call, with
>     other users or any service), you need to script this.
>
>
>
> In the case of user A being in a call with other users, I didn't have 
> to script this in order to receive a 486 busy response.  I don't see 
> this in the cfg script. This seems to be default OpenSIPS behaviour in 
> the case of calling other users.

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From jwilkie at usipcom.com  Tue Aug 30 12:28:45 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Tue, 30 Aug 2016 06:28:45 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <5b66860a-9256-1552-4a8c-200bb2d0b4bd@opensips.org>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
	<CANFkrT2j38nm0U+Ea2go-+ULD3nKtYwrojwFhdnrvknOUmw6tA@mail.gmail.com>
	<5b66860a-9256-1552-4a8c-200bb2d0b4bd@opensips.org>
Message-ID: <CANFkrT2dfhVrY=Fg68tf6=ceKCfDkj32mDMGqUrN6QRj9oZ2+Q@mail.gmail.com>

php.i686                           5.3.3-48.el6_8
@updates

php-cli.i686                       5.3.3-48.el6_8
@updates

php-common.i686                    5.3.3-48.el6_8
@updates

php-gd.i686                        5.3.3-48.el6_8
@updates

php-mysql.i686                     5.3.3-48.el6_8
@updates

php-pdo.i686                       5.3.3-48.el6_8
@updates

php-pear.noarch                    1:1.9.4-5.el6
@base

php-xmlrpc.i686                    5.3.3-48.el6_8
@updates

*Installed packages, channel pear.php.net <http://pear.php.net>:*

=========================================

*Package*           *Version* *State*

Archive_Tar       1.3.7   stable

Console_Getopt    1.2.3   stable

Log               1.12.8  stable

MDB2              2.4.1   stable

MDB2_Driver_mysql 1.4.1   stable

PEAR              1.9.4   stable

Structures_Graph  1.0.4   stable

XML_RPC           1.5.4   stable

XML_Util          1.2.1   stable


freeradius-mysql.i686              2.2.6-6.el6_7
@updates

lua-sql-mysql.i686                 2.1.1-7.el6
@epel

mysql.i686                         5.1.73-7.el6
@base

mysql-devel.i686                   5.1.73-7.el6
@base

mysql-libs.i686                    5.1.73-7.el6
@base

mysql-server.i686                  5.1.73-7.el6
@base

php-mysql.i686                     5.3.3-48.el6_8
@updates


I'll run those tests with mysqli.  I'm not currently using that though.
Will report today.

Jeff Wilkie
Chief Technology Officer
US IP Communications
919.297.1057


*"This e-mail communication and any attachments may contain confidential
and privileged information and is for use by the designated addressee(s)
named above only. Any files transmitted with it are confidential and
intended solely for the use of the individual to whom it is addressed. Any
views or opinions presented are solely those of the author and do not
necessarily represent those of USIPCOM, LLC. If you are not the intended
addressee, you are hereby notified that you have received this
communication in error and that any use or reproduction of this email or
its contents is strictly prohibited and may be unlawful. If you have
received this communication in error, please notify us immediately by
replying to this message and deleting it from your computer. Thank you". *

On Tue, Aug 30, 2016 at 4:12 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jeff,
>
> We are getting closer :).
>
> I found this http://stackoverflow.com/questions/9014877/mysql-real-
> escape-string-error-despite-connecting-to-database-what-am-i-doing
>
> where the problem goes into mysql versus mysqli. I guess your MDB2 uses
> mysqli ?? Could you list all the php packages you have on your system along
> with all the php pears (pear list) ?
>
> Also, for testing purposes, try to change mysql_real_escape_string() into
> mysqli_real_escape_string()
>
>
> Thanks,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 30.08.2016 00:41, Jeff Wilkie wrote:
>
> Got this from a var dump on $link when i process if !is_resource.  Not
> sure if this actually provides any information that may help.  I'm digging
> online to see why the $link is not connecting as a resource in this portion
> for whatever reason:
>
> Warning: mysql_real_escape_string() expects parameter 2 to be resource,
> object given in /var/www/opensips_cp/web/tools/system/dialplan/dialplan.php
> on line 288
>
>
>
>
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From jwilkie at usipcom.com  Tue Aug 30 16:08:43 2016
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Tue, 30 Aug 2016 10:08:43 -0400
Subject: [OpenSIPS-Users] OPENSIPS_CP DIALPLAN
In-Reply-To: <CANFkrT2dfhVrY=Fg68tf6=ceKCfDkj32mDMGqUrN6QRj9oZ2+Q@mail.gmail.com>
References: <CANFkrT2AiFNgU4VG_39PXndxnZwNXfJ4X294dkLp66un3Pv4cg@mail.gmail.com>
	<77fecf3f-7b81-221f-534b-7a3ffd85a6c5@opensips.org>
	<CANFkrT3q2QCiL-nM6TEwDVM74xeNWNkDHYf6Mb2ZiBXxLZoGpw@mail.gmail.com>
	<20922eff-670b-bc86-8ee6-8fe72776fd15@opensips.org>
	<CANFkrT25NbUKiGnsOtEvyvsxSd7gXsutaKD1_O6pC3S_Ytk4+A@mail.gmail.com>
	<fd837ae6-b585-0999-a443-207b6d5a00bb@opensips.org>
	<CANFkrT0YtqdZ_TeM7BytVw0pdygJPKYsBG2c-QH5qbJvUpoTgw@mail.gmail.com>
	<CANFkrT0RyFq+75wbWM20Yf6fg8AD8C6n2WwtSC18GH4FiTbvOA@mail.gmail.com>
	<CANFkrT1bPzv4=Exy0_jjZkNBH-FVpm7U2ieaMyzyEqS+n8t+UQ@mail.gmail.com>
	<d62a4f32-1022-6576-121a-c19b8aa9f49f@opensips.org>
	<CANFkrT1MRTcx4xzGFEo24sZ+_czuhVGJtRAob-b2809tUZbWZQ@mail.gmail.com>
	<f015014d-ed0e-8c95-fe22-7a8619d92a5a@opensips.org>
	<CANFkrT2j38nm0U+Ea2go-+ULD3nKtYwrojwFhdnrvknOUmw6tA@mail.gmail.com>
	<5b66860a-9256-1552-4a8c-200bb2d0b4bd@opensips.org>
	<CANFkrT2dfhVrY=Fg68tf6=ceKCfDkj32mDMGqUrN6QRj9oZ2+Q@mail.gmail.com>
Message-ID: <CANFkrT1ZAqwTDob1-vERU7dCk1OMJnHB1uZ-kWm=r4iO7nhdjQ@mail.gmail.com>

So reading the following:

*Parameters*

*unescaped_string*

The string that is to be escaped.
*link_identifier*

The MySQL connection. If the link identifier is not specified, the last
link opened by mysql_connect
<https://dev.mysql.com/doc/apis-php/en/apis-php-function.mysql-connect.html> is
assumed. If no such link is found, it will try to create one as if
mysql_connect
<https://dev.mysql.com/doc/apis-php/en/apis-php-function.mysql-connect.html>
had
been called with no arguments. If no connection is found or established, an
E_WARNING level error is generated.

*Return Values*

Returns the escaped string, or FALSE on error.

*Errors/Exceptions*

Executing this function without a MySQL connection present will also emit
E_WARNING level PHP errors. Only execute this function with a valid MySQL
connection present.
I decided to remove the $link from each of the sections under the "modify"
portion of the code.  Assuming since in the code we call the DB connection
in the include statement anyway, the $link should already be established
and able to be assumed.  The code executes as expected with no warnings.
This does not tell me why is doesn't work yet but it does tell me that the
connection to the DB is established and working as expected along with the
code minus this one parameter.

I checked all of my patch versions and PDO with mysqli shouldn't be
required until 5.5 where I'm on 5.3.  Using PDO would have me rewriting
this entire section I would think.

Thanks

Jeff
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From Agalya_Ramachandran at comcast.com  Tue Aug 30 17:42:05 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 30 Aug 2016 15:42:05 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
Message-ID: <57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>

Hi Bogdan,

In understand why these warnings are occurred.

Yes, I initially used the default script with children as 4. With this config sending 10 calls at 2cps am getting these warnings continuously.
After that I changed the children to 12. And now it's all good.

But I have a question. When I try to run the load test in opensips 1.11, I had children as 1, and I could run like 100 cps.
Why in the opensips 2.1.4 version we are getting this warnings.  What has changed between 1.11 and 2.1.4?

The only way to avoid this is to keep increasing the children?
In the new opensips in order to run 100cps I should increase my children to somewhere around 150?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 6:20 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

The message that usually pops up first :
    WARNING:core:utimer task <tm-utimer> already scheduled for....

tells that the given timer routing is slow (tm-utimer in our case) and the previous execution is not completed when the next triggering happens. This TM routine is responsible for SIP retransmissions.

The second message
    WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx us delay in execution
tells how long the timer job spent between being scheduled (triggered) and being lunched in execution (the 2 events happen in different processes).

In your case, you have 4 processes, one UDP listener and you do 2cps ? and you get one of the warnings ? do you use the default script ?

Best regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
Yes Bogdan , I do get those warnings.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 5:20 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before those warnings, have you got an warning like:
        WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled for....

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Does OpenSIPS recover from this ?
Answer: It depends upon the volume of the call. If am sending too many calls (may be more than 50 calls), OpenSIPS is not recovering. And am restarting
But if I send less than 10 calls then it is OK.

Have you seen other job reported, besides "tm-utimer" ?
Answer: I also have "blcore-expire" in logs, like once every 10 times of "tm-utimer"

I also observed that increasing the children process in config file, is solving this issue.

P.S:

e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to keep children as 12.
     For even 10 calls, if I keep children as 4, 4 to 6 calls are passing and other calls are failing.
Why do we need to increase the children process?

Just to confirm, is opensips 2.1.4 version  a stable release?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 10:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Question : after you get these logs, you mentioned the inability of handling new calls. Does OpenSIPS recover from this ? or you have to restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya






_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users



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From bogdan at opensips.org  Tue Aug 30 18:16:57 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Aug 2016 19:16:57 +0300
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
	<57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
Message-ID: <c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>

Hi Agalya,

Before answering, a fast question to you :).

What kind of processing do you in the script for the INVITEs ? is it 
something time consuming, potential blocking like DB queries, DNS, rest 
queries ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 30.08.2016 18:42, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> In understand why these warnings are occurred.
>
> Yes, I initially used the default script with children as 4. With this 
> config sending 10 calls at 2cps am getting these warnings continuously.
>
> After that I changed the children to 12. And now it?s all good.
>
> But I have a question. When I try to run the load test in opensips 
> 1.11, I had children as 1, and I could run like 100 cps.
>
> Why in the opensips 2.1.4 version we are getting this warnings.  What 
> has changed between 1.11 and 2.1.4?
>
> The only way to avoid this is to keep increasing the children?
>
> In the new opensips in order to run 100cps I should increase my 
> children to somewhere around 150?
>
> Regards,
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Tuesday, August 30, 2016 6:20 AM
> *To:* Ramachandran, Agalya (Contractor) 
> <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list 
> <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
> Hi Agalya,
>
> The message that usually pops up first :
>     WARNING:core:utimer task <tm-utimer> already scheduled for....
>
> tells that the given timer routing is slow (tm-utimer in our case) and 
> the previous execution is not completed when the next triggering 
> happens. This TM routine is responsible for SIP retransmissions.
>
> The second message
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx 
> us delay in execution
> tells how long the timer job spent between being scheduled (triggered) 
> and being lunched in execution (the 2 events happen in different 
> processes).
>
> In your case, you have 4 processes, one UDP listener and you do 2cps ? 
> and you get one of the warnings ? do you use the default script ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
>
>     Yes Bogdan , I do get those warnings.
>
>     Regards,
>     Agalya
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Thursday, August 25, 2016 5:20 PM
>     *To:* Ramachandran, Agalya (Contractor)
>     <Agalya_Ramachandran at comcast.com>
>     <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling
>     list <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>     Hi Agalya,
>
>     Before those warnings, have you got an warning like:
>             WARNING:core:utimer_ticker: utimer task <tm-utimer>
>     already scheduled for....
>
>     Regards,
>
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
>
>         Hi Bogdan,
>
>         *Does OpenSIPS recover from this ?*
>
>         Answer: It depends upon the volume of the call. If am sending
>         too many calls (may be more than 50 calls), OpenSIPS is not
>         recovering. And am restarting
>
>         But if I send less than 10 calls then it is OK.
>
>         *Have you seen other job reported, besides "tm-utimer" ?*
>
>         Answer: I also have ?blcore-expire? in logs, like once every
>         10 times of "tm-utimer"
>
>         I also observed that increasing the children process in config
>         file, is solving this issue.
>
>         P.S:
>
>         e.g : Sending 10 calls to opensips, at 2 calls per sec, then I
>         need to keep children as 12.
>
>              For even 10 calls, if I keep children as 4, 4 to 6 calls
>         are passing and other calls are failing.
>
>         Why do we need to increase the children process?
>
>         Just to confirm, is opensips 2.1.4 version  a stable release?
>
>         Regards,
>         Agalya
>
>         *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>         *Sent:* Thursday, August 25, 2016 10:54 AM
>         *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>; Ramachandran, Agalya
>         (Contractor) <Agalya_Ramachandran at comcast.com>
>         <mailto:Agalya_Ramachandran at comcast.com>
>         *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>         Hi Agalya,
>
>         Question : after you get these logs, you mentioned the
>         inability of handling new calls. Does OpenSIPS recover from
>         this ? or you have to restart after such an event ?
>
>         Have you seen other job reported, besides "tm-utimer" ?
>
>         Regards,
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
>
>             Hi team,
>
>             Am using opensips 2.1.4 version and trying to do load
>             test. When running the test cases, am getting too many
>             warnings in OpenSIPS log.
>
>             These warnings are coming continuously and its more than 2
>             page in log file.
>
>             Below are the warnings.
>
>             Aug 24 15:27:34 sturn-test
>             /usr/local/sbin/opensips[12015]:
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>             a 39450000 us delay in execution
>
>             Aug 24 15:27:34 sturn-test
>             /usr/local/sbin/opensips[12015]:
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>             a 39450000 us delay in execution
>
>             Aug 24 15:27:34 sturn-test
>             /usr/local/sbin/opensips[12015]:
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>             a 39450000 us delay in execution
>
>             Aug 24 15:27:34 sturn-test
>             /usr/local/sbin/opensips[12015]:
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>             a 39450000 us delay in execution
>
>             Aug 24 15:27:34 sturn-test
>             /usr/local/sbin/opensips[12015]:
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>             a 39450000 us delay in execution
>
>             Aug 24 15:27:34 sturn-test
>             /usr/local/sbin/opensips[12015]:
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>             a 39450000 us delay in execution
>
>             When I get this Warning, calls are getting failed.
>
>             Why am getting this warning and what could be done to
>             avoid it?
>
>             Regards,
>             Agalya
>
>
>
>
>
>
>             _______________________________________________
>
>             Users mailing list
>
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From Agalya_Ramachandran at comcast.com  Tue Aug 30 18:43:04 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 30 Aug 2016 16:43:04 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
	<57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
	<c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>
Message-ID: <c3ae268b6d134113843804dd38af6228@COPDCEX28.cable.comcast.com>

Hi Bogdan,

Yes, am doing rest queries using async.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 12:17 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before answering, a fast question to you :).

What kind of processing do you in the script for the INVITEs ? is it something time consuming, potential blocking like DB queries, DNS, rest queries ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 30.08.2016 18:42, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

In understand why these warnings are occurred.

Yes, I initially used the default script with children as 4. With this config sending 10 calls at 2cps am getting these warnings continuously.
After that I changed the children to 12. And now it's all good.

But I have a question. When I try to run the load test in opensips 1.11, I had children as 1, and I could run like 100 cps.
Why in the opensips 2.1.4 version we are getting this warnings.  What has changed between 1.11 and 2.1.4?

The only way to avoid this is to keep increasing the children?
In the new opensips in order to run 100cps I should increase my children to somewhere around 150?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 6:20 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

The message that usually pops up first :
    WARNING:core:utimer task <tm-utimer> already scheduled for....

tells that the given timer routing is slow (tm-utimer in our case) and the previous execution is not completed when the next triggering happens. This TM routine is responsible for SIP retransmissions.

The second message
    WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx us delay in execution
tells how long the timer job spent between being scheduled (triggered) and being lunched in execution (the 2 events happen in different processes).

In your case, you have 4 processes, one UDP listener and you do 2cps ? and you get one of the warnings ? do you use the default script ?

Best regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
Yes Bogdan , I do get those warnings.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 5:20 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before those warnings, have you got an warning like:
        WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled for....

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Does OpenSIPS recover from this ?
Answer: It depends upon the volume of the call. If am sending too many calls (may be more than 50 calls), OpenSIPS is not recovering. And am restarting
But if I send less than 10 calls then it is OK.

Have you seen other job reported, besides "tm-utimer" ?
Answer: I also have "blcore-expire" in logs, like once every 10 times of "tm-utimer"

I also observed that increasing the children process in config file, is solving this issue.

P.S:

e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to keep children as 12.
     For even 10 calls, if I keep children as 4, 4 to 6 calls are passing and other calls are failing.
Why do we need to increase the children process?

Just to confirm, is opensips 2.1.4 version  a stable release?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 10:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Question : after you get these logs, you mentioned the inability of handling new calls. Does OpenSIPS recover from this ? or you have to restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya







_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users




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From denis7979 at mail.ru  Tue Aug 30 21:26:48 2016
From: denis7979 at mail.ru (Denis)
Date: Tue, 30 Aug 2016 22:26:48 +0300
Subject: [OpenSIPS-Users] Opensips 2.2.1 dr_rules is empty
In-Reply-To: <1788533534.20160819175110@ptl.ru>
References: <1788533534.20160819175110@ptl.ru>
Message-ID: <286472539.20160830222648@ptl.ru>

Hello!

Some time ago i wrote about a problem when Opensips began reject all calls with 404 code.
Unfortunately i did not received any answer.
Since that time i have made upgrade from 2.1.2 to 2.2.1 and today i have got the same problem. Opensips began rejects all calls with 404 code. 

Based on the information from log, i make suggest that problem began when dr_reload command has been received. 
And the problem has been solved, again, when the dr_reload command has been received (the period about two commands is about a couple of minutes).

I please help to solve the problem. 

Thank you.

mailto:denis7979 at mail.ru


Hello!

Recently i begun to get such problem.
Opensips rejects all calls with 404 code. I can solve the problem by making dr_reload.
I tried to analyze a log and what i could found you can see in attachment.
As i could understand last problem began after dr_reload.

Thank you for any help.


mailto:denis7979 at mail.ru
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From pat at voxtelesys.com  Tue Aug 30 22:40:42 2016
From: pat at voxtelesys.com (Pat Burke)
Date: Tue, 30 Aug 2016 15:40:42 -0500
Subject: [OpenSIPS-Users] db_extra_bye values after loose_route
Message-ID: <68356998ddf8b58dccb8ea11a5e2e756@voxtelesys.com>

Hello,

I am trying to handle a case where the same call can be routed through the system (inbound call that is transferred to another phone number). ?I am trying to add values to the cdr record in the db_extra_bye values. ?Everything works if I populate the values prior to calling loose_route. ?But that doesn't work for me because am trying to use a dialog variable ($dlg_val(xxx)) to store/retrieve information in cache to calculate the duration of the call - in order to get the dialog values I have to call loose_route. ?The issue comes in because I the call can go through the system multiple times (up to twice) so I can't use the call id ($ci) to identify the call (both instances of the call have the same call id).


Any suggestions?

Regards,
Pat Burke




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From bogdan at opensips.org  Tue Aug 30 22:44:32 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Aug 2016 23:44:32 +0300
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <c3ae268b6d134113843804dd38af6228@COPDCEX28.cable.comcast.com>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
	<57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
	<c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>
	<c3ae268b6d134113843804dd38af6228@COPDCEX28.cable.comcast.com>
Message-ID: <9a23e9e3-d4be-ada4-55a8-1591adeecf11@opensips.org>

OK, and if you stop the traffic, dothe warnings go away ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 30.08.2016 19:43, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> Yes, am doing rest queries using async.
>
> Regards,
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Tuesday, August 30, 2016 12:17 PM
> *To:* Ramachandran, Agalya (Contractor) 
> <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list 
> <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
> Hi Agalya,
>
> Before answering, a fast question to you :).
>
> What kind of processing do you in the script for the INVITEs ? is it 
> something time consuming, potential blocking like DB queries, DNS, 
> rest queries ?
>
> Regards,
>
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 30.08.2016 18:42, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi Bogdan,
>
>     In understand why these warnings are occurred.
>
>     Yes, I initially used the default script with children as 4. With
>     this config sending 10 calls at 2cps am getting these warnings
>     continuously.
>
>     After that I changed the children to 12. And now it?s all good.
>
>     But I have a question. When I try to run the load test in opensips
>     1.11, I had children as 1, and I could run like 100 cps.
>
>     Why in the opensips 2.1.4 version we are getting this warnings.
>      What has changed between 1.11 and 2.1.4?
>
>     The only way to avoid this is to keep increasing the children?
>
>     In the new opensips in order to run 100cps I should increase my
>     children to somewhere around 150?
>
>     Regards,
>     Agalya
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Tuesday, August 30, 2016 6:20 AM
>     *To:* Ramachandran, Agalya (Contractor)
>     <Agalya_Ramachandran at comcast.com>
>     <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling
>     list <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>     Hi Agalya,
>
>     The message that usually pops up first :
>     WARNING:core:utimer task <tm-utimer> already scheduled for....
>
>     tells that the given timer routing is slow (tm-utimer in our case)
>     and the previous execution is not completed when the next
>     triggering happens. This TM routine is responsible for SIP
>     retransmissions.
>
>     The second message
>     WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx
>     us delay in execution
>     tells how long the timer job spent between being scheduled
>     (triggered) and being lunched in execution (the 2 events happen in
>     different processes).
>
>     In your case, you have 4 processes, one UDP listener and you do
>     2cps ? and you get one of the warnings ? do you use the default
>     script ?
>
>     Best regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
>
>         Yes Bogdan , I do get those warnings.
>
>         Regards,
>         Agalya
>
>         *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>         *Sent:* Thursday, August 25, 2016 5:20 PM
>         *To:* Ramachandran, Agalya (Contractor)
>         <Agalya_Ramachandran at comcast.com>
>         <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users
>         mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>
>         *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>         Hi Agalya,
>
>         Before those warnings, have you got an warning like:
>         WARNING:core:utimer_ticker: utimer task <tm-utimer> already
>         scheduled for....
>
>         Regards,
>
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
>
>             Hi Bogdan,
>
>             *Does OpenSIPS recover from this ?*
>
>             Answer: It depends upon the volume of the call. If am
>             sending too many calls (may be more than 50 calls),
>             OpenSIPS is not recovering. And am restarting
>
>             But if I send less than 10 calls then it is OK.
>
>             *Have you seen other job reported, besides "tm-utimer" ?*
>
>             Answer: I also have ?blcore-expire? in logs, like once
>             every 10 times of "tm-utimer"
>
>             I also observed that increasing the children process in
>             config file, is solving this issue.
>
>             P.S:
>
>             e.g : Sending 10 calls to opensips, at 2 calls per sec,
>             then I need to keep children as 12.
>
>                  For even 10 calls, if I keep children as 4, 4 to 6
>             calls are passing and other calls are failing.
>
>             Why do we need to increase the children process?
>
>             Just to confirm, is opensips 2.1.4 version  a stable release?
>
>             Regards,
>             Agalya
>
>             *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>             *Sent:* Thursday, August 25, 2016 10:54 AM
>             *To:* OpenSIPS users mailling list
>             <users at lists.opensips.org>
>             <mailto:users at lists.opensips.org>; Ramachandran, Agalya
>             (Contractor) <Agalya_Ramachandran at comcast.com>
>             <mailto:Agalya_Ramachandran at comcast.com>
>             *Subject:* Re: [OpenSIPS-Users] Too many warnings in
>             opensips log
>
>             Hi Agalya,
>
>             Question : after you get these logs, you mentioned the
>             inability of handling new calls. Does OpenSIPS recover
>             from this ? or you have to restart after such an event ?
>
>             Have you seen other job reported, besides "tm-utimer" ?
>
>             Regards,
>
>
>
>
>             Bogdan-Andrei Iancu
>
>             OpenSIPS Founder and Developer
>
>             http://www.opensips-solutions.com
>
>             On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
>
>                 Hi team,
>
>                 Am using opensips 2.1.4 version and trying to do load
>                 test. When running the test cases, am getting too many
>                 warnings in OpenSIPS log.
>
>                 These warnings are coming continuously and its more
>                 than 2 page in log file.
>
>                 Below are the warnings.
>
>                 Aug 24 15:27:34 sturn-test
>                 /usr/local/sbin/opensips[12015]:
>                 WARNING:core:handle_timer_job: utimer job <tm-utimer>
>                 has a 39450000 us delay in execution
>
>                 Aug 24 15:27:34 sturn-test
>                 /usr/local/sbin/opensips[12015]:
>                 WARNING:core:handle_timer_job: utimer job <tm-utimer>
>                 has a 39450000 us delay in execution
>
>                 Aug 24 15:27:34 sturn-test
>                 /usr/local/sbin/opensips[12015]:
>                 WARNING:core:handle_timer_job: utimer job <tm-utimer>
>                 has a 39450000 us delay in execution
>
>                 Aug 24 15:27:34 sturn-test
>                 /usr/local/sbin/opensips[12015]:
>                 WARNING:core:handle_timer_job: utimer job <tm-utimer>
>                 has a 39450000 us delay in execution
>
>                 Aug 24 15:27:34 sturn-test
>                 /usr/local/sbin/opensips[12015]:
>                 WARNING:core:handle_timer_job: utimer job <tm-utimer>
>                 has a 39450000 us delay in execution
>
>                 Aug 24 15:27:34 sturn-test
>                 /usr/local/sbin/opensips[12015]:
>                 WARNING:core:handle_timer_job: utimer job <tm-utimer>
>                 has a 39450000 us delay in execution
>
>                 When I get this Warning, calls are getting failed.
>
>                 Why am getting this warning and what could be done to
>                 avoid it?
>
>                 Regards,
>                 Agalya
>
>
>
>
>
>
>
>                 _______________________________________________
>
>                 Users mailing list
>
>                 Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From Agalya_Ramachandran at comcast.com  Tue Aug 30 22:58:16 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 30 Aug 2016 20:58:16 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <9a23e9e3-d4be-ada4-55a8-1591adeecf11@opensips.org>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
	<57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
	<c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>
	<c3ae268b6d134113843804dd38af6228@COPDCEX28.cable.comcast.com>
	<9a23e9e3-d4be-ada4-55a8-1591adeecf11@opensips.org>
Message-ID: <047f225df42746be97512a193caead3e@COPDCEX28.cable.comcast.com>

Yes. You are right.

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 4:45 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

OK, and if you stop the traffic, do the warnings go away ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 30.08.2016 19:43, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Yes, am doing rest queries using async.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 12:17 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before answering, a fast question to you :).

What kind of processing do you in the script for the INVITEs ? is it something time consuming, potential blocking like DB queries, DNS, rest queries ?

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 30.08.2016 18:42, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

In understand why these warnings are occurred.

Yes, I initially used the default script with children as 4. With this config sending 10 calls at 2cps am getting these warnings continuously.
After that I changed the children to 12. And now it's all good.

But I have a question. When I try to run the load test in opensips 1.11, I had children as 1, and I could run like 100 cps.
Why in the opensips 2.1.4 version we are getting this warnings.  What has changed between 1.11 and 2.1.4?

The only way to avoid this is to keep increasing the children?
In the new opensips in order to run 100cps I should increase my children to somewhere around 150?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 6:20 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

The message that usually pops up first :
    WARNING:core:utimer task <tm-utimer> already scheduled for....

tells that the given timer routing is slow (tm-utimer in our case) and the previous execution is not completed when the next triggering happens. This TM routine is responsible for SIP retransmissions.

The second message
    WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx us delay in execution
tells how long the timer job spent between being scheduled (triggered) and being lunched in execution (the 2 events happen in different processes).

In your case, you have 4 processes, one UDP listener and you do 2cps ? and you get one of the warnings ? do you use the default script ?

Best regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
Yes Bogdan , I do get those warnings.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 5:20 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before those warnings, have you got an warning like:
        WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled for....

Regards,






Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Does OpenSIPS recover from this ?
Answer: It depends upon the volume of the call. If am sending too many calls (may be more than 50 calls), OpenSIPS is not recovering. And am restarting
But if I send less than 10 calls then it is OK.

Have you seen other job reported, besides "tm-utimer" ?
Answer: I also have "blcore-expire" in logs, like once every 10 times of "tm-utimer"

I also observed that increasing the children process in config file, is solving this issue.

P.S:

e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to keep children as 12.
     For even 10 calls, if I keep children as 4, 4 to 6 calls are passing and other calls are failing.
Why do we need to increase the children process?

Just to confirm, is opensips 2.1.4 version  a stable release?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 10:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Question : after you get these logs, you mentioned the inability of handling new calls. Does OpenSIPS recover from this ? or you have to restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,






Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya








_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users





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From bogdan at opensips.org  Tue Aug 30 23:40:53 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 31 Aug 2016 00:40:53 +0300
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <047f225df42746be97512a193caead3e@COPDCEX28.cable.comcast.com>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
	<57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
	<c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>
	<c3ae268b6d134113843804dd38af6228@COPDCEX28.cable.comcast.com>
	<9a23e9e3-d4be-ada4-55a8-1591adeecf11@opensips.org>
	<047f225df42746be97512a193caead3e@COPDCEX28.cable.comcast.com>
Message-ID: <6f1180c1-54ea-a31f-c81e-0b5aa3d9984f@opensips.org>

Is it ok to send you a testing patch for tomorrow ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 30.08.2016 23:58, Ramachandran, Agalya (Contractor) wrote:
>
> Yes. You are right.
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Tuesday, August 30, 2016 4:45 PM
> *To:* Ramachandran, Agalya (Contractor) 
> <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list 
> <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
> OK, and if you stop the traffic, do the warnings go away ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 30.08.2016 19:43, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi Bogdan,
>
>     Yes, am doing rest queries using async.
>
>     Regards,
>     Agalya
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Tuesday, August 30, 2016 12:17 PM
>     *To:* Ramachandran, Agalya (Contractor)
>     <Agalya_Ramachandran at comcast.com>
>     <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling
>     list <users at lists.opensips.org> <mailto:users at lists.opensips.org>
>     *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>     Hi Agalya,
>
>     Before answering, a fast question to you :).
>
>     What kind of processing do you in the script for the INVITEs ? is
>     it something time consuming, potential blocking like DB queries,
>     DNS, rest queries ?
>
>     Regards,
>
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 30.08.2016 18:42, Ramachandran, Agalya (Contractor) wrote:
>
>         Hi Bogdan,
>
>         In understand why these warnings are occurred.
>
>         Yes, I initially used the default script with children as 4.
>         With this config sending 10 calls at 2cps am getting these
>         warnings continuously.
>
>         After that I changed the children to 12. And now it?s all good.
>
>         But I have a question. When I try to run the load test in
>         opensips 1.11, I had children as 1, and I could run like 100 cps.
>
>         Why in the opensips 2.1.4 version we are getting this
>         warnings.  What has changed between 1.11 and 2.1.4?
>
>         The only way to avoid this is to keep increasing the children?
>
>         In the new opensips in order to run 100cps I should increase
>         my children to somewhere around 150?
>
>         Regards,
>         Agalya
>
>         *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>         *Sent:* Tuesday, August 30, 2016 6:20 AM
>         *To:* Ramachandran, Agalya (Contractor)
>         <Agalya_Ramachandran at comcast.com>
>         <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users
>         mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>
>         *Subject:* Re: [OpenSIPS-Users] Too many warnings in opensips log
>
>         Hi Agalya,
>
>         The message that usually pops up first :
>             WARNING:core:utimer task <tm-utimer> already scheduled for....
>
>         tells that the given timer routing is slow (tm-utimer in our
>         case) and the previous execution is not completed when the
>         next triggering happens. This TM routine is responsible for
>         SIP retransmissions.
>
>         The second message
>             WARNING:core:handle_timer_job: utimer job <tm-utimer> has
>         a xxxxx us delay in execution
>         tells how long the timer job spent between being scheduled
>         (triggered) and being lunched in execution (the 2 events
>         happen in different processes).
>
>         In your case, you have 4 processes, one UDP listener and you
>         do 2cps ? and you get one of the warnings ? do you use the
>         default script ?
>
>         Best regards,
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
>
>             Yes Bogdan , I do get those warnings.
>
>             Regards,
>             Agalya
>
>             *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>             *Sent:* Thursday, August 25, 2016 5:20 PM
>             *To:* Ramachandran, Agalya (Contractor)
>             <Agalya_Ramachandran at comcast.com>
>             <mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users
>             mailling list <users at lists.opensips.org>
>             <mailto:users at lists.opensips.org>
>             *Subject:* Re: [OpenSIPS-Users] Too many warnings in
>             opensips log
>
>             Hi Agalya,
>
>             Before those warnings, have you got an warning like:
>                     WARNING:core:utimer_ticker: utimer task
>             <tm-utimer> already scheduled for....
>
>             Regards,
>
>
>
>
>
>             Bogdan-Andrei Iancu
>
>             OpenSIPS Founder and Developer
>
>             http://www.opensips-solutions.com
>
>             On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
>
>                 Hi Bogdan,
>
>                 *Does OpenSIPS recover from this ?*
>
>                 Answer: It depends upon the volume of the call. If am
>                 sending too many calls (may be more than 50 calls),
>                 OpenSIPS is not recovering. And am restarting
>
>                 But if I send less than 10 calls then it is OK.
>
>                 *Have you seen other job reported, besides "tm-utimer" ?*
>
>                 Answer: I also have ?blcore-expire? in logs, like once
>                 every 10 times of "tm-utimer"
>
>                 I also observed that increasing the children process
>                 in config file, is solving this issue.
>
>                 P.S:
>
>                 e.g : Sending 10 calls to opensips, at 2 calls per
>                 sec, then I need to keep children as 12.
>
>                      For even 10 calls, if I keep children as 4, 4 to
>                 6 calls are passing and other calls are failing.
>
>                 Why do we need to increase the children process?
>
>                 Just to confirm, is opensips 2.1.4 version  a stable
>                 release?
>
>                 Regards,
>                 Agalya
>
>                 *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>                 *Sent:* Thursday, August 25, 2016 10:54 AM
>                 *To:* OpenSIPS users mailling list
>                 <users at lists.opensips.org>
>                 <mailto:users at lists.opensips.org>; Ramachandran,
>                 Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
>                 <mailto:Agalya_Ramachandran at comcast.com>
>                 *Subject:* Re: [OpenSIPS-Users] Too many warnings in
>                 opensips log
>
>                 Hi Agalya,
>
>                 Question : after you get these logs, you mentioned the
>                 inability of handling new calls. Does OpenSIPS recover
>                 from this ? or you have to restart after such an event ?
>
>                 Have you seen other job reported, besides "tm-utimer" ?
>
>                 Regards,
>
>
>
>
>
>                 Bogdan-Andrei Iancu
>
>                 OpenSIPS Founder and Developer
>
>                 http://www.opensips-solutions.com
>
>                 On 24.08.2016 18:31, Ramachandran, Agalya (Contractor)
>                 wrote:
>
>                     Hi team,
>
>                     Am using opensips 2.1.4 version and trying to do
>                     load test. When running the test cases, am getting
>                     too many warnings in OpenSIPS log.
>
>                     These warnings are coming continuously and its
>                     more than 2 page in log file.
>
>                     Below are the warnings.
>
>                     Aug 24 15:27:34 sturn-test
>                     /usr/local/sbin/opensips[12015]:
>                     WARNING:core:handle_timer_job: utimer job
>                     <tm-utimer> has a 39450000 us delay in execution
>
>                     Aug 24 15:27:34 sturn-test
>                     /usr/local/sbin/opensips[12015]:
>                     WARNING:core:handle_timer_job: utimer job
>                     <tm-utimer> has a 39450000 us delay in execution
>
>                     Aug 24 15:27:34 sturn-test
>                     /usr/local/sbin/opensips[12015]:
>                     WARNING:core:handle_timer_job: utimer job
>                     <tm-utimer> has a 39450000 us delay in execution
>
>                     Aug 24 15:27:34 sturn-test
>                     /usr/local/sbin/opensips[12015]:
>                     WARNING:core:handle_timer_job: utimer job
>                     <tm-utimer> has a 39450000 us delay in execution
>
>                     Aug 24 15:27:34 sturn-test
>                     /usr/local/sbin/opensips[12015]:
>                     WARNING:core:handle_timer_job: utimer job
>                     <tm-utimer> has a 39450000 us delay in execution
>
>                     Aug 24 15:27:34 sturn-test
>                     /usr/local/sbin/opensips[12015]:
>                     WARNING:core:handle_timer_job: utimer job
>                     <tm-utimer> has a 39450000 us delay in execution
>
>                     When I get this Warning, calls are getting failed.
>
>                     Why am getting this warning and what could be done
>                     to avoid it?
>
>                     Regards,
>                     Agalya
>
>
>
>
>
>
>
>
>                     _______________________________________________
>
>                     Users mailing list
>
>                     Users at lists.opensips.org
>                     <mailto:Users at lists.opensips.org>
>
>                     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From razvan at opensips.org  Wed Aug 31 10:24:33 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 31 Aug 2016 11:24:33 +0300
Subject: [OpenSIPS-Users] db_extra_bye values after loose_route
In-Reply-To: <68356998ddf8b58dccb8ea11a5e2e756@voxtelesys.com>
References: <68356998ddf8b58dccb8ea11a5e2e756@voxtelesys.com>
Message-ID: <1df225aa-8bb5-812c-a5a2-2f24e9d158fb@opensips.org>

Hi, Pat!

Unfortunately there is no other way to do what you need. The idea is 
that the accounting is done when you call loose_route(), and before that 
you don't have the dialog - therefore you don't have the dlg_vals.

The only solution I can think of is to use local_cache, and find a way 
to uniqly store the keys per dialog so that they don't overlap when you 
have multiple legs.

As good news, this has been sorted out in the master branch, and will be 
available starting from OpenSIPS 2.3 release.

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/30/2016 11:40 PM, Pat Burke wrote:
> Hello,
>
> I am trying to handle a case where the same call can be routed through 
> the system (inbound call that is transferred to another phone number). 
>  I am trying to add values to the cdr record in the db_extra_bye 
> values.  Everything works if I populate the values prior to calling 
> loose_route.  But that doesn't work for me because am trying to use a 
> dialog variable ($dlg_val(xxx)) to store/retrieve information in cache 
> to calculate the duration of the call - in order to get the dialog 
> values I have to call loose_route.  The issue comes in because I the 
> call can go through the system multiple times (up to twice) so I can't 
> use the call id ($ci) to identify the call (both instances of the call 
> have the same call id).
>
> Any suggestions?
>
> Regards,
> *Pat Burke*
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From hyavari at rocketmail.com  Wed Aug 31 10:52:31 2016
From: hyavari at rocketmail.com (=?UTF-8?Q?=E2=80=AAH_Yavari=E2=80=AC_=E2=80=AA?=)
Date: Wed, 31 Aug 2016 08:52:31 +0000 (UTC)
Subject: [OpenSIPS-Users] opensips and mariadb
References: <1099241853.4177712.1472633551762.ref@mail.yahoo.com>
Message-ID: <1099241853.4177712.1472633551762@mail.yahoo.com>

Hi to all,
I'm compiling opensips 2.2.1 on CentOS 7 + MariaDB, But I can't compile db_mysql module. OpenSIPS and Mariadb are compatible with each other? 
I can use MariaDB as OpenSIPS db engine?

Regards,H.Yavari
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From Agalya_Ramachandran at comcast.com  Wed Aug 31 15:29:06 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 31 Aug 2016 13:29:06 +0000
Subject: [OpenSIPS-Users] opensips and mariadb
In-Reply-To: <1099241853.4177712.1472633551762@mail.yahoo.com>
References: <1099241853.4177712.1472633551762.ref@mail.yahoo.com>
	<1099241853.4177712.1472633551762@mail.yahoo.com>
Message-ID: <4c62b793bf6341f79ea5fa611bf41659@COPDCEX28.cable.comcast.com>

Hi Yavari,

Yes, it is compatible. Make sure you install proper my-sql dependencies.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of ?H Yavari? ??
Sent: Wednesday, August 31, 2016 4:53 AM
To: OpenSIPS Users Mailling List <users at lists.opensips.org>
Subject: [OpenSIPS-Users] opensips and mariadb

Hi to all,

I'm compiling opensips 2.2.1 on CentOS 7 + MariaDB, But I can't compile db_mysql module. OpenSIPS and Mariadb are compatible with each other?
I can use MariaDB as OpenSIPS db engine?


Regards,
H.Yavari
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From Agalya_Ramachandran at comcast.com  Wed Aug 31 16:45:21 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 31 Aug 2016 14:45:21 +0000
Subject: [OpenSIPS-Users] Too many warnings in opensips log
In-Reply-To: <6f1180c1-54ea-a31f-c81e-0b5aa3d9984f@opensips.org>
References: <84a13bc271f146d5bdd2dd0fc2f2494a@COPDCEX28.cable.comcast.com>
	<f7d6c243-9d19-22ae-dce9-b5ab1641fb52@opensips.org>
	<778991147c214d19bf63d58656cbae6b@COPDCEX28.cable.comcast.com>
	<b48e1bb4-cee6-5349-25a5-9a71720a089a@opensips.org>
	<da8809bd24084d75a2ca0c05569310a5@COPDCEX28.cable.comcast.com>
	<7488dbd9-589f-6cba-ed62-264e23aefb6e@opensips.org>
	<57d53198132a4163b637e596a2c89c85@COPDCEX28.cable.comcast.com>
	<c8124e4d-ae65-fbbc-59eb-552f377b425d@opensips.org>
	<c3ae268b6d134113843804dd38af6228@COPDCEX28.cable.comcast.com>
	<9a23e9e3-d4be-ada4-55a8-1591adeecf11@opensips.org>
	<047f225df42746be97512a193caead3e@COPDCEX28.cable.comcast.com>
	<6f1180c1-54ea-a31f-c81e-0b5aa3d9984f@opensips.org>
Message-ID: <4c21dfaf87b3465c9635f5ae38221012@COPDCEX28.cable.comcast.com>

Hi Bogdan,

Sure you can send it by tomorrow.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 5:41 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Is it ok to send you a testing patch for tomorrow ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 30.08.2016 23:58, Ramachandran, Agalya (Contractor) wrote:
Yes. You are right.

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 4:45 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

OK, and if you stop the traffic, do the warnings go away ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 30.08.2016 19:43, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Yes, am doing rest queries using async.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 12:17 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before answering, a fast question to you :).

What kind of processing do you in the script for the INVITEs ? is it something time consuming, potential blocking like DB queries, DNS, rest queries ?

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 30.08.2016 18:42, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

In understand why these warnings are occurred.

Yes, I initially used the default script with children as 4. With this config sending 10 calls at 2cps am getting these warnings continuously.
After that I changed the children to 12. And now it's all good.

But I have a question. When I try to run the load test in opensips 1.11, I had children as 1, and I could run like 100 cps.
Why in the opensips 2.1.4 version we are getting this warnings.  What has changed between 1.11 and 2.1.4?

The only way to avoid this is to keep increasing the children?
In the new opensips in order to run 100cps I should increase my children to somewhere around 150?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Tuesday, August 30, 2016 6:20 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

The message that usually pops up first :
    WARNING:core:utimer task <tm-utimer> already scheduled for....

tells that the given timer routing is slow (tm-utimer in our case) and the previous execution is not completed when the next triggering happens. This TM routine is responsible for SIP retransmissions.

The second message
    WARNING:core:handle_timer_job: utimer job <tm-utimer> has a xxxxx us delay in execution
tells how long the timer job spent between being scheduled (triggered) and being lunched in execution (the 2 events happen in different processes).

In your case, you have 4 processes, one UDP listener and you do 2cps ? and you get one of the warnings ? do you use the default script ?

Best regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.08.2016 16:17, Ramachandran, Agalya (Contractor) wrote:
Yes Bogdan , I do get those warnings.

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 5:20 PM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Before those warnings, have you got an warning like:
        WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled for....

Regards,







Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.08.2016 18:08, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Does OpenSIPS recover from this ?
Answer: It depends upon the volume of the call. If am sending too many calls (may be more than 50 calls), OpenSIPS is not recovering. And am restarting
But if I send less than 10 calls then it is OK.

Have you seen other job reported, besides "tm-utimer" ?
Answer: I also have "blcore-expire" in logs, like once every 10 times of "tm-utimer"

I also observed that increasing the children process in config file, is solving this issue.

P.S:

e.g : Sending 10 calls to opensips, at 2 calls per sec, then I need to keep children as 12.
     For even 10 calls, if I keep children as 4, 4 to 6 calls are passing and other calls are failing.
Why do we need to increase the children process?

Just to confirm, is opensips 2.1.4 version  a stable release?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, August 25, 2016 10:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Too many warnings in opensips log

Hi Agalya,

Question : after you get these logs, you mentioned the inability of handling new calls. Does OpenSIPS recover from this ? or you have to restart after such an event ?

Have you seen other job reported, besides "tm-utimer" ?

Regards,







Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 24.08.2016 18:31, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Am using opensips 2.1.4 version and trying to do load test. When running the test cases, am getting too many warnings in OpenSIPS log.
These warnings are coming continuously and its more than 2 page in log file.
Below are the warnings.

Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution
Aug 24 15:27:34 sturn-test /usr/local/sbin/opensips[12015]: WARNING:core:handle_timer_job: utimer job <tm-utimer> has a 39450000 us delay in execution

When I get this Warning, calls are getting failed.
Why am getting this warning and what could be done to avoid it?

Regards,
Agalya









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From jim at devito.cc  Wed Aug 31 20:54:26 2016
From: jim at devito.cc (Jim DeVito)
Date: Wed, 31 Aug 2016 11:54:26 -0700
Subject: [OpenSIPS-Users] Possible Bug / Unexpected behavior with ACC.
Message-ID: <baf8aef2347d890a452a1892d494a172@mail.devito.cc>

Hi All,

Strange situation. I'm using the ACC module with Flatstore and it was 
working fine until the update to 2.2.1 a couple days ago. It seems the 
variable $rU is populated with hex for all 0's in very specific 
situations. Namely receiving a 503 back from the carrier. This is 
causing it to get put into the ACC Flat store file. See below for the 
specifics and relevant configuration info. I do have a dev system I can 
reproduce the problem on if there is some specific debugging you 
suggest.

ACC Flatstore file line (the 
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00 is the bad 
part. It should be 011610408287261)

INVITE|TM1AqTt8A57InEKZ75B293||20160831182524021875-339cad6d567da38f5d19c82b315b47ef|503|Service 
Unavailable|1472667931|\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00|115|sip-proxy02|from-c5-ns|||||

modparam("acc", "db_url","flatstore:/var/log/opensips/acc")
modparam("acc", "db_extra", "to_tn=$rU; ....etc....)

$rU is the extra parameter that is supposed to be 011610408287261 but it 
comes out as 
\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00\x00

Thanks!!
-- 
Jim DeVito