From spanda at 3clogic.com  Tue Oct  1 02:29:36 2019
From: spanda at 3clogic.com (Sasmita Panda)
Date: Tue, 1 Oct 2019 11:59:36 +0530
Subject: [OpenSIPS-Users] Need some clarification in rtpengine module .
In-Reply-To: <CALXCt0p7HOuF5NHdF4H-iVXt2FQt9N0tTXBkBX180DBtsVQrTQ@mail.gmail.com>
References: <CALf-Q44yD9frWiVk7jvW9uXhYHj73kbsdNa+ba4yvfkhnguz2w@mail.gmail.com>
 <53569728-e877-e5dc-84fd-2a0e929edbd6@opensips.org>
 <CALf-Q45HmkuFQrvndAd40inbGiwPmELbjpYBQka7PWcTOELmiA@mail.gmail.com>
 <CAFjCFzk9ROq3SMc3j9HY_XADHEctO4DMhq9TyxXO69SoTH_06w@mail.gmail.com>
 <CALf-Q47N0BWgMO3EayYJc5KMCMi=kVCGgk3=4umEdqoxTYgopQ@mail.gmail.com>
 <CALXCt0qOQS-Uck8DpUSvRB68HajjmMo+2aUT4EYxumSvP1u8ew@mail.gmail.com>
 <CALf-Q44pDWuRMbUUgOm3L6760-wgtMrmOUa1V7yyqLvmcagFfQ@mail.gmail.com>
 <CALf-Q47pSLO-wJM2XQv=nRw=nVyNepCXZbEqQvyZtpPEBP1vUw@mail.gmail.com>
 <CALXCt0p7HOuF5NHdF4H-iVXt2FQt9N0tTXBkBX180DBtsVQrTQ@mail.gmail.com>
Message-ID: <CALf-Q44Q8hhfNDD5CZ=Jtg+9XcPv5ZBdZm1m1wqXJTn2yCCwVg@mail.gmail.com>

Thank you Giovanni . Its  solved my problem .


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


On Mon, Sep 30, 2019 at 6:57 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> using the "loop-protect" flag, that is documented in opensips module too (
> https://opensips.org/html/docs/modules/3.0.x/rtpengine.html ), is
> supported since february 2018 in rtpengine
>
>
> https://github.com/sipwise/rtpengine/commit/1477f9796f868857317b25fe4ddc56c1e44698af
>
> On Mon, Sep 30, 2019 at 12:50 PM Sasmita Panda <spanda at 3clogic.com> wrote:
>
>> I am using below revision of rtpengine .
>> git-master-3102357
>>
>> I have to update the rtpengine code to use this flag ?
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>>
>> On Mon, Sep 30, 2019 at 4:15 PM Sasmita Panda <spanda at 3clogic.com> wrote:
>>
>>> Hi ,
>>>
>>> I have gone through the document , but not done with the changes yet .
>>>
>>> I have tried to run rtpengine with the flag "loop-protect" , according
>>> to the explanation , rtpengine has to insert the attribute
>>> "a=rtpengine:..." in the SDP . But this is not happening with me .
>>>
>>> Now I am suspecting , did you mean I have ti insert that attribute
>>> through opensips in the SDP body while sending command to the rtpenigne ?
>>> May be I am asking a stupid question but I am stuck here .
>>> Nothing working for me . Please do help .
>>>
>>>
>>>
>>> *Thanks & Regards*
>>> *Sasmita Panda*
>>> *Senior Network Testing and Software Engineer*
>>> *3CLogic , ph:07827611765*
>>>
>>>
>>> On Fri, Sep 27, 2019 at 5:05 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> there is documentation available:
>>>>
>>>> https://github.com/sipwise/rtpengine
>>>>
>>>> "
>>>> loop protect
>>>>
>>>> Inserts a custom attribute (a=rtpengine:...) into the outgoing SDP to
>>>> prevent *rtpengine* processing and rewriting the same SDP multiple
>>>> times. This is useful if your setup involves signalling loops and need to
>>>> make sure that *rtpengine* doesn't start looping media packets back to
>>>> itself. When this flag is present and *rtpengine* sees a matching
>>>> attribute already present in the SDP, it will leave the SDP untouched and
>>>> not process the message.
>>>> "
>>>>
>>>> On Fri, Sep 27, 2019 at 1:12 PM Sasmita Panda <spanda at 3clogic.com>
>>>> wrote:
>>>>
>>>>> I am actually not getting how to do this .
>>>>>
>>>>> Somehow add_body_part changing the Content-Type header .
>>>>>
>>>>> Before adding this function :
>>>>> Content-Type: application/sdp
>>>>>
>>>>> After adding this function .
>>>>> Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42
>>>>>
>>>>> --OSS-unique-boundary-42
>>>>> Content-Type: application/sdp
>>>>>
>>>>> This is get added .
>>>>>
>>>>> Whats the use of this . I think rather than this , I can add any
>>>>> custom header and compare that custom header before sending command to
>>>>> rtpengine . Will that work ?
>>>>>
>>>>>
>>>>> *Thanks & Regards*
>>>>> *Sasmita Panda*
>>>>> *Senior Network Testing and Software Engineer*
>>>>> *3CLogic , ph:07827611765*
>>>>>
>>>>>
>>>>> On Fri, Sep 27, 2019 at 1:26 PM Callum Guy <callum.guy at x-on.co.uk>
>>>>> wrote:
>>>>>
>>>>>> I think your syntax is wrong?
>>>>>>
>>>>>>
>>>>>> https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#func_add_body_part
>>>>>>
>>>>>>
>>>>>> On Fri, 27 Sep 2019 at 08:36, Sasmita Panda <spanda at 3clogic.com>
>>>>>> wrote:
>>>>>>
>>>>>>> How I will add a attribute in the SDP body . In the rtpengine module
>>>>>>> there is no flag which will add a attribute directly .
>>>>>>>
>>>>>>> If I am doing like below
>>>>>>>
>>>>>>> add_body_part("$var(body)", "a=sdpmangled:yes\r\n");
>>>>>>>
>>>>>>> Its not adding this attribute in the body . Is there any other option to do this ?
>>>>>>>
>>>>>>> *Thanks & Regards*
>>>>>>> *Sasmita Panda*
>>>>>>> *Senior Network Testing and Software Engineer*
>>>>>>> *3CLogic , ph:07827611765*
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Sep 26, 2019 at 7:37 PM Bogdan-Andrei Iancu <
>>>>>>> bogdan at opensips.org> wrote:
>>>>>>>
>>>>>>>> Hi Sasmita,
>>>>>>>>
>>>>>>>> Not sure if rtpengine has a similar built-in feature (their doc is
>>>>>>>> something that can be improved :P), but you can achieve the same from
>>>>>>>> script level I guess (adding and testing that SDP 'a' line).
>>>>>>>>
>>>>>>>> Best regards,
>>>>>>>>
>>>>>>>> Bogdan-Andrei Iancu
>>>>>>>>
>>>>>>>> OpenSIPS Founder and Developer
>>>>>>>>   https://www.opensips-solutions.com
>>>>>>>> OpenSIPS Summit 2019
>>>>>>>>   https://www.opensips.org/events/Summit-2019Amsterdam/
>>>>>>>>
>>>>>>>> On 9/24/19 9:42 AM, Sasmita Panda wrote:
>>>>>>>>
>>>>>>>> Hi All,
>>>>>>>>
>>>>>>>> rtpengine and rtpproxy module somehow plays same role with some
>>>>>>>> advance features in rtpengine .
>>>>>>>>
>>>>>>>> There is a parameter in rtpproxy module " nortpproxy_str " through
>>>>>>>> which we can manage the involvement  of rtpproxy in the call leg .
>>>>>>>>
>>>>>>>> modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
>>>>>>>>
>>>>>>>> In rtpengine there is no such parameter having similar functionality .
>>>>>>>> Is there any other way which will behave similarly in rtpengine .
>>>>>>>>
>>>>>>>> Basically what I wanted to do is , in a call having 2 different leg
>>>>>>>> .
>>>>>>>> If in one leg rtpengine is already involved ,
>>>>>>>> I don,'t want to involved rtpengine again in the other leg .
>>>>>>>>
>>>>>>>>
>>>>>>>> *Thanks & Regards*
>>>>>>>> *Sasmita Panda*
>>>>>>>> *Senior Network Testing and Software Engineer*
>>>>>>>> *3CLogic , ph:07827611765*
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing listUsers at lists.opensips.orghttp://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
>>>>>>>
>>>>>>
>>>>>>
>>>>>> *0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   **
>>>>>> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>>>>>>   <https://twitter.com/xonuk> *
>>>>>>
>>>>>> X-on is a trading name of Storacall Technology Ltd a limited company
>>>>>> registered in England and Wales.
>>>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel
>>>>>> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
>>>>>> The information in this e-mail is confidential and for use by the
>>>>>> addressee(s) only. If you are not the intended recipient, please notify
>>>>>> X-on immediately on +44(0)333 332 0000 and delete the
>>>>>> message from your computer. If you are not a named addressee you must
>>>>>> not use, disclose, disseminate, distribute, copy, print or reply to this
>>>>>> email. Views or opinions expressed by an individual
>>>>>> within this email may not necessarily reflect the views of X-on or
>>>>>> its associated companies. Although X-on routinely screens for viruses,
>>>>>> addressees should scan this email and any attachments
>>>>>> for viruses. X-on makes no representation or warranty as to the
>>>>>> absence of viruses in this email or any attachments.
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
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From callum.guy at x-on.co.uk  Tue Oct  1 04:30:17 2019
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Tue, 1 Oct 2019 09:30:17 +0100
Subject: [OpenSIPS-Users] launch() and rest_get()
Message-ID: <CAFjCFzm9LGnK1iUzhBvP5pu5WoBHRT51rX+HHC4n1_9gEyO3Ow@mail.gmail.com>

Hi All,

I wanted to report the following behaviour in 3.0 - launch wrapping
rest_get does not continue execution:

route {
  xlog("L_INFO", "INFO: LAUNCHING.....! \n");
  rest_get("https://www.google.com", $avp(response_body));
  xlog("L_INFO", "INFO: LAUNCHED.....!!!! \n");
}

2019-10-01T09:25:51.030715+01:00 FR-P-SIPREG-1 opensips[78990]: INFO:
LAUNCHING.....!
2019-10-01T09:25:51.342875+01:00 FR-P-SIPREG-1 opensips[78990]: INFO:
LAUNCHED.....!!!!
2019-10-01T09:25:51.485666+01:00 FR-P-SIPREG-1 opensips[78992]: INFO:
LAUNCHING.....!
2019-10-01T09:25:51.806213+01:00 FR-P-SIPREG-1 opensips[78992]: INFO:
LAUNCHED.....!!!!

route {
  xlog("L_INFO", "INFO: LAUNCHING.....! \n");
  launch(rest_get("https://www.google.com", $avp(response_body)));
  xlog("L_INFO", "INFO: LAUNCHED.....!!!! \n");
}

2019-10-01T09:26:05.485315+01:00 FR-P-SIPREG-1 opensips[79100]: INFO:
LAUNCHING.....!
2019-10-01T09:26:06.726854+01:00 FR-P-SIPREG-1 opensips[79103]: INFO:
LAUNCHING.....!

# *opensips -V*
version: opensips 3.0.0 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, HP_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, sigio_rt, select.
main.c compiled on 13:02:44 May 30 2019 with gcc 4.8.5

This can be avoided simply by using async however I don't believe this to
be the correct bahviour?

Many thanks,

Callum

-- 





*0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   ** 
<https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel> 
  <https://twitter.com/xonuk> *


X-on
is a trading name of Storacall 
Technology Ltd a limited company registered in
England and Wales.


Registered Office : Avaland House, 110 London Road, Apsley, Hemel 
Hempstead,
Herts, HP3 9SD. Company Registration No. 2578478.

The 
information in this e-mail is confidential and for use by the addressee(s)

only. If you are not the intended recipient, please notify X-on immediately 
on +44(0)333 332 0000 and delete the
message from your computer. If you are 
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disclose, disseminate, distribute, 
copy, print or reply to this email. Views
or opinions expressed by an 
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From razvan at opensips.org  Tue Oct  1 05:54:38 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 1 Oct 2019 12:54:38 +0300
Subject: [OpenSIPS-Users] Opensips 3.0 load display
In-Reply-To: <050D766A-B85C-49A9-BBEF-32E9572EDD92@genesys.com>
References: <CAG2rjpty8t2x9RnTTi0+3N5LZ6+XUOuNawk7Yd_6XJTvBXyNpg@mail.gmail.com>
 <050D766A-B85C-49A9-BBEF-32E9572EDD92@genesys.com>
Message-ID: <616cc39d-4105-4c26-8bd0-28b33ddcd9cc@opensips.org>

Hi, Mehdi!

If you want to measure the Calls per second, you will have to use the 
ratelimit module[1], which is able to count the calls per second, if you 
choose a window_size & slot_period [2] values that result in 1 second 
window.

[1] https://opensips.org/html/docs/modules/3.0.x/ratelimit.html
[2] 
https://opensips.org/html/docs/modules/3.0.x/ratelimit.html#param_window_size

Best regards,
Răzvan

On 9/30/19 3:48 PM, Ben Newlin wrote:
> Shortly, yes. The load statistic does not measure calls per second. Per 
> the documentation [1], it measures the percentage of OpenSIPS processes 
> that are actively processing messages.
> 
> There is no built in statistic for calls per second that I am aware of.
> 
> [1] 
> https://www.opensips.org/Documentation/Interface-CoreStatistics-3-0#toc15
> 
> Ben Newlin
> 
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Mehdi 
> Shirazi <m.shirazi at gmail.com>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Monday, September 30, 2019 at 12:56 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *[OpenSIPS-Users] Opensips 3.0 load display
> 
> Hi
> 
> I use Opensips3.0 as simple stateless proxy.I want to see how many 
> Invites per second was processed. I used :
> (opensips-cli): mi get_statistics load:
> 
> But all results are zero.Am I doing something wrong ?
> 
> Regards
> 
> M.Shirazi
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From razvan at opensips.org  Tue Oct  1 06:04:02 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 1 Oct 2019 13:04:02 +0300
Subject: [OpenSIPS-Users] sql_cacher with cachedb_local
In-Reply-To: <CAG2rjptocVuFD7g9OZDFjcPg0iJ4vikTL9YLVp-357m_AAO=Xw@mail.gmail.com>
References: <CAG2rjps72o=413J8_DufzWGqX+G83b_ZtBOMf599HdY5VyeCNw@mail.gmail.com>
 <90521e57-94d4-4986-6732-0262e1ad2c99@opensips.org>
 <CAG2rjpsEhd07+QdSCpRd2kcywZe0B3J-UXCO2bGXYxsxfBq_vg@mail.gmail.com>
 <CAG2rjptocVuFD7g9OZDFjcPg0iJ4vikTL9YLVp-357m_AAO=Xw@mail.gmail.com>
Message-ID: <f43b8ff1-066d-1998-fcc8-80f9842151f7@opensips.org>

Hi, Mehdi!

Unfortunately there's no way you could fetch the cached values, as these 
are only loaded _after_ the startup_route is ran. So even if you go over 
MI to the MI interface, you will still not get the value.
You may try to set the sql_cacher in an on_demand mode.

Best regards,
Răzvan

On 9/21/19 4:27 PM, Mehdi Shirazi wrote:
> Hi
> Is there a way to see content of sql_cacher loaded inside of 
> cachedb_local with "mi cache_fetch" command ?
> 
> |Regards
> |
> 
> |M.Shirazi
> |
> 
> 
> On Mon, Sep 16, 2019 at 6:23 PM Mehdi Shirazi <m.shirazi at gmail.com 
> <mailto:m.shirazi at gmail.com>> wrote:
> 
>     Hi
>     Thanks for reply.
>     inside main route it was working ok but inside startup_route (2.4.6
>     and 3.0) not working.
>     Regards
>     M.Shirazi
> 
>     On Thu, Sep 12, 2019 at 8:16 PM Bogdan-Andrei Iancu
>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
> 
>         HI Mehdi,
> 
>         Do you get that error only when using it from startup_route ?
> 
>         Also, try to change the cachedb_url to "local:///collection2"
> 
>         Regards,
> 
>         Bogdan-Andrei Iancu
> 
>         OpenSIPS Founder and Developer
>            https://www.opensips-solutions.com
>         OpenSIPS Summit 2019
>            https://www.opensips.org/events/Summit-2019Amsterdam/
> 
>         On 8/30/19 10:12 AM, Mehdi Shirazi wrote:
>>         Hi
>>         Please help me configuring sql_cacher.
>>         I use Opensips 2.4.6 .
>>         modparam("cachedb_local", "cachedb_url",
>>         "local:group2:///collection2")
>>         modparam("sql_cacher", "cache_table",
>>         "id=st_caching
>>         db_url=mysql://opensips:1234ms at localhost/opensips
>>         cachedb_url=local:group2:///collection2
>>         table=STrig
>>         key=number
>>         columns=cause")
>>         startup_route {
>>             xlog("The cause is
>>         $sql_cached_value(st_caching:cause:88443322)");
>>             }
>>         in log:
>>         ERROR:sql_cacher:get_rld_vers_from_cache: Failed to get reload
>>         version integer from cachedb
>>         ERROR:sql_cacher:pv_get_sql_cached_value: Error fetching from
>>         cachedb
>>
>>         Regards
>>         M.Shirazi
>>
>>
>>         _______________________________________________
>>         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
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From fgast+opensips at only640k.net  Tue Oct  1 06:34:50 2019
From: fgast+opensips at only640k.net (Fabian Gast)
Date: Tue, 1 Oct 2019 12:34:50 +0200 (CEST)
Subject: [OpenSIPS-Users] TLS Cypher Renegotiation & WireShark or
 SIPTrace
In-Reply-To: <1569485341259-0.post@n2.nabble.com>
References: <1569485341259-0.post@n2.nabble.com>
Message-ID: <830338746.37509.1569926090806.JavaMail.zimbra@only640k.net>


----- Ursprüngliche Mail -----
> Von: "JamesH" <james at ip-sentinel.com>

> 
> Alternatively without having to build a HOMER server how do I dump all sip
> traffic that OpenSIPS sends to & fro? I'm currently trying to use
> sip_trace("$var(trace_id)", "d"); but with no output. I've also gone for
> log level 4 but that doesn't get everything and Debug mode is segfaulting on
> OpenSSL versions and I don't want the pain of rebuilding the thing.
> 

There are some solutions for this, i use something like 

loadmodule "proto_hep.so"
loadmodule "siptrace.so"
listen = hep_udp:127.0.0.1:60667
modparam("proto_hep", "hep_id", "[hep_dst] 127.0.0.1:50667; version=2")
modparam("siptrace", "trace_on", 0)
modparam("siptrace", "trace_id", "[tid]uri=hep:hep_dst")
...
route{
	sip_trace("tid");

in my configs. 

Then, to get the traces, 
ngrep -W byline -d lo port 50667

To enable the traces during runtime: 
 opensipsctl fifo sip_trace on
 opensipsctl fifo sip_trace tid on

And grab the trace with: 
 ngrep -W byline -d lo port 50667

(Not sure why it is needed to enable both the global and the trace-id... )

Best, 

Fabian


From farmorg at gmail.com  Tue Oct  1 10:22:26 2019
From: farmorg at gmail.com (Mark Farmer)
Date: Tue, 1 Oct 2019 15:22:26 +0100
Subject: [OpenSIPS-Users] Segfault on mi reload_routes
In-Reply-To: <CAOvQDetOA931jaRAEPQM95ZbeO3hghRLfojt-+P1_VXrFx3RbA@mail.gmail.com>
References: <CAOvQDevHk25_kSgA1g2vYjiTMF9hGLz9J1K0pamv-kMePxmtsw@mail.gmail.com>
 <CAOvQDetimYjB3vMempHj__nB7_eYFAZgmUbUAEkV7EVRQQrXNQ@mail.gmail.com>
 <CAOvQDesLM4s0=-5LaapbboD2jAnswbc2Vvk0CE_gyfNVOx4SXQ@mail.gmail.com>
 <71c5148f-d405-35b6-1e5e-ac244f32bb8b@opensips.org>
 <CAOvQDetOA931jaRAEPQM95ZbeO3hghRLfojt-+P1_VXrFx3RbA@mail.gmail.com>
Message-ID: <CAOvQDeux13=0sn7ga9Ve76TQA95Ki66Ox65vr-sPsrt26QaK+w@mail.gmail.com>

I saw the bug was fixed:

https://github.com/OpenSIPS/opensips/issues/1839

So I updated to latest git and all seems well now :)



On Thu, 26 Sep 2019 at 09:34, Mark Farmer <farmorg at gmail.com> wrote:

> Excellent, thank you.
>
>
> On Wed, 25 Sep 2019 at 17:12, Liviu Chircu <liviu at opensips.org> wrote:
>
>> Hi, Mark!
>>
>> Thank you for the nice backtrace!  I managed to reproduce the crash and
>> opened up a GitHub issue for it [1].
>> This should bring us one step closer to getting it fixed, so I suggest
>> we move the discussion over there.
>>
>> Cheers,
>>
>> [1]: https://github.com/OpenSIPS/opensips/issues/1839
>>
>> Liviu Chircu
>> OpenSIPS Developer
>> http://www.opensips-solutions.com
>>
>> On 25.09.2019 18:29, Mark Farmer wrote:
>> > OK, I figured out my mistake with gdb so I now have a 'bt full' output
>> > of the core dump
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Mark Farmer
> farmorg at gmail.com
>


-- 
Mark Farmer
farmorg at gmail.com
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From bogdan at opensips.org  Tue Oct  1 10:56:36 2019
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 1 Oct 2019 17:56:36 +0300
Subject: [OpenSIPS-Users] Segfault on mi reload_routes
In-Reply-To: <CAOvQDeux13=0sn7ga9Ve76TQA95Ki66Ox65vr-sPsrt26QaK+w@mail.gmail.com>
References: <CAOvQDevHk25_kSgA1g2vYjiTMF9hGLz9J1K0pamv-kMePxmtsw@mail.gmail.com>
 <CAOvQDetimYjB3vMempHj__nB7_eYFAZgmUbUAEkV7EVRQQrXNQ@mail.gmail.com>
 <CAOvQDesLM4s0=-5LaapbboD2jAnswbc2Vvk0CE_gyfNVOx4SXQ@mail.gmail.com>
 <71c5148f-d405-35b6-1e5e-ac244f32bb8b@opensips.org>
 <CAOvQDetOA931jaRAEPQM95ZbeO3hghRLfojt-+P1_VXrFx3RbA@mail.gmail.com>
 <CAOvQDeux13=0sn7ga9Ve76TQA95Ki66Ox65vr-sPsrt26QaK+w@mail.gmail.com>
Message-ID: <b3b48483-b20c-46f7-1694-644962d5bab0@opensips.org>

Yes, the fix will be part of 3.0.1 release (today). As the routes 
reloading is a new feature starting with 3.0.0, whatever future you may 
face here, just let me know.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Summit 2019
   https://www.opensips.org/events/Summit-2019Amsterdam/

On 10/1/19 5:22 PM, Mark Farmer wrote:
> I saw the bug was fixed:
>
> https://github.com/OpenSIPS/opensips/issues/1839
>
> So I updated to latest git and all seems well now :)
>
>
>
> On Thu, 26 Sep 2019 at 09:34, Mark Farmer <farmorg at gmail.com 
> <mailto:farmorg at gmail.com>> wrote:
>
>     Excellent, thank you.
>
>
>     On Wed, 25 Sep 2019 at 17:12, Liviu Chircu <liviu at opensips.org
>     <mailto:liviu at opensips.org>> wrote:
>
>         Hi, Mark!
>
>         Thank you for the nice backtrace!  I managed to reproduce the
>         crash and
>         opened up a GitHub issue for it [1].
>         This should bring us one step closer to getting it fixed, so I
>         suggest
>         we move the discussion over there.
>
>         Cheers,
>
>         [1]: https://github.com/OpenSIPS/opensips/issues/1839
>
>         Liviu Chircu
>         OpenSIPS Developer
>         http://www.opensips-solutions.com
>
>         On 25.09.2019 18:29, Mark Farmer wrote:
>         > OK, I figured out my mistake with gdb so I now have a 'bt
>         full' output
>         > of the core dump
>
>         _______________________________________________
>         Users mailing list
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>     -- 
>     Mark Farmer
>     farmorg at gmail.com <mailto:farmorg at gmail.com>
>
>
>
> -- 
> Mark Farmer
> farmorg at gmail.com <mailto:farmorg at gmail.com>
>
> _______________________________________________
> 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  Tue Oct  1 11:50:39 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 1 Oct 2019 18:50:39 +0300
Subject: [OpenSIPS-Users] [Release] Heads up,
	OpenSIPS 3.0.1 planned for 1st of October
In-Reply-To: <ce15ed1e-4bdd-8697-348e-15e3d3b096b6@opensips.org>
References: <fb973b81-3cf6-fb8b-7439-430dcdfb5eba@opensips.org>
 <ce15ed1e-4bdd-8697-348e-15e3d3b096b6@opensips.org>
Message-ID: <135f2d87-cfe6-a278-ddf7-fd15f6328899@opensips.org>

Aaaand its done :) I am happy to announce the new OpenSIPS 3.0.1 minor 
release is now available for you. You can find a full list of the the 
changes here:
https://opensips.org/pub/opensips/3.0.1/ChangeLog

Thank you all for helping us doing this!

Happy hacking!
-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


On 9/19/19 2:39 PM, Bogdan-Andrei Iancu wrote:
> Did I say 3.1 ?? my bad, it is 3.0.1 minor in the 3.0 family :)
> 
> Lesson: don't do major work before the morning coffee ;)
> 
> Regards,
> 
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS Summit 2019
>    https://www.opensips.org/events/Summit-2019Amsterdam/
> 
> On 9/19/19 11:21 AM, Bogdan-Andrei Iancu wrote:
>> Hi all,
>>
>> It has been a while since the OpenSIPS 3.0 release and thanks to our 
>> awesome community a great amount of testing and reporting work was 
>> done. Of course, this translated in good chunk of fixes that make 
>> OpenSIPS 3.x a more stable and shining piece of software.
>>
>> Shortly said, it is time for a new minor release, it is time for 
>> OpenSIPS 3.1. This is planned for 1st of October.
>>
>> We still have some fixes (and testing of course) on the pipe, due to 
>> the release date, but please take good usage of the remaining time and 
>> open reports if you are aware of any issues affecting OpenSIPS 3.0.
>>
>> Many thanks,
>> -- 
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>    https://www.opensips-solutions.com
>> OpenSIPS Summit 2019
>>    https://www.opensips.org/events/Summit-2019Amsterdam/
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> _______________________________________________
> Devel mailing list
> Devel at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
> 


From suharik71 at gmail.com  Wed Oct  2 03:20:03 2019
From: suharik71 at gmail.com (=?UTF-8?B?0JDQvdGC0L7QvSDQldGA0YjQvtCy?=)
Date: Wed, 2 Oct 2019 10:20:03 +0300
Subject: [OpenSIPS-Users] event_route doesn't catch events
Message-ID: <CAMp8X_vTmf9pfsS6qHTCYGM0LQ1wfw8hHCenPQcdgt2gcHdVow@mail.gmail.com>

Hello,
I want to catch dialog changes in event_route, but nothing happens
here is an example config

event_route[E_DLG_STATE_CHANGED] {
  xlog("L_INFO", "NEW CALL DIALOG STATE $param(new_state)");
  $avp(attrs) = "Call-Id";
  $avp(vals) = $ci;
  $avp(attrs) = "State";
  $avp(vals) = $param(new_state);
  if (!raise_event("E_SIP_DIALOG", $avp(attrs), $avp(vals)))
    xlog("L_ERR", "cannot raise E_SIP_DIALOG event\n");
  exit;
}

when calling in the logs there are no messages from event_route

# opensips -V
version: opensips 3.0.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, HP_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, sigio_rt, select.
git revision: dac56fa49
main.c compiled on 05:45:10 Oct  2 2019 with gcc 4.8.5
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From razvan at opensips.org  Wed Oct  2 04:22:46 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 2 Oct 2019 11:22:46 +0300
Subject: [OpenSIPS-Users] event_route doesn't catch events
In-Reply-To: <CAMp8X_vTmf9pfsS6qHTCYGM0LQ1wfw8hHCenPQcdgt2gcHdVow@mail.gmail.com>
References: <CAMp8X_vTmf9pfsS6qHTCYGM0LQ1wfw8hHCenPQcdgt2gcHdVow@mail.gmail.com>
Message-ID: <c2597cca-29eb-aada-ecd3-5cbbccee7404@opensips.org>

Are you sure you are calling `match_dialog()` for the sequential 
requests? Also, are you loading the `event_route` module in your script?

Best regards,
Răzvan

On 10/2/19 10:20 AM, Антон Ершов wrote:
> Hello,
> I want to catch dialog changes in event_route, but nothing happens
> here is an example config
> 
> event_route[E_DLG_STATE_CHANGED] {
>    xlog("L_INFO", "NEW CALL DIALOG STATE $param(new_state)");
>    $avp(attrs) = "Call-Id";
>    $avp(vals) = $ci;
>    $avp(attrs) = "State";
>    $avp(vals) = $param(new_state);
>    if (!raise_event("E_SIP_DIALOG", $avp(attrs), $avp(vals)))
>      xlog("L_ERR", "cannot raise E_SIP_DIALOG event\n");
>    exit;
> }
> 
> when calling in the logs there are no messages from event_route
> 
> # opensips -V
> version: opensips 3.0.1 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
> Q_MALLOC, F_MALLOC, HP_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, sigio_rt, select.
> git revision: dac56fa49
> main.c compiled on 05:45:10 Oct  2 2019 with gcc 4.8.5
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From suharik71 at gmail.com  Wed Oct  2 04:45:04 2019
From: suharik71 at gmail.com (=?UTF-8?B?0JDQvdGC0L7QvSDQldGA0YjQvtCy?=)
Date: Wed, 2 Oct 2019 11:45:04 +0300
Subject: [OpenSIPS-Users] event_route doesn't catch events
In-Reply-To: <c2597cca-29eb-aada-ecd3-5cbbccee7404@opensips.org>
References: <CAMp8X_vTmf9pfsS6qHTCYGM0LQ1wfw8hHCenPQcdgt2gcHdVow@mail.gmail.com>
 <c2597cca-29eb-aada-ecd3-5cbbccee7404@opensips.org>
Message-ID: <CAMp8X_tZqOhJ3Q3uza_o82mXW_DOHXptiOnkJJ8n8cPfceVdzA@mail.gmail.com>

sorry really my mistake

ср, 2 окт. 2019 г. в 11:26, Răzvan Crainea <razvan at opensips.org>:

> Are you sure you are calling `match_dialog()` for the sequential
> requests? Also, are you loading the `event_route` module in your script?
>
> Best regards,
> Răzvan
>
> On 10/2/19 10:20 AM, Антон Ершов wrote:
> > Hello,
> > I want to catch dialog changes in event_route, but nothing happens
> > here is an example config
> >
> > event_route[E_DLG_STATE_CHANGED] {
> >    xlog("L_INFO", "NEW CALL DIALOG STATE $param(new_state)");
> >    $avp(attrs) = "Call-Id";
> >    $avp(vals) = $ci;
> >    $avp(attrs) = "State";
> >    $avp(vals) = $param(new_state);
> >    if (!raise_event("E_SIP_DIALOG", $avp(attrs), $avp(vals)))
> >      xlog("L_ERR", "cannot raise E_SIP_DIALOG event\n");
> >    exit;
> > }
> >
> > when calling in the logs there are no messages from event_route
> >
> > # opensips -V
> > version: opensips 3.0.1 (x86_64/linux)
> > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> > Q_MALLOC, F_MALLOC, HP_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, sigio_rt, select.
> > git revision: dac56fa49
> > main.c compiled on 05:45:10 Oct  2 2019 with gcc 4.8.5
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
> _______________________________________________
> 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 Oct  2 07:40:46 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 2 Oct 2019 14:40:46 +0300
Subject: [OpenSIPS-Users] OpenSIPS at Astricon 2019
Message-ID: <10e27c61-b84f-37ad-8de0-e5df2fd24236@opensips.org>

Hi, Everyone!

I am glad to announce you that OpenSIPS is going to be present at 
Astricon 2019, Oct 29-30, in Atlanta. If you manage to get there, make 
sure you don't miss Liviu's presentation about Stir & Shaken [1], or my 
presentation about Call Center Solutions!

[1] 
https://astricon2019.sched.com/event/VMdL/stir-shaken-a-way-to-fight-back-call-deception-with-opensips?iframe=yes&w=100%&sidebar=yes&bg=no
[2] 
https://astricon2019.sched.com/event/VDA8/hybrid-call-center-solutions-with-opensips?iframe=yes&w=100%&sidebar=yes&bg=no#

See you in Atlanta!
-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From suharik71 at gmail.com  Wed Oct  2 08:04:05 2019
From: suharik71 at gmail.com (=?UTF-8?B?0JDQvdGC0L7QvSDQldGA0YjQvtCy?=)
Date: Wed, 2 Oct 2019 15:04:05 +0300
Subject: [OpenSIPS-Users] opensips 3.0.1 $dlg_val to event_route
Message-ID: <CAMp8X_v5Ax+GpLbUGBxem=e2Y97RBTfP5rBfD5Szzw237hTLNQ@mail.gmail.com>

good afternoon
tell me in version opensips 3.0.1 you can pass $dlg_val to event_route as
it was possible in version opensips 2.4
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From razvan at opensips.org  Wed Oct  2 09:48:36 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 2 Oct 2019 16:48:36 +0300
Subject: [OpenSIPS-Users] opensips 3.0.1 $dlg_val to event_route
In-Reply-To: <CAMp8X_v5Ax+GpLbUGBxem=e2Y97RBTfP5rBfD5Szzw237hTLNQ@mail.gmail.com>
References: <CAMp8X_v5Ax+GpLbUGBxem=e2Y97RBTfP5rBfD5Szzw237hTLNQ@mail.gmail.com>
Message-ID: <55fc13c5-564f-97c2-0433-bdbbd215459a@opensips.org>

Hello!

You can try to fetch the values of a dialog using the get_dialog_vals() 
[1], by indexing the values based on the callid.

[1] 
https://opensips.org/html/docs/modules/3.0.x/dialog.html#func_get_dialog_vals

Best regards,
Răzvan

On 10/2/19 3:04 PM, Антон Ершов wrote:
> good afternoon
> tell me in version opensips 3.0.1 you can pass $dlg_val to event_route 
> as it was possible in version opensips 2.4
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From tito at xsvoce.com  Wed Oct  2 22:03:24 2019
From: tito at xsvoce.com (Tito Cumpen)
Date: Wed, 2 Oct 2019 19:03:24 -0700
Subject: [OpenSIPS-Users] sip tls port and wss
Message-ID: <CANZPVB6gHHh9pgXZW=ipNtn5oq=00K==CZVwwyhu9cKngw4XMg@mail.gmail.com>

Hello,

I was wondering if sip tls and wss can use the same port to listen for
incoming connections? If not is a viable option to create two separate
servers and federate them for sip tls and wss?

Thanks,
Tito
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From gmaruzz at gmail.com  Thu Oct  3 02:14:29 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 3 Oct 2019 08:14:29 +0200
Subject: [OpenSIPS-Users] sip tls port and wss
In-Reply-To: <CANZPVB6gHHh9pgXZW=ipNtn5oq=00K==CZVwwyhu9cKngw4XMg@mail.gmail.com>
References: <CANZPVB6gHHh9pgXZW=ipNtn5oq=00K==CZVwwyhu9cKngw4XMg@mail.gmail.com>
Message-ID: <CALXCt0rjoxOh_nijJrRnjH+1_aA8mbuXpGWSROgoGZU_pKsatw@mail.gmail.com>

wss is a form of http(s) that goes "upgraded", has nothing to do with
"usual" tls...
usually you want to have wss on port 443, so clients on restrictive
firewall can reach you (because firewall "thinks" is https)
(for those restricted clients you then need a turn server able to listen
tls on port 443 of another machine, so they can get srtp via another "like
https")
if you don't care about restrictive firewalls, then you can put wss on any
port

On Thu, Oct 3, 2019 at 4:06 AM Tito Cumpen <tito at xsvoce.com> wrote:

> Hello,
>
> I was wondering if sip tls and wss can use the same port to listen for
> incoming connections? If not is a viable option to create two separate
> servers and federate them for sip tls and wss?
>
> Thanks,
> Tito
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From tito at xsvoce.com  Thu Oct  3 02:22:55 2019
From: tito at xsvoce.com (Tito Cumpen)
Date: Wed, 2 Oct 2019 23:22:55 -0700
Subject: [OpenSIPS-Users] sip tls port and wss
In-Reply-To: <CALXCt0rjoxOh_nijJrRnjH+1_aA8mbuXpGWSROgoGZU_pKsatw@mail.gmail.com>
References: <CANZPVB6gHHh9pgXZW=ipNtn5oq=00K==CZVwwyhu9cKngw4XMg@mail.gmail.com>
 <CALXCt0rjoxOh_nijJrRnjH+1_aA8mbuXpGWSROgoGZU_pKsatw@mail.gmail.com>
Message-ID: <CANZPVB7NWqyjkY=B7tSdBEmOX-KDAURqXFsB2jYGDO_RsRoCsg@mail.gmail.com>

Thanks for the reply Giovanni,

The restrictive firewall concern is the reason why I have tls sip running
on 443 already. Therefore I cannot account for the same destination
listening port within the same server.


On Wed, Oct 2, 2019, 11:17 PM Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> wss is a form of http(s) that goes "upgraded", has nothing to do with
> "usual" tls...
> usually you want to have wss on port 443, so clients on restrictive
> firewall can reach you (because firewall "thinks" is https)
> (for those restricted clients you then need a turn server able to listen
> tls on port 443 of another machine, so they can get srtp via another "like
> https")
> if you don't care about restrictive firewalls, then you can put wss on any
> port
>
> On Thu, Oct 3, 2019 at 4:06 AM Tito Cumpen <tito at xsvoce.com> wrote:
>
>> Hello,
>>
>> I was wondering if sip tls and wss can use the same port to listen for
>> incoming connections? If not is a viable option to create two separate
>> servers and federate them for sip tls and wss?
>>
>> Thanks,
>> Tito
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From gmaruzz at gmail.com  Thu Oct  3 02:30:26 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 3 Oct 2019 08:30:26 +0200
Subject: [OpenSIPS-Users] sip tls port and wss
In-Reply-To: <CANZPVB7NWqyjkY=B7tSdBEmOX-KDAURqXFsB2jYGDO_RsRoCsg@mail.gmail.com>
References: <CANZPVB6gHHh9pgXZW=ipNtn5oq=00K==CZVwwyhu9cKngw4XMg@mail.gmail.com>
 <CALXCt0rjoxOh_nijJrRnjH+1_aA8mbuXpGWSROgoGZU_pKsatw@mail.gmail.com>
 <CANZPVB7NWqyjkY=B7tSdBEmOX-KDAURqXFsB2jYGDO_RsRoCsg@mail.gmail.com>
Message-ID: <CALXCt0ofBc7DoCyH6XWLc8vbzu_A9EUH198arjyRqGDOOsq+0w@mail.gmail.com>

tls is not wss, and a moderately smart firewall will block tls on 443
wss is less likely to be blocked (need a deep packet inspection)

-giovanni

On Thu, Oct 3, 2019 at 8:23 AM Tito Cumpen <tito at xsvoce.com> wrote:

> Thanks for the reply Giovanni,
>
> The restrictive firewall concern is the reason why I have tls sip running
> on 443 already. Therefore I cannot account for the same destination
> listening port within the same server.
>
>
> On Wed, Oct 2, 2019, 11:17 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> wss is a form of http(s) that goes "upgraded", has nothing to do with
>> "usual" tls...
>> usually you want to have wss on port 443, so clients on restrictive
>> firewall can reach you (because firewall "thinks" is https)
>> (for those restricted clients you then need a turn server able to listen
>> tls on port 443 of another machine, so they can get srtp via another "like
>> https")
>> if you don't care about restrictive firewalls, then you can put wss on
>> any port
>>
>> On Thu, Oct 3, 2019 at 4:06 AM Tito Cumpen <tito at xsvoce.com> wrote:
>>
>>> Hello,
>>>
>>> I was wondering if sip tls and wss can use the same port to listen for
>>> incoming connections? If not is a viable option to create two separate
>>> servers and federate them for sip tls and wss?
>>>
>>> Thanks,
>>> Tito
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From farmorg at gmail.com  Thu Oct  3 06:35:09 2019
From: farmorg at gmail.com (Mark Farmer)
Date: Thu, 3 Oct 2019 11:35:09 +0100
Subject: [OpenSIPS-Users] Dialplan Help
Message-ID: <CAOvQDesYoRb2qqBEqToMYrQCh2dxvAKaU9WJwJ0UG9C9cg5hJg@mail.gmail.com>

Hi everyone

I'm having a problem with my rule not matching, I'm trying to remove the +
at the beginning of the input DID using a regex rule:

Matching exp:
^\+(44[1-9][0-9]+$)

Substitution exp:
^\+(.*)$

Replacement exp:
\1

But dp_translate() never matches.

mi dp_translate input=+441205875104 dpid=1
ERROR: command 'dp_translate' returned: 404: No translation

Where am I going wrong?

Thanks
Mark.
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From farmorg at gmail.com  Thu Oct  3 06:46:50 2019
From: farmorg at gmail.com (Mark Farmer)
Date: Thu, 3 Oct 2019 11:46:50 +0100
Subject: [OpenSIPS-Users] Dialplan Help
In-Reply-To: <CAOvQDesYoRb2qqBEqToMYrQCh2dxvAKaU9WJwJ0UG9C9cg5hJg@mail.gmail.com>
References: <CAOvQDesYoRb2qqBEqToMYrQCh2dxvAKaU9WJwJ0UG9C9cg5hJg@mail.gmail.com>
Message-ID: <CAOvQDes-Te6xp82A+B_+V97rw2F4KWgDBergOTxxXZand6gGsg@mail.gmail.com>

Never mind, seems setting matching operator to regex is the wrong thing to
do.

Sorry!


On Thu, 3 Oct 2019 at 11:35, Mark Farmer <farmorg at gmail.com> wrote:

> Hi everyone
>
> I'm having a problem with my rule not matching, I'm trying to remove the +
> at the beginning of the input DID using a regex rule:
>
> Matching exp:
> ^\+(44[1-9][0-9]+$)
>
> Substitution exp:
> ^\+(.*)$
>
> Replacement exp:
> \1
>
> But dp_translate() never matches.
>
> mi dp_translate input=+441205875104 dpid=1
> ERROR: command 'dp_translate' returned: 404: No translation
>
> Where am I going wrong?
>
> Thanks
> Mark.
>
>

-- 
Mark Farmer
farmorg at gmail.com
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From razvan at opensips.org  Thu Oct  3 12:24:54 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Thu, 3 Oct 2019 19:24:54 +0300
Subject: [OpenSIPS-Users] [BLOG] Re-homing your calls using OpenSIPS 3.0
Message-ID: <0a60a1e9-f170-47f3-e903-355841d87780@opensips.org>

Hi, everyone!

Check out my blog post presenting how you can do calls re-homing using 
OpenSIPS 3.0:
https://blog.opensips.org/2019/10/03/re-homing-your-calls-with-opensips-3-0/

Best regards,
-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From igorolhovskiy at gmail.com  Thu Oct  3 12:50:56 2019
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 3 Oct 2019 19:50:56 +0300
Subject: [OpenSIPS-Users] Getting what is eating 100% CPU
Message-ID: <EEA361D6-2110-4CF0-A9F0-5D6800FC4F33@getmailspring.com>

#Hi!

Is there any way of getting what eating 100% CPU on OpenSIPS 2.4.6?
Problem is it appears quite occasional and can't reproduce it, and problem disappears on reboot of process.

#opensipsctl fifo ps
Process:: ID=0 PID=24470 Type=attendant
Process:: ID=1 PID=24471 Type=MI FIFO
Process:: ID=2 PID=24472 Type=time_keeper
Process:: ID=3 PID=24473 Type=timer
Process:: ID=4 PID=24474 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=5 PID=24475 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=6 PID=24476 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=7 PID=24477 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=8 PID=24478 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=9 PID=24479 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=10 PID=24480 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=11 PID=24481 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=12 PID=24482 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=13 PID=24483 Type=SIP receiver udp:172.31.37.148:5060
Process:: ID=14 PID=24484 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=15 PID=24485 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=16 PID=24486 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=17 PID=24487 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=18 PID=24488 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=19 PID=24489 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=20 PID=24490 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=21 PID=24491 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=22 PID=24492 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=23 PID=24493 Type=SIP receiver udp:172.31.37.148:9094
Process:: ID=24 PID=24494 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=25 PID=24495 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=26 PID=24496 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=27 PID=24497 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=28 PID=24498 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=29 PID=24499 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=30 PID=24500 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=31 PID=24501 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=32 PID=24502 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=33 PID=24503 Type=SIP receiver udp:172.31.38.207:5060
Process:: ID=34 PID=24504 Type=TCP receiver
Process:: ID=35 PID=24505 Type=TCP receiver
Process:: ID=36 PID=24506 Type=TCP receiver
Process:: ID=37 PID=24507 Type=TCP receiver
Process:: ID=38 PID=24508 Type=TCP receiver
Process:: ID=39 PID=24509 Type=TCP receiver
Process:: ID=40 PID=24510 Type=TCP receiver
Process:: ID=41 PID=24511 Type=TCP receiver
Process:: ID=42 PID=24512 Type=Timer handler
Process:: ID=43 PID=24513 Type=TCP main

In the logs
Oct 3 16:22:37 ip-172-31-37-148 /usr/local/sbin/opensips[24473]: WARNING:core:timer_ticker: timer task <nh-timer> already scheduled for 29546470 ms (now 33307710 ms), it may overlap..
#opensipsctl fifo get_statistics load: core: shmem:
load:load:: 30
load:load1m:: 30
load:load10m:: 30
load:load-all:: 29
load:load1m-all:: 29
load:load10m-all:: 29
load:load-proc-1:: 0
load:load1m-proc-1:: 0
load:load10m-proc-1:: 0
load:load-proc-2:: 0
load:load1m-proc-2:: 0
load:load10m-proc-2:: 0
load:load-proc-3:: 0
load:load1m-proc-3:: 0
load:load10m-proc-3:: 0
load:load-proc-4:: 0
load:load1m-proc-4:: 0
load:load10m-proc-4:: 0
load:load-proc-5:: 0
load:load1m-proc-5:: 0
load:load10m-proc-5:: 0
load:load-proc-6:: 0
load:load1m-proc-6:: 0
load:load10m-proc-6:: 0
load:load-proc-7:: 0
load:load1m-proc-7:: 0
load:load10m-proc-7:: 0
load:load-proc-8:: 0
load:load1m-proc-8:: 0
load:load10m-proc-8:: 0
load:load-proc-9:: 0
load:load1m-proc-9:: 0
load:load10m-proc-9:: 0
load:load-proc-10:: 0
load:load1m-proc-10:: 0
load:load10m-proc-10:: 0
load:load-proc-11:: 0
load:load1m-proc-11:: 0
load:load10m-proc-11:: 0
load:load-proc-12:: 0
load:load1m-proc-12:: 0
load:load10m-proc-12:: 0
load:load-proc-13:: 0
load:load1m-proc-13:: 0
load:load10m-proc-13:: 0
load:load-proc-14:: 0
load:load1m-proc-14:: 0
load:load10m-proc-14:: 0
load:load-proc-15:: 0
load:load1m-proc-15:: 0
load:load10m-proc-15:: 0
load:load-proc-16:: 0
load:load1m-proc-16:: 0
load:load10m-proc-16:: 0
load:load-proc-17:: 0
load:load1m-proc-17:: 0
load:load10m-proc-17:: 0
load:load-proc-18:: 0
load:load1m-proc-18:: 0
load:load10m-proc-18:: 0
load:load-proc-19:: 0
load:load1m-proc-19:: 0
load:load10m-proc-19:: 0
load:load-proc-20:: 0
load:load1m-proc-20:: 0
load:load10m-proc-20:: 0
load:load-proc-21:: 0
load:load1m-proc-21:: 0
load:load10m-proc-21:: 0
load:load-proc-22:: 0
load:load1m-proc-22:: 0
load:load10m-proc-22:: 0
load:load-proc-23:: 0
load:load1m-proc-23:: 0
load:load10m-proc-23:: 0
load:load-proc-24:: 100
load:load1m-proc-24:: 100
load:load10m-proc-24:: 100
load:load-proc-25:: 100
load:load1m-proc-25:: 100
load:load10m-proc-25:: 100
load:load-proc-26:: 100
load:load1m-proc-26:: 100
load:load10m-proc-26:: 100
load:load-proc-27:: 100
load:load1m-proc-27:: 100
load:load10m-proc-27:: 100
load:load-proc-28:: 100
load:load1m-proc-28:: 100
load:load10m-proc-28:: 100
load:load-proc-29:: 100
load:load1m-proc-29:: 100
load:load10m-proc-29:: 100
load:load-proc-30:: 100
load:load1m-proc-30:: 100
load:load10m-proc-30:: 100
load:load-proc-31:: 100
load:load1m-proc-31:: 100
load:load10m-proc-31:: 100
load:load-proc-32:: 100
load:load1m-proc-32:: 100
load:load10m-proc-32:: 100
load:load-proc-33:: 100
load:load1m-proc-33:: 100
load:load10m-proc-33:: 100
load:load-proc-34:: 0
load:load1m-proc-34:: 0
load:load10m-proc-34:: 0
load:load-proc-35:: 0
load:load1m-proc-35:: 0
load:load10m-proc-35:: 0
load:load-proc-36:: 100
load:load1m-proc-36:: 100
load:load10m-proc-36:: 100
load:load-proc-37:: 0
load:load1m-proc-37:: 0
load:load10m-proc-37:: 0
load:load-proc-38:: 0
load:load1m-proc-38:: 0
load:load10m-proc-38:: 0
load:load-proc-39:: 0
load:load1m-proc-39:: 0
load:load10m-proc-39:: 0
load:load-proc-40:: 0
load:load1m-proc-40:: 0
load:load10m-proc-40:: 0
load:load-proc-41:: 0
load:load1m-proc-41:: 0
load:load10m-proc-41:: 0
load:load-proc-42:: 100
load:load1m-proc-42:: 100
load:load10m-proc-42:: 100
load:load-proc-43:: 0
load:load1m-proc-43:: 0
load:load10m-proc-43:: 0
core:rcv_requests:: 50509
core:rcv_replies:: 31039
core:fwd_requests:: 59
core:fwd_replies:: 0
core:drop_requests:: 0
core:drop_replies:: 0
core:err_requests:: 0
core:err_replies:: 0
core:bad_URIs_rcvd:: 0
core:unsupported_methods:: 0
core:bad_msg_hdr:: 0
core:timestamp:: 33581
shmem:total_size:: 33554432
shmem:used_size:: 5929544
shmem:real_used_size:: 6187752
shmem:max_used_size:: 7568672
shmem:free_size:: 27366680
shmem:fragments:: 8137

As OpenSIPS is not crashing, I also don't know how to get core dump here.
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From todd at firestreamcloud.com  Thu Oct  3 13:57:13 2019
From: todd at firestreamcloud.com (Todd Routhier)
Date: Thu, 3 Oct 2019 12:57:13 -0500
Subject: [OpenSIPS-Users] MI command failed with code 406 (Unexpected method)
Message-ID: <CA+za3LL=hRTK=2M6Hp0o46Ldu2w_bXbzoJMP7vYRxWT6oBx1nw@mail.gmail.com>

I installed a recent version of OpenSIPs and I think the latest version of
OpenSIPs Control Panel. I will follow up with exact versions in just a bit.

Basic functions such as adding/removing users work but many other functions
such as clicking the little telephone icon in the user's list to see
contacts registered to the SIP account, result in the error

MI command failed with code 406 (Unexpected method)

I have checked the box configuration and believe I have it set correctly
but, obviously I am missing something. Have googled my brains out, read the
book which seems a bit out-dated and I am at the end of my rope, out of
ideas.

Thought I would post here and hope the experts can offer some advice and/or
guidance. I will follow this post up with some more specifics such as
version info and some of my config info.

Thanks in advance for any assistance you can provide.

--Todd
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From tito at xsvoce.com  Thu Oct  3 17:49:07 2019
From: tito at xsvoce.com (Tito Cumpen)
Date: Thu, 3 Oct 2019 14:49:07 -0700
Subject: [OpenSIPS-Users] Checking ongoing Dialog
Message-ID: <CANZPVB46WKxyTwFNPFMVtunW+16jhd8uvKX84h2DvW7AYPmwMw@mail.gmail.com>

Hello Group,


I am running into an issue caused by injecting(voip push) a newly
registered branch as the destination to an ongoing dialog. Is there a way
to check whether the call has already been answered before injecting the
branch and forwarding the invite to the new AS location?

Thanks,
Tito
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From razvan at opensips.org  Fri Oct  4 02:51:41 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 4 Oct 2019 09:51:41 +0300
Subject: [OpenSIPS-Users] Getting what is eating 100% CPU
In-Reply-To: <EEA361D6-2110-4CF0-A9F0-5D6800FC4F33@getmailspring.com>
References: <EEA361D6-2110-4CF0-A9F0-5D6800FC4F33@getmailspring.com>
Message-ID: <27826ccc-dcb3-3b93-ba98-c3a74864afe3@opensips.org>

This looks like a deadlock: could you indicate the exact revision of 
OpenSIPS you are using?
Also, when opensips is running in 100%, could you run the `opensipsctl 
trap` and post the resulted gdb file?

Best regards,
Răzvan

On 10/3/19 7:50 PM, Igor Olhovskiy wrote:
> #Hi!
> 
> Is there any way of getting what eating 100% CPU on OpenSIPS 2.4.6?
> Problem is it appears quite occasional and can't reproduce it, and 
> problem disappears on reboot of process.
> 
> #opensipsctl fifo ps
> Process::  ID=0 PID=24470 Type=attendant
> Process::  ID=1 PID=24471 Type=MI FIFO
> Process::  ID=2 PID=24472 Type=time_keeper
> Process::  ID=3 PID=24473 Type=timer
> Process::  ID=4 PID=24474 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=5 PID=24475 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=6 PID=24476 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=7 PID=24477 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=8 PID=24478 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=9 PID=24479 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=10 PID=24480 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=11 PID=24481 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=12 PID=24482 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=13 PID=24483 Type=SIP receiver udp:172.31.37.148:5060
> Process::  ID=14 PID=24484 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=15 PID=24485 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=16 PID=24486 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=17 PID=24487 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=18 PID=24488 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=19 PID=24489 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=20 PID=24490 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=21 PID=24491 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=22 PID=24492 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=23 PID=24493 Type=SIP receiver udp:172.31.37.148:9094
> Process::  ID=24 PID=24494 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=25 PID=24495 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=26 PID=24496 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=27 PID=24497 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=28 PID=24498 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=29 PID=24499 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=30 PID=24500 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=31 PID=24501 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=32 PID=24502 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=33 PID=24503 Type=SIP receiver udp:172.31.38.207:5060
> Process::  ID=34 PID=24504 Type=TCP receiver
> Process::  ID=35 PID=24505 Type=TCP receiver
> Process::  ID=36 PID=24506 Type=TCP receiver
> Process::  ID=37 PID=24507 Type=TCP receiver
> Process::  ID=38 PID=24508 Type=TCP receiver
> Process::  ID=39 PID=24509 Type=TCP receiver
> Process::  ID=40 PID=24510 Type=TCP receiver
> Process::  ID=41 PID=24511 Type=TCP receiver
> Process::  ID=42 PID=24512 Type=Timer handler
> Process::  ID=43 PID=24513 Type=TCP main
> 
> In the logs
> 
> Oct  3 16:22:37 ip-172-31-37-148 /usr/local/sbin/opensips[24473]: 
> WARNING:core:timer_ticker: timer task <nh-timer> already scheduled for 
> 29546470 ms (now 33307710 ms), it may overlap..
> 
> #opensipsctl fifo get_statistics load: core: shmem:
> load:load:: 30
> load:load1m:: 30
> load:load10m:: 30
> load:load-all:: 29
> load:load1m-all:: 29
> load:load10m-all:: 29
> load:load-proc-1:: 0
> load:load1m-proc-1:: 0
> load:load10m-proc-1:: 0
> load:load-proc-2:: 0
> load:load1m-proc-2:: 0
> load:load10m-proc-2:: 0
> load:load-proc-3:: 0
> load:load1m-proc-3:: 0
> load:load10m-proc-3:: 0
> load:load-proc-4:: 0
> load:load1m-proc-4:: 0
> load:load10m-proc-4:: 0
> load:load-proc-5:: 0
> load:load1m-proc-5:: 0
> load:load10m-proc-5:: 0
> load:load-proc-6:: 0
> load:load1m-proc-6:: 0
> load:load10m-proc-6:: 0
> load:load-proc-7:: 0
> load:load1m-proc-7:: 0
> load:load10m-proc-7:: 0
> load:load-proc-8:: 0
> load:load1m-proc-8:: 0
> load:load10m-proc-8:: 0
> load:load-proc-9:: 0
> load:load1m-proc-9:: 0
> load:load10m-proc-9:: 0
> load:load-proc-10:: 0
> load:load1m-proc-10:: 0
> load:load10m-proc-10:: 0
> load:load-proc-11:: 0
> load:load1m-proc-11:: 0
> load:load10m-proc-11:: 0
> load:load-proc-12:: 0
> load:load1m-proc-12:: 0
> load:load10m-proc-12:: 0
> load:load-proc-13:: 0
> load:load1m-proc-13:: 0
> load:load10m-proc-13:: 0
> load:load-proc-14:: 0
> load:load1m-proc-14:: 0
> load:load10m-proc-14:: 0
> load:load-proc-15:: 0
> load:load1m-proc-15:: 0
> load:load10m-proc-15:: 0
> load:load-proc-16:: 0
> load:load1m-proc-16:: 0
> load:load10m-proc-16:: 0
> load:load-proc-17:: 0
> load:load1m-proc-17:: 0
> load:load10m-proc-17:: 0
> load:load-proc-18:: 0
> load:load1m-proc-18:: 0
> load:load10m-proc-18:: 0
> load:load-proc-19:: 0
> load:load1m-proc-19:: 0
> load:load10m-proc-19:: 0
> load:load-proc-20:: 0
> load:load1m-proc-20:: 0
> load:load10m-proc-20:: 0
> load:load-proc-21:: 0
> load:load1m-proc-21:: 0
> load:load10m-proc-21:: 0
> load:load-proc-22:: 0
> load:load1m-proc-22:: 0
> load:load10m-proc-22:: 0
> load:load-proc-23:: 0
> load:load1m-proc-23:: 0
> load:load10m-proc-23:: 0
> load:load-proc-24:: 100
> load:load1m-proc-24:: 100
> load:load10m-proc-24:: 100
> load:load-proc-25:: 100
> load:load1m-proc-25:: 100
> load:load10m-proc-25:: 100
> load:load-proc-26:: 100
> load:load1m-proc-26:: 100
> load:load10m-proc-26:: 100
> load:load-proc-27:: 100
> load:load1m-proc-27:: 100
> load:load10m-proc-27:: 100
> load:load-proc-28:: 100
> load:load1m-proc-28:: 100
> load:load10m-proc-28:: 100
> load:load-proc-29:: 100
> load:load1m-proc-29:: 100
> load:load10m-proc-29:: 100
> load:load-proc-30:: 100
> load:load1m-proc-30:: 100
> load:load10m-proc-30:: 100
> load:load-proc-31:: 100
> load:load1m-proc-31:: 100
> load:load10m-proc-31:: 100
> load:load-proc-32:: 100
> load:load1m-proc-32:: 100
> load:load10m-proc-32:: 100
> load:load-proc-33:: 100
> load:load1m-proc-33:: 100
> load:load10m-proc-33:: 100
> load:load-proc-34:: 0
> load:load1m-proc-34:: 0
> load:load10m-proc-34:: 0
> load:load-proc-35:: 0
> load:load1m-proc-35:: 0
> load:load10m-proc-35:: 0
> load:load-proc-36:: 100
> load:load1m-proc-36:: 100
> load:load10m-proc-36:: 100
> load:load-proc-37:: 0
> load:load1m-proc-37:: 0
> load:load10m-proc-37:: 0
> load:load-proc-38:: 0
> load:load1m-proc-38:: 0
> load:load10m-proc-38:: 0
> load:load-proc-39:: 0
> load:load1m-proc-39:: 0
> load:load10m-proc-39:: 0
> load:load-proc-40:: 0
> load:load1m-proc-40:: 0
> load:load10m-proc-40:: 0
> load:load-proc-41:: 0
> load:load1m-proc-41:: 0
> load:load10m-proc-41:: 0
> load:load-proc-42:: 100
> load:load1m-proc-42:: 100
> load:load10m-proc-42:: 100
> load:load-proc-43:: 0
> load:load1m-proc-43:: 0
> load:load10m-proc-43:: 0
> core:rcv_requests:: 50509
> core:rcv_replies:: 31039
> core:fwd_requests:: 59
> core:fwd_replies:: 0
> core:drop_requests:: 0
> core:drop_replies:: 0
> core:err_requests:: 0
> core:err_replies:: 0
> core:bad_URIs_rcvd:: 0
> core:unsupported_methods:: 0
> core:bad_msg_hdr:: 0
> core:timestamp:: 33581
> shmem:total_size:: 33554432
> shmem:used_size:: 5929544
> shmem:real_used_size:: 6187752
> shmem:max_used_size:: 7568672
> shmem:free_size:: 27366680
> shmem:fragments:: 8137
> 
> As OpenSIPS is not crashing, I also don't know how to get core dump here.
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From razvan at opensips.org  Fri Oct  4 02:55:00 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 4 Oct 2019 09:55:00 +0300
Subject: [OpenSIPS-Users] MI command failed with code 406 (Unexpected
 method)
In-Reply-To: <CA+za3LL=hRTK=2M6Hp0o46Ldu2w_bXbzoJMP7vYRxWT6oBx1nw@mail.gmail.com>
References: <CA+za3LL=hRTK=2M6Hp0o46Ldu2w_bXbzoJMP7vYRxWT6oBx1nw@mail.gmail.com>
Message-ID: <4aee4d70-9445-e591-cbce-1074ad5b5775@opensips.org>

If you're using the last year release (2.4), this will not work with the 
latest OpenSIPS CP (8.3.0), as this is only meant to work with OpenSIPS 
3.0. For OpenSIPS 2.4 you need to use OpenSIPS Control Panel version 7.2.3.
So please provide us the exact versions you're using for both software 
to provide you more details.

Best regards,
Răzvan

On 10/3/19 8:57 PM, Todd Routhier wrote:
> I installed a recent version of OpenSIPs and I think the latest version 
> of OpenSIPs Control Panel. I will follow up with exact versions in just 
> a bit.
> 
> Basic functions such as adding/removing users work but many other 
> functions such as clicking the little telephone icon in the user's list 
> to see contacts registered to the SIP account, result in the error
> 
> MI command failed with code 406 (Unexpected method)
> 
> I have checked the box configuration and believe I have it set correctly 
> but, obviously I am missing something. Have googled my brains out, read 
> the book which seems a bit out-dated and I am at the end of my rope, out 
> of ideas.
> 
> Thought I would post here and hope the experts can offer some advice 
> and/or guidance. I will follow this post up with some more specifics 
> such as version info and some of my config info.
> 
> Thanks in advance for any assistance you can provide.
> 
> --Todd
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From razvan at opensips.org  Fri Oct  4 02:56:53 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 4 Oct 2019 09:56:53 +0300
Subject: [OpenSIPS-Users] Checking ongoing Dialog
In-Reply-To: <CANZPVB46WKxyTwFNPFMVtunW+16jhd8uvKX84h2DvW7AYPmwMw@mail.gmail.com>
References: <CANZPVB46WKxyTwFNPFMVtunW+16jhd8uvKX84h2DvW7AYPmwMw@mail.gmail.com>
Message-ID: <8015d387-7e38-e0f8-5b3d-c6d6f035eec6@opensips.org>

I believe you can check the $DLG_state in branch_route for the newly 
injected branch.
However, are you saying that an INVITE is sent towards a destination 
_after_ the call has been answered? Not sure if that's possible, could 
you provide us a trace?

Best regards,
Razvan

On 10/4/19 12:49 AM, Tito Cumpen wrote:
> Hello Group,
> 
> 
> I am running into an issue caused by injecting(voip push) a newly 
> registered branch as the destination to an ongoing dialog. Is there a 
> way to check whether the call has already been answered before injecting 
> the branch and forwarding the invite to the new AS location?
> 
> Thanks,
> Tito
> 
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From igorolhovskiy at gmail.com  Fri Oct  4 03:33:08 2019
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 4 Oct 2019 10:33:08 +0300
Subject: [OpenSIPS-Users] Getting what is eating 100% CPU
In-Reply-To: <27826ccc-dcb3-3b93-ba98-c3a74864afe3@opensips.org>
References: <27826ccc-dcb3-3b93-ba98-c3a74864afe3@opensips.org>
Message-ID: <00A9B7C0-8A8D-40FB-BCA0-C81CC325F642@getmailspring.com>

opensips -V
version: opensips 2.4.6 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, 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, sigio_rt, select.
git revision: 50a37b3a2
main.c compiled on 01:19:10 Jul 17 2019 with gcc 7

Will try to get opensipsctl trap when problem occures. Thanks!
On Oct 4 2019, at 9:51 am, Răzvan Crainea <razvan at opensips.org> wrote:
> This looks like a deadlock: could you indicate the exact revision of
> OpenSIPS you are using?
> Also, when opensips is running in 100%, could you run the `opensipsctl
> trap` and post the resulted gdb file?
>
> Best regards,
> Răzvan
>
> On 10/3/19 7:50 PM, Igor Olhovskiy wrote:
> > #Hi!
> >
> > Is there any way of getting what eating 100% CPU on OpenSIPS 2.4.6?
> > Problem is it appears quite occasional and can't reproduce it, and
> > problem disappears on reboot of process.
> >
> > #opensipsctl fifo ps
> > Process:: ID=0 PID=24470 Type=attendant
> > Process:: ID=1 PID=24471 Type=MI FIFO
> > Process:: ID=2 PID=24472 Type=time_keeper
> > Process:: ID=3 PID=24473 Type=timer
> > Process:: ID=4 PID=24474 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=5 PID=24475 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=6 PID=24476 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=7 PID=24477 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=8 PID=24478 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=9 PID=24479 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=10 PID=24480 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=11 PID=24481 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=12 PID=24482 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=13 PID=24483 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=14 PID=24484 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=15 PID=24485 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=16 PID=24486 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=17 PID=24487 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=18 PID=24488 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=19 PID=24489 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=20 PID=24490 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=21 PID=24491 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=22 PID=24492 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=23 PID=24493 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=24 PID=24494 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=25 PID=24495 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=26 PID=24496 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=27 PID=24497 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=28 PID=24498 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=29 PID=24499 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=30 PID=24500 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=31 PID=24501 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=32 PID=24502 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=33 PID=24503 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=34 PID=24504 Type=TCP receiver
> > Process:: ID=35 PID=24505 Type=TCP receiver
> > Process:: ID=36 PID=24506 Type=TCP receiver
> > Process:: ID=37 PID=24507 Type=TCP receiver
> > Process:: ID=38 PID=24508 Type=TCP receiver
> > Process:: ID=39 PID=24509 Type=TCP receiver
> > Process:: ID=40 PID=24510 Type=TCP receiver
> > Process:: ID=41 PID=24511 Type=TCP receiver
> > Process:: ID=42 PID=24512 Type=Timer handler
> > Process:: ID=43 PID=24513 Type=TCP main
> >
> > In the logs
> > Oct 3 16:22:37 ip-172-31-37-148 /usr/local/sbin/opensips[24473]:
> > WARNING:core:timer_ticker: timer task <nh-timer> already scheduled for
> > 29546470 ms (now 33307710 ms), it may overlap..
> >
> > #opensipsctl fifo get_statistics load: core: shmem:
> > load:load:: 30
> > load:load1m:: 30
> > load:load10m:: 30
> > load:load-all:: 29
> > load:load1m-all:: 29
> > load:load10m-all:: 29
> > load:load-proc-1:: 0
> > load:load1m-proc-1:: 0
> > load:load10m-proc-1:: 0
> > load:load-proc-2:: 0
> > load:load1m-proc-2:: 0
> > load:load10m-proc-2:: 0
> > load:load-proc-3:: 0
> > load:load1m-proc-3:: 0
> > load:load10m-proc-3:: 0
> > load:load-proc-4:: 0
> > load:load1m-proc-4:: 0
> > load:load10m-proc-4:: 0
> > load:load-proc-5:: 0
> > load:load1m-proc-5:: 0
> > load:load10m-proc-5:: 0
> > load:load-proc-6:: 0
> > load:load1m-proc-6:: 0
> > load:load10m-proc-6:: 0
> > load:load-proc-7:: 0
> > load:load1m-proc-7:: 0
> > load:load10m-proc-7:: 0
> > load:load-proc-8:: 0
> > load:load1m-proc-8:: 0
> > load:load10m-proc-8:: 0
> > load:load-proc-9:: 0
> > load:load1m-proc-9:: 0
> > load:load10m-proc-9:: 0
> > load:load-proc-10:: 0
> > load:load1m-proc-10:: 0
> > load:load10m-proc-10:: 0
> > load:load-proc-11:: 0
> > load:load1m-proc-11:: 0
> > load:load10m-proc-11:: 0
> > load:load-proc-12:: 0
> > load:load1m-proc-12:: 0
> > load:load10m-proc-12:: 0
> > load:load-proc-13:: 0
> > load:load1m-proc-13:: 0
> > load:load10m-proc-13:: 0
> > load:load-proc-14:: 0
> > load:load1m-proc-14:: 0
> > load:load10m-proc-14:: 0
> > load:load-proc-15:: 0
> > load:load1m-proc-15:: 0
> > load:load10m-proc-15:: 0
> > load:load-proc-16:: 0
> > load:load1m-proc-16:: 0
> > load:load10m-proc-16:: 0
> > load:load-proc-17:: 0
> > load:load1m-proc-17:: 0
> > load:load10m-proc-17:: 0
> > load:load-proc-18:: 0
> > load:load1m-proc-18:: 0
> > load:load10m-proc-18:: 0
> > load:load-proc-19:: 0
> > load:load1m-proc-19:: 0
> > load:load10m-proc-19:: 0
> > load:load-proc-20:: 0
> > load:load1m-proc-20:: 0
> > load:load10m-proc-20:: 0
> > load:load-proc-21:: 0
> > load:load1m-proc-21:: 0
> > load:load10m-proc-21:: 0
> > load:load-proc-22:: 0
> > load:load1m-proc-22:: 0
> > load:load10m-proc-22:: 0
> > load:load-proc-23:: 0
> > load:load1m-proc-23:: 0
> > load:load10m-proc-23:: 0
> > load:load-proc-24:: 100
> > load:load1m-proc-24:: 100
> > load:load10m-proc-24:: 100
> > load:load-proc-25:: 100
> > load:load1m-proc-25:: 100
> > load:load10m-proc-25:: 100
> > load:load-proc-26:: 100
> > load:load1m-proc-26:: 100
> > load:load10m-proc-26:: 100
> > load:load-proc-27:: 100
> > load:load1m-proc-27:: 100
> > load:load10m-proc-27:: 100
> > load:load-proc-28:: 100
> > load:load1m-proc-28:: 100
> > load:load10m-proc-28:: 100
> > load:load-proc-29:: 100
> > load:load1m-proc-29:: 100
> > load:load10m-proc-29:: 100
> > load:load-proc-30:: 100
> > load:load1m-proc-30:: 100
> > load:load10m-proc-30:: 100
> > load:load-proc-31:: 100
> > load:load1m-proc-31:: 100
> > load:load10m-proc-31:: 100
> > load:load-proc-32:: 100
> > load:load1m-proc-32:: 100
> > load:load10m-proc-32:: 100
> > load:load-proc-33:: 100
> > load:load1m-proc-33:: 100
> > load:load10m-proc-33:: 100
> > load:load-proc-34:: 0
> > load:load1m-proc-34:: 0
> > load:load10m-proc-34:: 0
> > load:load-proc-35:: 0
> > load:load1m-proc-35:: 0
> > load:load10m-proc-35:: 0
> > load:load-proc-36:: 100
> > load:load1m-proc-36:: 100
> > load:load10m-proc-36:: 100
> > load:load-proc-37:: 0
> > load:load1m-proc-37:: 0
> > load:load10m-proc-37:: 0
> > load:load-proc-38:: 0
> > load:load1m-proc-38:: 0
> > load:load10m-proc-38:: 0
> > load:load-proc-39:: 0
> > load:load1m-proc-39:: 0
> > load:load10m-proc-39:: 0
> > load:load-proc-40:: 0
> > load:load1m-proc-40:: 0
> > load:load10m-proc-40:: 0
> > load:load-proc-41:: 0
> > load:load1m-proc-41:: 0
> > load:load10m-proc-41:: 0
> > load:load-proc-42:: 100
> > load:load1m-proc-42:: 100
> > load:load10m-proc-42:: 100
> > load:load-proc-43:: 0
> > load:load1m-proc-43:: 0
> > load:load10m-proc-43:: 0
> > core:rcv_requests:: 50509
> > core:rcv_replies:: 31039
> > core:fwd_requests:: 59
> > core:fwd_replies:: 0
> > core:drop_requests:: 0
> > core:drop_replies:: 0
> > core:err_requests:: 0
> > core:err_replies:: 0
> > core:bad_URIs_rcvd:: 0
> > core:unsupported_methods:: 0
> > core:bad_msg_hdr:: 0
> > core:timestamp:: 33581
> > shmem:total_size:: 33554432
> > shmem:used_size:: 5929544
> > shmem:real_used_size:: 6187752
> > shmem:max_used_size:: 7568672
> > shmem:free_size:: 27366680
> > shmem:fragments:: 8137
> >
> > As OpenSIPS is not crashing, I also don't know how to get core dump here.
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From igorolhovskiy at gmail.com  Fri Oct  4 03:32:18 2019
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 4 Oct 2019 10:32:18 +0300
Subject: [OpenSIPS-Users] Getting what is eating 100% CPU
In-Reply-To: <27826ccc-dcb3-3b93-ba98-c3a74864afe3@opensips.org>
References: <27826ccc-dcb3-3b93-ba98-c3a74864afe3@opensips.org>
Message-ID: <00A9B7C0-8A8D-40FB-BCA0-C81CC325F642@getmailspring.com>

opensips -V
version: opensips 2.4.6 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, 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, sigio_rt, select.
git revision: 50a37b3a2
main.c compiled on 01:19:10 Jul 17 2019 with gcc 7

Will try to get opensipsctl trap when problem occures. Thanks!
On Oct 4 2019, at 9:51 am, Răzvan Crainea <razvan at opensips.org> wrote:
> This looks like a deadlock: could you indicate the exact revision of
> OpenSIPS you are using?
> Also, when opensips is running in 100%, could you run the `opensipsctl
> trap` and post the resulted gdb file?
>
> Best regards,
> Răzvan
>
> On 10/3/19 7:50 PM, Igor Olhovskiy wrote:
> > #Hi!
> >
> > Is there any way of getting what eating 100% CPU on OpenSIPS 2.4.6?
> > Problem is it appears quite occasional and can't reproduce it, and
> > problem disappears on reboot of process.
> >
> > #opensipsctl fifo ps
> > Process:: ID=0 PID=24470 Type=attendant
> > Process:: ID=1 PID=24471 Type=MI FIFO
> > Process:: ID=2 PID=24472 Type=time_keeper
> > Process:: ID=3 PID=24473 Type=timer
> > Process:: ID=4 PID=24474 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=5 PID=24475 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=6 PID=24476 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=7 PID=24477 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=8 PID=24478 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=9 PID=24479 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=10 PID=24480 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=11 PID=24481 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=12 PID=24482 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=13 PID=24483 Type=SIP receiver udp:172.31.37.148:5060
> > Process:: ID=14 PID=24484 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=15 PID=24485 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=16 PID=24486 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=17 PID=24487 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=18 PID=24488 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=19 PID=24489 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=20 PID=24490 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=21 PID=24491 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=22 PID=24492 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=23 PID=24493 Type=SIP receiver udp:172.31.37.148:9094
> > Process:: ID=24 PID=24494 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=25 PID=24495 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=26 PID=24496 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=27 PID=24497 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=28 PID=24498 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=29 PID=24499 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=30 PID=24500 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=31 PID=24501 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=32 PID=24502 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=33 PID=24503 Type=SIP receiver udp:172.31.38.207:5060
> > Process:: ID=34 PID=24504 Type=TCP receiver
> > Process:: ID=35 PID=24505 Type=TCP receiver
> > Process:: ID=36 PID=24506 Type=TCP receiver
> > Process:: ID=37 PID=24507 Type=TCP receiver
> > Process:: ID=38 PID=24508 Type=TCP receiver
> > Process:: ID=39 PID=24509 Type=TCP receiver
> > Process:: ID=40 PID=24510 Type=TCP receiver
> > Process:: ID=41 PID=24511 Type=TCP receiver
> > Process:: ID=42 PID=24512 Type=Timer handler
> > Process:: ID=43 PID=24513 Type=TCP main
> >
> > In the logs
> > Oct 3 16:22:37 ip-172-31-37-148 /usr/local/sbin/opensips[24473]:
> > WARNING:core:timer_ticker: timer task <nh-timer> already scheduled for
> > 29546470 ms (now 33307710 ms), it may overlap..
> >
> > #opensipsctl fifo get_statistics load: core: shmem:
> > load:load:: 30
> > load:load1m:: 30
> > load:load10m:: 30
> > load:load-all:: 29
> > load:load1m-all:: 29
> > load:load10m-all:: 29
> > load:load-proc-1:: 0
> > load:load1m-proc-1:: 0
> > load:load10m-proc-1:: 0
> > load:load-proc-2:: 0
> > load:load1m-proc-2:: 0
> > load:load10m-proc-2:: 0
> > load:load-proc-3:: 0
> > load:load1m-proc-3:: 0
> > load:load10m-proc-3:: 0
> > load:load-proc-4:: 0
> > load:load1m-proc-4:: 0
> > load:load10m-proc-4:: 0
> > load:load-proc-5:: 0
> > load:load1m-proc-5:: 0
> > load:load10m-proc-5:: 0
> > load:load-proc-6:: 0
> > load:load1m-proc-6:: 0
> > load:load10m-proc-6:: 0
> > load:load-proc-7:: 0
> > load:load1m-proc-7:: 0
> > load:load10m-proc-7:: 0
> > load:load-proc-8:: 0
> > load:load1m-proc-8:: 0
> > load:load10m-proc-8:: 0
> > load:load-proc-9:: 0
> > load:load1m-proc-9:: 0
> > load:load10m-proc-9:: 0
> > load:load-proc-10:: 0
> > load:load1m-proc-10:: 0
> > load:load10m-proc-10:: 0
> > load:load-proc-11:: 0
> > load:load1m-proc-11:: 0
> > load:load10m-proc-11:: 0
> > load:load-proc-12:: 0
> > load:load1m-proc-12:: 0
> > load:load10m-proc-12:: 0
> > load:load-proc-13:: 0
> > load:load1m-proc-13:: 0
> > load:load10m-proc-13:: 0
> > load:load-proc-14:: 0
> > load:load1m-proc-14:: 0
> > load:load10m-proc-14:: 0
> > load:load-proc-15:: 0
> > load:load1m-proc-15:: 0
> > load:load10m-proc-15:: 0
> > load:load-proc-16:: 0
> > load:load1m-proc-16:: 0
> > load:load10m-proc-16:: 0
> > load:load-proc-17:: 0
> > load:load1m-proc-17:: 0
> > load:load10m-proc-17:: 0
> > load:load-proc-18:: 0
> > load:load1m-proc-18:: 0
> > load:load10m-proc-18:: 0
> > load:load-proc-19:: 0
> > load:load1m-proc-19:: 0
> > load:load10m-proc-19:: 0
> > load:load-proc-20:: 0
> > load:load1m-proc-20:: 0
> > load:load10m-proc-20:: 0
> > load:load-proc-21:: 0
> > load:load1m-proc-21:: 0
> > load:load10m-proc-21:: 0
> > load:load-proc-22:: 0
> > load:load1m-proc-22:: 0
> > load:load10m-proc-22:: 0
> > load:load-proc-23:: 0
> > load:load1m-proc-23:: 0
> > load:load10m-proc-23:: 0
> > load:load-proc-24:: 100
> > load:load1m-proc-24:: 100
> > load:load10m-proc-24:: 100
> > load:load-proc-25:: 100
> > load:load1m-proc-25:: 100
> > load:load10m-proc-25:: 100
> > load:load-proc-26:: 100
> > load:load1m-proc-26:: 100
> > load:load10m-proc-26:: 100
> > load:load-proc-27:: 100
> > load:load1m-proc-27:: 100
> > load:load10m-proc-27:: 100
> > load:load-proc-28:: 100
> > load:load1m-proc-28:: 100
> > load:load10m-proc-28:: 100
> > load:load-proc-29:: 100
> > load:load1m-proc-29:: 100
> > load:load10m-proc-29:: 100
> > load:load-proc-30:: 100
> > load:load1m-proc-30:: 100
> > load:load10m-proc-30:: 100
> > load:load-proc-31:: 100
> > load:load1m-proc-31:: 100
> > load:load10m-proc-31:: 100
> > load:load-proc-32:: 100
> > load:load1m-proc-32:: 100
> > load:load10m-proc-32:: 100
> > load:load-proc-33:: 100
> > load:load1m-proc-33:: 100
> > load:load10m-proc-33:: 100
> > load:load-proc-34:: 0
> > load:load1m-proc-34:: 0
> > load:load10m-proc-34:: 0
> > load:load-proc-35:: 0
> > load:load1m-proc-35:: 0
> > load:load10m-proc-35:: 0
> > load:load-proc-36:: 100
> > load:load1m-proc-36:: 100
> > load:load10m-proc-36:: 100
> > load:load-proc-37:: 0
> > load:load1m-proc-37:: 0
> > load:load10m-proc-37:: 0
> > load:load-proc-38:: 0
> > load:load1m-proc-38:: 0
> > load:load10m-proc-38:: 0
> > load:load-proc-39:: 0
> > load:load1m-proc-39:: 0
> > load:load10m-proc-39:: 0
> > load:load-proc-40:: 0
> > load:load1m-proc-40:: 0
> > load:load10m-proc-40:: 0
> > load:load-proc-41:: 0
> > load:load1m-proc-41:: 0
> > load:load10m-proc-41:: 0
> > load:load-proc-42:: 100
> > load:load1m-proc-42:: 100
> > load:load10m-proc-42:: 100
> > load:load-proc-43:: 0
> > load:load1m-proc-43:: 0
> > load:load10m-proc-43:: 0
> > core:rcv_requests:: 50509
> > core:rcv_replies:: 31039
> > core:fwd_requests:: 59
> > core:fwd_replies:: 0
> > core:drop_requests:: 0
> > core:drop_replies:: 0
> > core:err_requests:: 0
> > core:err_replies:: 0
> > core:bad_URIs_rcvd:: 0
> > core:unsupported_methods:: 0
> > core:bad_msg_hdr:: 0
> > core:timestamp:: 33581
> > shmem:total_size:: 33554432
> > shmem:used_size:: 5929544
> > shmem:real_used_size:: 6187752
> > shmem:max_used_size:: 7568672
> > shmem:free_size:: 27366680
> > shmem:fragments:: 8137
> >
> > As OpenSIPS is not crashing, I also don't know how to get core dump here.
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From fgast+opensips at only640k.net  Fri Oct  4 08:49:48 2019
From: fgast+opensips at only640k.net (Fabian Gast)
Date: Fri, 4 Oct 2019 14:49:48 +0200 (CEST)
Subject: [OpenSIPS-Users] Dialplan Help
In-Reply-To: <CAOvQDes-Te6xp82A+B_+V97rw2F4KWgDBergOTxxXZand6gGsg@mail.gmail.com>
References: <CAOvQDesYoRb2qqBEqToMYrQCh2dxvAKaU9WJwJ0UG9C9cg5hJg@mail.gmail.com>
 <CAOvQDes-Te6xp82A+B_+V97rw2F4KWgDBergOTxxXZand6gGsg@mail.gmail.com>
Message-ID: <1658119170.43534.1570193388973.JavaMail.zimbra@only640k.net>

If you just want to remove the "+" at the beginning, you can use something like 

$avp(normalized) = $(avp(number){re.subst,/^\+//});

Fabian


> 
> On Thu, 3 Oct 2019 at 11:35, Mark Farmer < [ mailto:farmorg at gmail.com |
> farmorg at gmail.com ] > wrote:
> 
> 
> 
> Hi everyone
> I'm having a problem with my rule not matching, I'm trying to remove the + at
> the beginning of the input DID using a regex rule:
> 


From suharik71 at gmail.com  Fri Oct  4 09:37:27 2019
From: suharik71 at gmail.com (=?UTF-8?B?0JDQvdGC0L7QvSDQldGA0YjQvtCy?=)
Date: Fri, 4 Oct 2019 16:37:27 +0300
Subject: [OpenSIPS-Users] opensips 3.0.1 segfault
Message-ID: <CAMp8X_srDrrFRQdoObixN8mLARCGSY1S=t_YsF=ahBGEbAooQw@mail.gmail.com>

Hi,
I discovered this situation. my config may not be very correct, but when it
is executed, the opensips falls into segfault.

event_route[E_DLG_STATE_CHANGED] {
...

if ( get_dialogs_by_profile("reknumber", $avp(reknum),
$avp(dlg_jsons), $avp(callcount)) ) {	xlog("L_INFO", "[$param(callid)]
- advertising number  $avp(reknum) has $avp(callcount) other calls
\n");    } else {	xlog("L_INFO", "[$param(callid)] - this profile does
not have active dialogs \n");	$avp(callcount) = 0;    }

...

}

when I try to get a dialog profile which is not I get this error

ERROR:core:do_action: Failed to get fixups for command <get_dialogs_by_profile>

and after some time opensips falls in segfault

gdb /usr/sbin/opensips core.30467
GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-115.el7
Copyright (C) 2013 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-redhat-linux-gnu".
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>...
Reading symbols from /usr/sbin/opensips...Reading symbols from
/usr/sbin/opensips...(no debugging symbols found)...done.
(no debugging symbols found)...done.
[New LWP 30467]
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib64/libthread_db.so.1".
Core was generated by `/usr/sbin/opensips -DDdd -f
/etc/opensips/opensips.cfg.j2 -p python3.6 /etc/ope'.
Program terminated with signal 11, Segmentation fault.
#0  0x00007faed2906d54 in dlg_get_json_out () from
/usr/lib64/opensips/modules/dialog.so
Missing separate debuginfos, use: debuginfo-install
opensips-3.0.1.20191002.b04ffc28d-1.el7.x86_64
(gdb) bt full
#0  0x00007faed2906d54 in dlg_get_json_out () from
/usr/lib64/opensips/modules/dialog.so
No symbol table info available.
#1  0x00007faed29073df in w_get_dlg_jsons_by_profile () from
/usr/lib64/opensips/modules/dialog.so
No symbol table info available.
#2  0x000000000042f3bc in do_action ()
No symbol table info available.
#3  0x0000000000434910 in run_action_list ()
No symbol table info available.
#4  0x0000000000465b77 in eval_expr ()
No symbol table info available.
#5  0x00000000004658f5 in eval_expr ()
No symbol table info available.
#6  0x000000000042f22a in do_action ()
No symbol table info available.
#7  0x0000000000434910 in run_action_list ()
No symbol table info available.
#8  0x0000000000434bf8 in run_top_route ()
No symbol table info available.
#9  0x00007faecf1d2e89 in route_run () from
/usr/lib64/opensips/modules/event_route.so
No symbol table info available.
#10 0x00007faecf1d2ee0 in route_received () from
/usr/lib64/opensips/modules/event_route.so
No symbol table info available.
#11 0x000000000045a66b in ipc_handle_job ()
No symbol table info available.
#12 0x000000000055aad0 in handle_io ()
No symbol table info available.
#13 0x000000000055c9ff in tcp_worker_proc_loop ()
No symbol table info available.
#14 0x0000000000566f6d in tcp_start_processes ()
No symbol table info available.
#15 0x000000000041e9f4 in main ()
No symbol table info available.
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From Sunil.More at novanet.net  Sat Oct  5 15:07:23 2019
From: Sunil.More at novanet.net (Sunil More)
Date: Sun, 6 Oct 2019 00:37:23 +0530
Subject: [OpenSIPS-Users] Issue : TLS Opensips restarts as soon as it
	receives a message
Message-ID: <CAAP4-Xm1_AESAK-DsdRCiG=KJq3XBuezAbqjCueMTZCbqjqBbQ@mail.gmail.com>

Hi Team,

We have recently configured opensips to use TLS. It started fine. However
as soon as it receives any message it restarts. I am using the
following environment

1. Ubuntu 18.04
2. Opensips 3.0.0
3. OpenSSL 1.1.1  11 Sep 2018

 Kindly let us know what can be done here.

Regards,
Sunil More
Ph: 919503338275
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From razvan at opensips.org  Mon Oct  7 03:24:15 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Mon, 7 Oct 2019 10:24:15 +0300
Subject: [OpenSIPS-Users] Issue : TLS Opensips restarts as soon as it
 receives a message
In-Reply-To: <CAAP4-Xm1_AESAK-DsdRCiG=KJq3XBuezAbqjCueMTZCbqjqBbQ@mail.gmail.com>
References: <CAAP4-Xm1_AESAK-DsdRCiG=KJq3XBuezAbqjCueMTZCbqjqBbQ@mail.gmail.com>
Message-ID: <15e6530b-bae8-cb62-d329-90d4912f11cc@opensips.org>

Hi, Sunil!

Please update OpenSIPS to the latest revision.
Also, follow this[1] thread for more information about this issue.

[1] https://github.com/OpenSIPS/opensips/issues/1799

Best regards,
Răzvan

On 10/5/19 10:07 PM, Sunil More wrote:
> Hi Team,
> 
> We have recently configured opensips to use TLS. It started fine. 
> However as soon as it receives any message it restarts. I am using the 
> following environment
> 
> 1. Ubuntu 18.04
> 2. Opensips 3.0.0
> 3. OpenSSL 1.1.1  11 Sep 2018
> 
>   Kindly let us know what can be done here.
> 
> Regards,
> Sunil More
> Ph: 919503338275
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From solarmon at one-n.co.uk  Mon Oct  7 05:14:43 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 7 Oct 2019 10:14:43 +0100
Subject: [OpenSIPS-Users] Convert 183 to 180?
Message-ID: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>

Hi,

Our system connected to openSIPS is not handling 183 (with and without SDP)
very well and is non-trivial for us to try to change the behavior. If we
receive an initial 183 without SDP, it does not processed a subsequent 183
with SDP.

We would like to explore how 183 (with and without SDP) can be converted to
180 (with and without SDP) see if this can resolve our issues. Can this be
achieved and are there any examples of how this could be done?

Thank you in advance for any help.

<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
Virus-free.
www.avg.com
<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
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From Ben.Newlin at genesys.com  Mon Oct  7 08:59:13 2019
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Mon, 7 Oct 2019 12:59:13 +0000
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
Message-ID: <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>

I think you are looking for this: https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of solarmon <solarmon at one-n.co.uk>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Monday, October 7, 2019 at 5:15 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] Convert 183 to 180?

Hi,

Our system connected to openSIPS is not handling 183 (with and without SDP) very well and is non-trivial for us to try to change the behavior. If we receive an initial 183 without SDP, it does not processed a subsequent 183 with SDP.

We would like to explore how 183 (with and without SDP) can be converted to 180 (with and without SDP) see if this can resolve our issues. Can this be achieved and are there any examples of how this could be done?

Thank you in advance for any help.

[Image removed by sender.]<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
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From solarmon at one-n.co.uk  Mon Oct  7 09:23:19 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 7 Oct 2019 14:23:19 +0100
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
 <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
Message-ID: <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>

Hi,

Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same
function is available for it too:

https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status


Thanks for the tip!


<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
Virus-free.
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<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On Mon, 7 Oct 2019 at 14:00, Ben Newlin <Ben.Newlin at genesys.com> wrote:

> I think you are looking for this:
> https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of solarmon <
> solarmon at one-n.co.uk>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Monday, October 7, 2019 at 5:15 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *[OpenSIPS-Users] Convert 183 to 180?
>
>
>
> Hi,
>
>
>
> Our system connected to openSIPS is not handling 183 (with and without
> SDP) very well and is non-trivial for us to try to change the behavior. If
> we receive an initial 183 without SDP, it does not processed a subsequent
> 183 with SDP.
>
>
>
> We would like to explore how 183 (with and without SDP) can be converted
> to 180 (with and without SDP) see if this can resolve our issues. Can this
> be achieved and are there any examples of how this could be done?
>
>
>
> Thank you in advance for any help.
>
>
>
> [image: Image removed by sender.]
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
> Virus-free. www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From Frank at wtild.com  Mon Oct  7 14:18:27 2019
From: Frank at wtild.com (Frank Lee)
Date: Mon, 7 Oct 2019 11:18:27 -0700
Subject: [OpenSIPS-Users] MySQL modparam with domain in username issue
Message-ID: <002301d57d3b$9ec83100$dc589300$@wtild.com>

Hello,

 

In the config file for "Virtual DB",  what is the correct syntax if I have a
"domain" after the username, e.g.:

 

modparam("db_virtual",
"db_urls","mysql://xxx at yyy:zzz at abc.mysql.database.azure.com:3306/opensips")

 

so when I use this syntax, opensips keep thinking that the host name is
"yyy", but the host name is abc.mysql.datbase.azure, not yyy.

 

But in MySQL Azure, it requires to have the @yyy domain after the username
or I can not login correctly.

 

Thank you!

 

 

Frank 

 

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From solarmon at one-n.co.uk  Wed Oct  9 05:02:40 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Wed, 9 Oct 2019 10:02:40 +0100
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
 <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
 <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>
Message-ID: <CAOXfywNa6Vz=9fnfz-qXQJ5GS1cQde02cJvd82ExoQXnW2rhfA@mail.gmail.com>

Hi,

Had a quick look at the change_reply_status() function and I'm a bit unsure
as it suitability for what I am wanting to do.

Our setup has opensips sitting between our system and a SIP provider. The
183 is coming from SIP provider and I would like to convert this to 180
before passing it on to our system.

The documentation for change_reply_status() suggest it is used to
"Intercept a SIP reply". I am not looking to change a 'reply' to the SIP
provider, or to our system. I want to convert the 183 message from the SIP
provider and pass it on as 180 to our system.

Will this  change_reply_status() function allow us to do that?

On Mon, 7 Oct 2019 at 14:23, solarmon <solarmon at one-n.co.uk> wrote:

> Hi,
>
> Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same
> function is available for it too:
>
>
> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status
>
>
> Thanks for the tip!
>
>
>
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free.
> www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
> <#m_-6109930205533376420_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> On Mon, 7 Oct 2019 at 14:00, Ben Newlin <Ben.Newlin at genesys.com> wrote:
>
>> I think you are looking for this:
>> https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users <users-bounces at lists.opensips.org> on behalf of solarmon <
>> solarmon at one-n.co.uk>
>> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Date: *Monday, October 7, 2019 at 5:15 AM
>> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject: *[OpenSIPS-Users] Convert 183 to 180?
>>
>>
>>
>> Hi,
>>
>>
>>
>> Our system connected to openSIPS is not handling 183 (with and without
>> SDP) very well and is non-trivial for us to try to change the behavior. If
>> we receive an initial 183 without SDP, it does not processed a subsequent
>> 183 with SDP.
>>
>>
>>
>> We would like to explore how 183 (with and without SDP) can be converted
>> to 180 (with and without SDP) see if this can resolve our issues. Can this
>> be achieved and are there any examples of how this could be done?
>>
>>
>>
>> Thank you in advance for any help.
>>
>>
>>
>> [image: Image removed by sender.]
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>
>> Virus-free. www.avg.com
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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From david.villasmil.work at gmail.com  Wed Oct  9 05:42:01 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 9 Oct 2019 10:42:01 +0100
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <CAOXfywNa6Vz=9fnfz-qXQJ5GS1cQde02cJvd82ExoQXnW2rhfA@mail.gmail.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
 <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
 <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>
 <CAOXfywNa6Vz=9fnfz-qXQJ5GS1cQde02cJvd82ExoQXnW2rhfA@mail.gmail.com>
Message-ID: <CAFGRPVrfbVyoiRVsB8A6UZLO0RaAG6WARp4rS3OU8gJ_EYvepw@mail.gmail.com>

That’s exactly what it does. You might want to check for the absence of sdp
with

if(has_body("application/sdp"))


On Wed, 9 Oct 2019 at 10:03, solarmon <solarmon at one-n.co.uk> wrote:

> Hi,
>
> Had a quick look at the change_reply_status() function and I'm a bit
> unsure as it suitability for what I am wanting to do.
>
> Our setup has opensips sitting between our system and a SIP provider. The
> 183 is coming from SIP provider and I would like to convert this to 180
> before passing it on to our system.
>
> The documentation for change_reply_status() suggest it is used to
> "Intercept a SIP reply". I am not looking to change a 'reply' to the SIP
> provider, or to our system. I want to convert the 183 message from the SIP
> provider and pass it on as 180 to our system.
>
> Will this  change_reply_status() function allow us to do that?
>
> On Mon, 7 Oct 2019 at 14:23, solarmon <solarmon at one-n.co.uk> wrote:
>
>> Hi,
>>
>> Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same
>> function is available for it too:
>>
>>
>> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status
>>
>>
>> Thanks for the tip!
>>
>>
>>
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free.
>> www.avg.com
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>> <#m_1222389664913481817_m_-6109930205533376420_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
>> On Mon, 7 Oct 2019 at 14:00, Ben Newlin <Ben.Newlin at genesys.com> wrote:
>>
>>> I think you are looking for this:
>>> https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status
>>>
>>>
>>>
>>> Ben Newlin
>>>
>>>
>>>
>>> *From: *Users <users-bounces at lists.opensips.org> on behalf of solarmon <
>>> solarmon at one-n.co.uk>
>>> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
>>> *Date: *Monday, October 7, 2019 at 5:15 AM
>>> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>>> *Subject: *[OpenSIPS-Users] Convert 183 to 180?
>>>
>>>
>>>
>>> Hi,
>>>
>>>
>>>
>>> Our system connected to openSIPS is not handling 183 (with and without
>>> SDP) very well and is non-trivial for us to try to change the behavior. If
>>> we receive an initial 183 without SDP, it does not processed a subsequent
>>> 183 with SDP.
>>>
>>>
>>>
>>> We would like to explore how 183 (with and without SDP) can be converted
>>> to 180 (with and without SDP) see if this can resolve our issues. Can this
>>> be achieved and are there any examples of how this could be done?
>>>
>>>
>>>
>>> Thank you in advance for any help.
>>>
>>>
>>>
>>> [image: Image removed by sender.]
>>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>>
>>> Virus-free. www.avg.com
>>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>>
>>>
>>> _______________________________________________
>>> 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
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From Ben.Newlin at genesys.com  Wed Oct  9 09:25:48 2019
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Wed, 9 Oct 2019 13:25:48 +0000
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <CAFGRPVrfbVyoiRVsB8A6UZLO0RaAG6WARp4rS3OU8gJ_EYvepw@mail.gmail.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
 <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
 <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>
 <CAOXfywNa6Vz=9fnfz-qXQJ5GS1cQde02cJvd82ExoQXnW2rhfA@mail.gmail.com>
 <CAFGRPVrfbVyoiRVsB8A6UZLO0RaAG6WARp4rS3OU8gJ_EYvepw@mail.gmail.com>
Message-ID: <CACE5756-B509-4E91-A6A7-B1D14FF6538E@genesys.com>

You are trying to change a reply because the 183 response is a reply to the INVITE. It is not the final reply, but it is still a reply. In fact, the change_reply_status  function explicitly does not allow you to change final replies, only provisional replies like 1xx.

You need to use an onreply_route [1], either the global one or a transaction one using t_on_reply [2], and the in that reply route you will check if the response is 183 (t_check_status [3]) and if it is use change_reply_status to change it to a 180.

[1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc4
[2] https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_on_reply
[3] https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_check_status


Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of David Villasmil <david.villasmil.work at gmail.com>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Wednesday, October 9, 2019 at 5:42 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Convert 183 to 180?

That’s exactly what it does. You might want to check for the absence of sdp with


if(has_body("application/sdp"))

On Wed, 9 Oct 2019 at 10:03, solarmon <solarmon at one-n.co.uk<mailto:solarmon at one-n.co.uk>> wrote:
Hi,

Had a quick look at the change_reply_status() function and I'm a bit unsure as it suitability for what I am wanting to do.

Our setup has opensips sitting between our system and a SIP provider. The 183 is coming from SIP provider and I would like to convert this to 180 before passing it on to our system.

The documentation for change_reply_status() suggest it is used to "Intercept a SIP reply". I am not looking to change a 'reply' to the SIP provider, or to our system. I want to convert the 183 message from the SIP provider and pass it on as 180 to our system.

Will this  change_reply_status() function allow us to do that?

On Mon, 7 Oct 2019 at 14:23, solarmon <solarmon at one-n.co.uk<mailto:solarmon at one-n.co.uk>> wrote:
Hi,

Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same function is available for it too:

https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status<https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status>

Thanks for the tip!


[Image removed by sender.]<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
Virus-free. www.avg.com<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>

On Mon, 7 Oct 2019 at 14:00, Ben Newlin <Ben.Newlin at genesys.com<mailto:Ben.Newlin at genesys.com>> wrote:
I think you are looking for this: https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status<https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status>

Ben Newlin

From: Users <users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>> on behalf of solarmon <solarmon at one-n.co.uk<mailto:solarmon at one-n.co.uk>>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Date: Monday, October 7, 2019 at 5:15 AM
To: OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Subject: [OpenSIPS-Users] Convert 183 to 180?

Hi,

Our system connected to openSIPS is not handling 183 (with and without SDP) very well and is non-trivial for us to try to change the behavior. If we receive an initial 183 without SDP, it does not processed a subsequent 183 with SDP.

We would like to explore how 183 (with and without SDP) can be converted to 180 (with and without SDP) see if this can resolve our issues. Can this be achieved and are there any examples of how this could be done?

Thank you in advance for any help.

Error! Filename not specified.<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
Virus-free. www.avg.com<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>

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

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
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From david.villasmil.work at gmail.com  Wed Oct  9 09:31:45 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 9 Oct 2019 14:31:45 +0100
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <CACE5756-B509-4E91-A6A7-B1D14FF6538E@genesys.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
 <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
 <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>
 <CAOXfywNa6Vz=9fnfz-qXQJ5GS1cQde02cJvd82ExoQXnW2rhfA@mail.gmail.com>
 <CAFGRPVrfbVyoiRVsB8A6UZLO0RaAG6WARp4rS3OU8gJ_EYvepw@mail.gmail.com>
 <CACE5756-B509-4E91-A6A7-B1D14FF6538E@genesys.com>
Message-ID: <CAFGRPVrFC3fNRqp0KWJ02xeumoti7rmjot069BO9_koD9AweMw@mail.gmail.com>

I think he’s just confused thinking that would change the reply TO THE
provider. But this is not the case, the reply 183 comes FROM the provider
and by “intercepting” it and changing it to 180, it’s the 180 what goes to
the caller.

On Wed, 9 Oct 2019 at 14:26, Ben Newlin <Ben.Newlin at genesys.com> wrote:

> You are trying to change a reply because the 183 response is a reply to
> the INVITE. It is not the final reply, but it is still a reply. In fact,
> the change_reply_status  function explicitly does not allow you to change
> final replies, only provisional replies like 1xx.
>
>
>
> You need to use an onreply_route [1], either the global one or a
> transaction one using t_on_reply [2], and the in that reply route you will
> check if the response is 183 (t_check_status [3]) and if it is use
> change_reply_status to change it to a 180.
>
>
>
> [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc4
>
> [2] https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_on_reply
>
> [3]
> https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_check_status
>
>
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of David
> Villasmil <david.villasmil.work at gmail.com>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Wednesday, October 9, 2019 at 5:42 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] Convert 183 to 180?
>
>
>
> That’s exactly what it does. You might want to check for the absence of
> sdp with
>
>
>
> if(has_body("application/sdp"))
>
>
>
> On Wed, 9 Oct 2019 at 10:03, solarmon <solarmon at one-n.co.uk> wrote:
>
> Hi,
>
>
>
> Had a quick look at the change_reply_status() function and I'm a bit
> unsure as it suitability for what I am wanting to do.
>
>
>
> Our setup has opensips sitting between our system and a SIP provider. The
> 183 is coming from SIP provider and I would like to convert this to 180
> before passing it on to our system.
>
>
>
> The documentation for change_reply_status() suggest it is used to
> "Intercept a SIP reply". I am not looking to change a 'reply' to the SIP
> provider, or to our system. I want to convert the 183 message from the SIP
> provider and pass it on as 180 to our system.
>
>
>
> Will this  change_reply_status() function allow us to do that?
>
>
>
> On Mon, 7 Oct 2019 at 14:23, solarmon <solarmon at one-n.co.uk> wrote:
>
> Hi,
>
>
>
> Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same
> function is available for it too:
>
>
>
>
> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status
>
>
>
>
> Thanks for the tip!
>
>
>
>
>
> [image: Image removed by sender.]
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
> Virus-free. www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
>
>
> On Mon, 7 Oct 2019 at 14:00, Ben Newlin <Ben.Newlin at genesys.com> wrote:
>
> I think you are looking for this:
> https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of solarmon <
> solarmon at one-n.co.uk>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Monday, October 7, 2019 at 5:15 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *[OpenSIPS-Users] Convert 183 to 180?
>
>
>
> Hi,
>
>
>
> Our system connected to openSIPS is not handling 183 (with and without
> SDP) very well and is non-trivial for us to try to change the behavior. If
> we receive an initial 183 without SDP, it does not processed a subsequent
> 183 with SDP.
>
>
>
> We would like to explore how 183 (with and without SDP) can be converted
> to 180 (with and without SDP) see if this can resolve our issues. Can this
> be achieved and are there any examples of how this could be done?
>
>
>
> Thank you in advance for any help.
>
>
>
> *Error! Filename not specified.*
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
> Virus-free. www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
>
>
> _______________________________________________
> 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
>
> --
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
> phone: +34669448337
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From arda.tekin at afiniti.com  Wed Oct  9 16:46:02 2019
From: arda.tekin at afiniti.com (Tekin, Arda)
Date: Wed, 9 Oct 2019 20:46:02 +0000
Subject: [OpenSIPS-Users] Adding custom headers when sending REFER with
	t_new_request function
Message-ID: <6f426af0254f465b9b6766cede3408ac@afiniti.com>

Hello,

I am trying to initiate and send a REFER message from OpenSIPS after call hold signaling completed.

t_new_request function of tm module allows to send this message immediately, this is how I used the function,

t_new_request("REFER", "sip:moh at 172.16.30.166", "1001 sip:1001 at 172.16.30.164", "1000 sip:1000 at 172.16.30.164", "text/plain Hello Alice!");

but I need to add some extra headers into this message before sending it. As far as I understand the last parameter (ctx) of function is used for that purpose.  Could you please explain how I can use last parameter in order to add multiple headers into message.

Thank you
Arda
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From fgast+opensips at only640k.net  Thu Oct 10 02:21:49 2019
From: fgast+opensips at only640k.net (Fabian Gast)
Date: Thu, 10 Oct 2019 08:21:49 +0200 (CEST)
Subject: [OpenSIPS-Users] Adding custom headers when sending REFER
	with	t_new_request function
In-Reply-To: <6f426af0254f465b9b6766cede3408ac@afiniti.com>
References: <6f426af0254f465b9b6766cede3408ac@afiniti.com>
Message-ID: <2066719973.35038.1570688509232.JavaMail.zimbra@only640k.net>

Hi,

The ctx parameter never worked for me, i allways got a
ERROR:tm:w_t_new_request: failed to add ctx AVP, ignorring...
for this.

I use something like

t_new_request("OPTIONS","sip:ping at TODOMAIN:5061;transport=tls","sip:pong at DOMAIN","sip:ping at DOMAIN");

local_route {
        if (is_method("OPTIONS") && $rd == "TODOMAIN") {        
                        append_hf("X-MY-HEADER: asdfasdf\n\n");
                
        }
}

maybe this helps you a little bit..

Fabian

----- Ursprüngliche Mail -----
> Von: "Tekin, Arda" <arda.tekin at afiniti.com>
> An: "OpenSIPS users mailling list" <users at lists.opensips.org>
> Gesendet: Mittwoch, 9. Oktober 2019 22:46:02
> Betreff: [OpenSIPS-Users] Adding custom headers when sending REFER with	t_new_request function

> Hello,
> 
> 
> 
> I am trying to initiate and send a REFER message from OpenSIPS after call hold
> signaling completed.
> 
> 
> 
> t_new_request function of tm module allows to send this message immediately,
> this is how I used the function,
> 
> 
> 
> t_new_request("REFER", "sip:moh at 172.16.30.166", "1001 sip:1001 at 172.16.30.164",
> "1000 sip:1000 at 172.16.30.164", "text/plain Hello Alice!");
> 
> 
> 
> but I need to add some extra headers into this message before sending it. As far
> as I understand the last parameter (ctx) of function is used for that purpose.
> Could you please explain how I can use last parameter in order to add multiple
> headers into message.


From johan at democon.be  Thu Oct 10 09:55:54 2019
From: johan at democon.be (johan)
Date: Thu, 10 Oct 2019 15:55:54 +0200
Subject: [OpenSIPS-Users] potential memory leak
Message-ID: <734fadf6-7b7a-0abd-e568-a784a2ab2ccc@democon.be>


     Hi,


Since I have added proto_smpp to my script, I see the following error 
after x days in my log.

Oct 10 13:51:56 hendrix /data/opensips/sbin/opensips[6857]: 
ERROR:proto_smpp:build_enquire_link_request: malloc error for payload
Oct 10 13:51:56 hendrix /data/opensips/sbin/opensips[6857]: 
ERROR:proto_smpp:send_enquire_link_request: error creating 
enquire_link_sm request
Oct 10 13:52:12 hendrix /data/opensips/sbin/opensips[6857]: 
ERROR:core:fm_malloc: not enough free pkg memory (136 bytes left, need 
2184), please increase the "-M" command line parameter!
Oct 10 13:52:12 hendrix /data/opensips/sbin/opensips[6857]: 
ERROR:core:receive_msg: no pkg mem left for sip_msg
Oct 10 13:52:14 hendrix /data/opensips/sbin/opensips[6857]: 
ERROR:core:fm_malloc: not enough free pkg memory (136 bytes left, need 
2184), please increase the "-M" command line parameter!


I have never had this before.

Can it be that there is a memory leak in proto_smpp ?

Can you please indicate how I need to troubleshoot this ?


BR,



From solarmon at one-n.co.uk  Thu Oct 10 10:18:30 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Thu, 10 Oct 2019 15:18:30 +0100
Subject: [OpenSIPS-Users] Convert 183 to 180?
In-Reply-To: <CAFGRPVrFC3fNRqp0KWJ02xeumoti7rmjot069BO9_koD9AweMw@mail.gmail.com>
References: <CAOXfywOjxM7PWSe=eqiS7ZtKnJvTa1MgbCEmg-MWagXEt=i2TA@mail.gmail.com>
 <42980DE7-8770-43D7-870F-FF2C77E05587@genesys.com>
 <CAOXfywPeDZLpoV_S4QSU+dcULfCYrYhibCF8qeFjaURgCQKGHw@mail.gmail.com>
 <CAOXfywNa6Vz=9fnfz-qXQJ5GS1cQde02cJvd82ExoQXnW2rhfA@mail.gmail.com>
 <CAFGRPVrfbVyoiRVsB8A6UZLO0RaAG6WARp4rS3OU8gJ_EYvepw@mail.gmail.com>
 <CACE5756-B509-4E91-A6A7-B1D14FF6538E@genesys.com>
 <CAFGRPVrFC3fNRqp0KWJ02xeumoti7rmjot069BO9_koD9AweMw@mail.gmail.com>
Message-ID: <CAOXfywMNDz6vOTHG0J7jOJLG0FdDHZTEsqyLBs6EdgByYVpEAg@mail.gmail.com>

Thanks all.

Apologies for my confusion and misunderstanding.

I have now tried it, and it does indeed work in the way I want.

On Wed, 9 Oct 2019 at 14:33, David Villasmil <david.villasmil.work at gmail.com>
wrote:

> I think he’s just confused thinking that would change the reply TO THE
> provider. But this is not the case, the reply 183 comes FROM the provider
> and by “intercepting” it and changing it to 180, it’s the 180 what goes to
> the caller.
>
> On Wed, 9 Oct 2019 at 14:26, Ben Newlin <Ben.Newlin at genesys.com> wrote:
>
>> You are trying to change a reply because the 183 response is a reply to
>> the INVITE. It is not the final reply, but it is still a reply. In fact,
>> the change_reply_status  function explicitly does not allow you to change
>> final replies, only provisional replies like 1xx.
>>
>>
>>
>> You need to use an onreply_route [1], either the global one or a
>> transaction one using t_on_reply [2], and the in that reply route you will
>> check if the response is 183 (t_check_status [3]) and if it is use
>> change_reply_status to change it to a 180.
>>
>>
>>
>> [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc4
>>
>> [2] https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_on_reply
>>
>> [3]
>> https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_check_status
>>
>>
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users <users-bounces at lists.opensips.org> on behalf of David
>> Villasmil <david.villasmil.work at gmail.com>
>> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Date: *Wednesday, October 9, 2019 at 5:42 AM
>> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject: *Re: [OpenSIPS-Users] Convert 183 to 180?
>>
>>
>>
>> That’s exactly what it does. You might want to check for the absence of
>> sdp with
>>
>>
>>
>> if(has_body("application/sdp"))
>>
>>
>>
>> On Wed, 9 Oct 2019 at 10:03, solarmon <solarmon at one-n.co.uk> wrote:
>>
>> Hi,
>>
>>
>>
>> Had a quick look at the change_reply_status() function and I'm a bit
>> unsure as it suitability for what I am wanting to do.
>>
>>
>>
>> Our setup has opensips sitting between our system and a SIP provider. The
>> 183 is coming from SIP provider and I would like to convert this to 180
>> before passing it on to our system.
>>
>>
>>
>> The documentation for change_reply_status() suggest it is used to
>> "Intercept a SIP reply". I am not looking to change a 'reply' to the SIP
>> provider, or to our system. I want to convert the 183 message from the SIP
>> provider and pass it on as 180 to our system.
>>
>>
>>
>> Will this  change_reply_status() function allow us to do that?
>>
>>
>>
>> On Mon, 7 Oct 2019 at 14:23, solarmon <solarmon at one-n.co.uk> wrote:
>>
>> Hi,
>>
>>
>>
>> Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same
>> function is available for it too:
>>
>>
>>
>>
>> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#change_reply_status
>>
>>
>>
>>
>> Thanks for the tip!
>>
>>
>>
>>
>>
>> [image: Image removed by sender.]
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>
>> Virus-free. www.avg.com
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>
>>
>>
>> On Mon, 7 Oct 2019 at 14:00, Ben Newlin <Ben.Newlin at genesys.com> wrote:
>>
>> I think you are looking for this:
>> https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users <users-bounces at lists.opensips.org> on behalf of solarmon <
>> solarmon at one-n.co.uk>
>> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Date: *Monday, October 7, 2019 at 5:15 AM
>> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject: *[OpenSIPS-Users] Convert 183 to 180?
>>
>>
>>
>> Hi,
>>
>>
>>
>> Our system connected to openSIPS is not handling 183 (with and without
>> SDP) very well and is non-trivial for us to try to change the behavior. If
>> we receive an initial 183 without SDP, it does not processed a subsequent
>> 183 with SDP.
>>
>>
>>
>> We would like to explore how 183 (with and without SDP) can be converted
>> to 180 (with and without SDP) see if this can resolve our issues. Can this
>> be achieved and are there any examples of how this could be done?
>>
>>
>>
>> Thank you in advance for any help.
>>
>>
>>
>> *Error! Filename not specified.*
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>
>> Virus-free. www.avg.com
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>> --
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.work at gmail.com
>>
>> phone: +34669448337
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From arda.tekin at afiniti.com  Thu Oct 10 12:07:47 2019
From: arda.tekin at afiniti.com (Tekin, Arda)
Date: Thu, 10 Oct 2019 16:07:47 +0000
Subject: [OpenSIPS-Users] Adding custom headers when sending
	REFER	with	t_new_request function
In-Reply-To: <2066719973.35038.1570688509232.JavaMail.zimbra@only640k.net>
References: <6f426af0254f465b9b6766cede3408ac@afiniti.com>
 <2066719973.35038.1570688509232.JavaMail.zimbra@only640k.net>
Message-ID: <db158dd2c0014fa889879cbcb2f39800@afiniti.com>

Thank you very much Fabian, your solution works pretty well. Now I am able to add headers by this way. 

I would be grateful if developers can also explain us how we can use ctx for t_new_request function. 

Kind regards,
Arda Tekin 

-----Original Message-----
From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Fabian Gast
Sent: Thursday, October 10, 2019 9:22 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Adding custom headers when sending REFER with t_new_request function

Attention: This email was sent from someone outside of Afiniti. Always use caution when opening attachments, clicking links from unknown senders or when receiving unexpected emails.



Hi,

The ctx parameter never worked for me, i allways got a
ERROR:tm:w_t_new_request: failed to add ctx AVP, ignorring...
for this.

I use something like

t_new_request("OPTIONS","sip:ping at TODOMAIN:5061;transport=tls","sip:pong at DOMAIN","sip:ping at DOMAIN");

local_route {
        if (is_method("OPTIONS") && $rd == "TODOMAIN") {
                        append_hf("X-MY-HEADER: asdfasdf\n\n");

        }
}

maybe this helps you a little bit..

Fabian

----- Ursprüngliche Mail -----
> Von: "Tekin, Arda" <arda.tekin at afiniti.com>
> An: "OpenSIPS users mailling list" <users at lists.opensips.org>
> Gesendet: Mittwoch, 9. Oktober 2019 22:46:02
> Betreff: [OpenSIPS-Users] Adding custom headers when sending REFER with       t_new_request function

> Hello,
>
>
>
> I am trying to initiate and send a REFER message from OpenSIPS after 
> call hold signaling completed.
>
>
>
> t_new_request function of tm module allows to send this message 
> immediately, this is how I used the function,
>
>
>
> t_new_request("REFER", "sip:moh at 172.16.30.166", "1001 
> sip:1001 at 172.16.30.164",
> "1000 sip:1000 at 172.16.30.164", "text/plain Hello Alice!");
>
>
>
> but I need to add some extra headers into this message before sending 
> it. As far as I understand the last parameter (ctx) of function is used for that purpose.
> Could you please explain how I can use last parameter in order to add 
> multiple headers into message.

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

From z at startit.ru  Thu Oct 10 15:44:57 2019
From: z at startit.ru (zzz)
Date: Thu, 10 Oct 2019 19:44:57 +0000
Subject: [OpenSIPS-Users] DB migration
Message-ID: <3fb8ff83e5b544ca9f85d8fc9ead6d0b@startit.ru>

Hello,
What would be the best way to migrate Opensips DB 2.4.x to 3.0.1 ?
I suppose the doc on the web https://www.opensips.org/Documentation/Migration-2-4-0-to-3-0-0
is wrong since it says there should be used
opensipsdbctl migrate opensips_2_4 opensips_3_0

best regards,
Yury
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From johan at democon.be  Fri Oct 11 02:07:57 2019
From: johan at democon.be (Johan De Clercq)
Date: Fri, 11 Oct 2019 06:07:57 +0000
Subject: [OpenSIPS-Users] DB migration
In-Reply-To: <3fb8ff83e5b544ca9f85d8fc9ead6d0b@startit.ru>
References: <3fb8ff83e5b544ca9f85d8fc9ead6d0b@startit.ru>
Message-ID: <DB7PR06MB495163FB90FDD3E391D233D9A7970@DB7PR06MB4951.eurprd06.prod.outlook.com>

THe doc is correct.   Do note that there are script changes

Outlook voor iOS<https://aka.ms/o0ukef> downloaden

________________________________
Van: Users <users-bounces at lists.opensips.org> namens zzz <z at startit.ru>
Verzonden: donderdag, oktober 10, 2019 9:47 PM
Aan: OpenSIPS users mailling list
Onderwerp: [OpenSIPS-Users] DB migration

Hello,
What would be the best way to migrate Opensips DB 2.4.x to 3.0.1 ?
I suppose the doc on the web https://www.opensips.org/Documentation/Migration-2-4-0-to-3-0-0
is wrong since it says there should be used
opensipsdbctl migrate opensips_2_4 opensips_3_0

best regards,
Yury
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From liviu at opensips.org  Fri Oct 11 02:11:18 2019
From: liviu at opensips.org (Liviu Chircu)
Date: Fri, 11 Oct 2019 09:11:18 +0300
Subject: [OpenSIPS-Users] DB migration
In-Reply-To: <3fb8ff83e5b544ca9f85d8fc9ead6d0b@startit.ru>
References: <3fb8ff83e5b544ca9f85d8fc9ead6d0b@startit.ru>
Message-ID: <4415168e-c1ba-2466-ac70-f096ef7bc274@opensips.org>

On 10.10.2019 22:44, zzz wrote:
>
> Hello,
>
> What would be the best way to migrate Opensips DB 2.4.x to 3.0.1 ?
>
> I suppose the doc on the web 
> https://www.opensips.org/Documentation/Migration-2-4-0-to-3-0-0
>
> is wrong since it says there should be used
>
> opensipsdbctl migrate opensips_2_4 opensips_3_0
>
Thank you for the tip!  I just fixed the docs.  The DB migration logic 
has been duplicated
into the opensips-cli now.

Cheers,

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

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From callum.guy at x-on.co.uk  Mon Oct 14 05:50:59 2019
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Mon, 14 Oct 2019 10:50:59 +0100
Subject: [OpenSIPS-Users] Presence behind NAT
Message-ID: <CAFjCFz=4uMqHjBVC_yyUMFFYa1Jkwh1tBBcdtp6uqNa_E4GwAQ@mail.gmail.com>

Hi All,

Hopefully a simple question related to the behaviour of the presence module
on my registrar.

I have configured OpenSIPs to enable dialoginfo updates against specific
user agents. When the user agent places/receives a call the PUBLISH event
is generated correctly. This resolves a UA at a locally served domain and
pushes the request out to the public address of the local server.  In my
environment this server is behind NAT and we end up seeing requests being
generated and sent from the internal interface to the public address of the
sending server which seems wrong, at at least inefficient.

Is anyone able to suggest an improvement to my configuration, perhaps I can
improve domain awareness somehow? Is this the correct behaviour?

Many thanks,

Callum

----

Relevant configuration (running OpenSIPs 3.0.1):

loadmodule "presence.so"
loadmodule "presence_dialoginfo.so"
loadmodule "pua.so"
loadmodule "pua_dialoginfo.so"

modparam("presence", "server_address", "sip:*public-ip*:5060")
modparam("presence", "mix_dialog_presence", 1)
modparam("presence", "fallback2db", 1)
modparam("presence", "max_expires_subscribe", 14700)
modparam("presence", "max_expires_publish", 14700)

...

# Fired when placing a call
route[ENABLE_BLF_CALLER] {
    xlog("L_INFO", "INFO: Enabling presence for calling party ($fu)
ID:$ci\n");
    dialoginfo_set("A");
}

# Fired when receiving a call
route[ENABLE_BLF_CALLEE] {
    xlog("L_INFO", "INFO: Enabling presence for called party ($ru)
ID:$ci\n");
    dialoginfo_set("B");
}

---

Example transmission:

192.168.153.210:5060 -> PUBLIC-IP(of example.com):5060

PUBLISH sip:12345 at example.com SIP/2.0
Via: SIP/2.0/UDP 192.168.153.210:5060;branch=z9hG4bK4316.2ff8a84.0
To: sip:12345 at example.com
From: <sip:12345 at example.com>;tag=260776ec80eaeff43e316f7930643583-5872
CSeq: 10 PUBLISH
Call-ID: 0168d45862da3b23-189847 at 192.168.153.210
Max-Forwards: 70
Content-Length: 643
User-Agent: me
Event: dialog
Expires: 301
Content-Type: application/dialog-info+xml

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" state="partial"
entity="sip:12345 at example.com"><dialog
id="1ca15ca3-215b-4d9d-b401-770408889a6a_I_3150144_A0_5da442c9a33f9.0"
call-id="1ca15ca3-215b-4d9d-b401-770408889a
_I_3150144_A0_5da442c9a33f9" local-tag="1323791516"
remote-tag="UvKaaK3vUvega"
direction="recipient"><state>early</state><remote><identity>
sip:01728726500 at 192.168.153.226</identity><target uri="
sip:01728726500 at 192.168.153.226"/></remote><
cal><identity>sip:12345 at example.com</identity><target uri="
sip:12345 at example.com"/></local></dialog></dialog-info>

-- 





*0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   ** 
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From rob.dyck at telus.net  Mon Oct 14 22:32:18 2019
From: rob.dyck at telus.net (Robert Dyck)
Date: Mon, 14 Oct 2019 19:32:18 -0700
Subject: [OpenSIPS-Users] Issue with opensips-3.0.1 using dbtext
Message-ID: <2910759.MZ9TNUCGKd@blacky.mylan>

I am test driving 3.0.1. Using menuconfig I compiled the core, the default modules and extra 
modules mysql, presence, presence_xml and xcap.

Using menuconfig I generated a residential script with auth and presence. I want to use db_text 
with this minimal installation. I tweaked the configuration to load the db_text module and the 
db_url. The configuration appears to be correct as far as syntax is concerned. I recycled the 
"subscriber" table from a working 2.4.5 installation.

Usrloc does not initialize at runtime. Where have I gone wrong?

modparam("usrloc", "working_mode_preset", "single-instance-sql-write-back") 

modparam("auth_db", "db_url", 


Oct 14 19:12:07 [16761] DBG:core:init_mod: register MI for db_text
Oct 14 19:12:07 [16761] DBG:core:init_mod: initializing module usrloc
Oct 14 19:12:07 [16761] DBG:usrloc:mod_init: initializing
Oct 14 19:12:07 [16761] DBG:usrloc:check_runtime_config: ul config: db_mode=-1, 
cluster_mode=0, rrp=1, sql_wm=2
Oct 14 19:12:07 [16761] INFO:usrloc:ul_init_locks: locks array size 512
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_bind_api> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:db_bind_mod: using export interface to bind db_textdb
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_use_table> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_init> in module db_textdb not found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_close> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_query> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_fetch_result> in module db_textdb 
not found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_raw_query> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_free_result> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_insert> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_delete> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_update> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_replace> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_last_inserted_id> in module 
db_textdb not found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_insert_update> in module db_textdb 
not found
Oct 14 19:12:07 [16761] ERROR:core:db_check_api: module db_textdb does not export 
db_use_table function
Oct 14 19:12:07 [16761] ERROR:usrloc:mod_init: failed to bind database module
Oct 14 19:12:07 [16761] ERROR:core:init_mod: failed to initialize module usrloc
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From todd at firestreamcloud.com  Tue Oct 15 13:33:30 2019
From: todd at firestreamcloud.com (Todd Routhier)
Date: Tue, 15 Oct 2019 12:33:30 -0500
Subject: [OpenSIPS-Users] MI command failed with code 406 (Unexpected
	method)
In-Reply-To: <4aee4d70-9445-e591-cbce-1074ad5b5775@opensips.org>
References: <CA+za3LL=hRTK=2M6Hp0o46Ldu2w_bXbzoJMP7vYRxWT6oBx1nw@mail.gmail.com>
 <4aee4d70-9445-e591-cbce-1074ad5b5775@opensips.org>
Message-ID: <CA+za3LKd4fC99bdJwQDdq44VKggHVC+v13iUVknFMok7YejnPw@mail.gmail.com>

That's what it was, a version mis-match.

I intended to follow up with exact versions and more info but I never saw
my post hit the list. I just noticed it today, I guess there was some sort
of delay in my first post to the list showing up.

In any case, thanks for the help, OpenSIPs CP is now working correctly.

Regards,


Todd

On Fri, Oct 4, 2019 at 1:56 AM Răzvan Crainea <razvan at opensips.org> wrote:

> If you're using the last year release (2.4), this will not work with the
> latest OpenSIPS CP (8.3.0), as this is only meant to work with OpenSIPS
> 3.0. For OpenSIPS 2.4 you need to use OpenSIPS Control Panel version 7.2.3.
> So please provide us the exact versions you're using for both software
> to provide you more details.
>
> Best regards,
> Răzvan
>
> On 10/3/19 8:57 PM, Todd Routhier wrote:
> > I installed a recent version of OpenSIPs and I think the latest version
> > of OpenSIPs Control Panel. I will follow up with exact versions in just
> > a bit.
> >
> > Basic functions such as adding/removing users work but many other
> > functions such as clicking the little telephone icon in the user's list
> > to see contacts registered to the SIP account, result in the error
> >
> > MI command failed with code 406 (Unexpected method)
> >
> > I have checked the box configuration and believe I have it set correctly
> > but, obviously I am missing something. Have googled my brains out, read
> > the book which seems a bit out-dated and I am at the end of my rope, out
> > of ideas.
> >
> > Thought I would post here and hope the experts can offer some advice
> > and/or guidance. I will follow this post up with some more specifics
> > such as version info and some of my config info.
> >
> > Thanks in advance for any assistance you can provide.
> >
> > --Todd
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From rob.dyck at telus.net  Tue Oct 15 13:54:15 2019
From: rob.dyck at telus.net (Robert Dyck)
Date: Tue, 15 Oct 2019 10:54:15 -0700
Subject: [OpenSIPS-Users] Issue with opensips-3.0.1 using dbtext
In-Reply-To: <2910759.MZ9TNUCGKd@blacky.mylan>
References: <2910759.MZ9TNUCGKd@blacky.mylan>
Message-ID: <4390023.QrvvnPI593@blacky.mylan>

Never mind. A stupid mistake on my part. I should have just copied from my working 2.4.5 
instead of relying on faulty memory.

On Monday, October 14, 2019 7:32:18 P.M. PDT Robert Dyck wrote:


I am test driving 3.0.1. Using menuconfig I compiled the core, the default modules and extra 
modules mysql, presence, presence_xml and xcap.

Using menuconfig I generated a residential script with auth and presence. I want to use db_text 
with this minimal installation. I tweaked the configuration to load the db_text module and the 
db_url. The configuration appears to be correct as far as syntax is concerned. I recycled the 
"subscriber" table from a working 2.4.5 installation.

Usrloc does not initialize at runtime. Where have I gone wrong?

modparam("usrloc", "working_mode_preset", "single-instance-sql-write-back") 

modparam("auth_db", "db_url", 


Oct 14 19:12:07 [16761] DBG:core:init_mod: register MI for db_text
Oct 14 19:12:07 [16761] DBG:core:init_mod: initializing module usrloc
Oct 14 19:12:07 [16761] DBG:usrloc:mod_init: initializing
Oct 14 19:12:07 [16761] DBG:usrloc:check_runtime_config: ul config: db_mode=-1, 
cluster_mode=0, rrp=1, sql_wm=2
Oct 14 19:12:07 [16761] INFO:usrloc:ul_init_locks: locks array size 512
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_bind_api> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:db_bind_mod: using export interface to bind db_textdb
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_use_table> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_init> in module db_textdb not found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_close> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_query> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_fetch_result> in module db_textdb 
not found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_raw_query> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_free_result> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_insert> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_delete> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_update> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_replace> in module db_textdb not 
found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_last_inserted_id> in module 
db_textdb not found
Oct 14 19:12:07 [16761] DBG:core:find_mod_export: <db_insert_update> in module db_textdb 
not found
Oct 14 19:12:07 [16761] ERROR:core:db_check_api: module db_textdb does not export 
db_use_table function
Oct 14 19:12:07 [16761] ERROR:usrloc:mod_init: failed to bind database module
Oct 14 19:12:07 [16761] ERROR:core:init_mod: failed to initialize module usrloc


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From todd at firestreamcloud.com  Tue Oct 15 14:01:50 2019
From: todd at firestreamcloud.com (Todd Routhier)
Date: Tue, 15 Oct 2019 13:01:50 -0500
Subject: [OpenSIPS-Users] OpenSIPs as Registration server in front of
	Asterisk
Message-ID: <CA+za3LL0_eQWYWqseQUZYdhH6c3aUdb_gENAGofyOKcZ4Ean5A@mail.gmail.com>

Problem: Calls from PSTN provider > Asterisk > OpenSIPs > SIP Endpoint have
intermittent audio issues. See below for details.

I am a long time Asterisk user but extremely new to OpenSIPs.

We are in the process of a migration from an older Asterisk server to a
newer version along with some other changes.

First order of business is for us to offload all registrations from our
current 1.8.x Asterisk server to OpenSIPs 2.4.6.

We have a setup that seems to be mostly working but intermittent audio
issues are what we are trying to eliminate.

When I say intermittent, audio seems to work for a particular end point in
certain situations or it doesn't. For example, we have some end
points which have no audio at all such as my personal soft-phone. I can't
get audio on any of three different soft-phones on my laptop, no audio in
either direction. But, I have a Grandstream phone on the same LAN which
works perfectly every time, on every call.

I have other end points which are Grandstream phones with perfectly working
audio in both directions, all the time, consistently.

I have other Grandstream end points which work for the same type of call
every time, with audio in both directions but the same phone has no audio
on slightly different types of calls (hard to explain what I mean by "types
of calls").

Ideally, we would not care about this working with Asterisk 1.8.x since we
are moving away from it but it's important for it to work as part of our
transition/migration.

I had horrible audio issues at first were it was hardly working at all or
one way audio consistently. I fixed this by setting nat=yes in the sip.conf
for the context pointing to the OpenSIPs server. I couldn't understand why
this fixed it since the OpenSIPs server and the Asterisk server both have
static IP's and are NOT behind any NAT of any sort. Only the end points
registered to OpenSIPs are behind end points.

Still I noticed that Asterisk was trying to send calls to the LAN IP of the
end points, so I tested nat=yes and it fixed most of the audio issues with
only the issues outlined above remaining.

My next steps are to see if I have good audio if I push calls to the newer
Asterisk server then to the end points registered to the OpenSIPs server.
Even if that works, it does not solve my current need to make this work
with Asterisk 1.8.x at least until the migration is complete.

Thanks in advance for any assistance with this.

Regards,

Todd
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From kurgan-rus at inbox.ru  Wed Oct 16 00:40:12 2019
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Wed, 16 Oct 2019 07:40:12 +0300
Subject: [OpenSIPS-Users] =?utf-8?q?Clustering_2=2E4_and_3=2E0?=
Message-ID: <1571200812.18647143@f116.i.mail.ru>


Hi list,
 
is it a good/bad idea to create a cluster of 2.4 and 3.0 versions?
Or it’s of vital importance to have the same version of instances in the cluster?
 
The idea is to create a federated user location cluster. [1]
 
[1]  https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation
 
-----------------------------------------------
BR, Alexey
http://alexeyka.zantsev.com/
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From razvan at opensips.org  Wed Oct 16 03:04:53 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 16 Oct 2019 10:04:53 +0300
Subject: [OpenSIPS-Users] OpenSIPs as Registration server in front of
 Asterisk
In-Reply-To: <CA+za3LL0_eQWYWqseQUZYdhH6c3aUdb_gENAGofyOKcZ4Ean5A@mail.gmail.com>
References: <CA+za3LL0_eQWYWqseQUZYdhH6c3aUdb_gENAGofyOKcZ4Ean5A@mail.gmail.com>
Message-ID: <754c40b0-d1ff-7a23-c425-69cf1fa17fc6@opensips.org>

Hi, Todd!

Can you provide a pcap of one of the calls that are not working?
Also, are these clients behind NAT? Do they use STUN?

Best regards,
Răzvan

On 10/15/19 9:01 PM, Todd Routhier wrote:
> Problem: Calls from PSTN provider > Asterisk > OpenSIPs > SIP Endpoint 
> have intermittent audio issues. See below for details.
> 
> I am a long time Asterisk user but extremely new to OpenSIPs.
> 
> We are in the process of a migration from an older Asterisk server to a 
> newer version along with some other changes.
> 
> First order of business is for us to offload all registrations from our 
> current 1.8.x Asterisk server to OpenSIPs 2.4.6.
> 
> We have a setup that seems to be mostly working but intermittent audio 
> issues are what we are trying to eliminate.
> 
> When I say intermittent, audio seems to work for a particular end 
> point in certain situations or it doesn't. For example, we have some end 
> points which have no audio at all such as my personal soft-phone. I 
> can't get audio on any of three different soft-phones on my laptop, no 
> audio in either direction. But, I have a Grandstream phone on the same 
> LAN which works perfectly every time, on every call.
> 
> I have other end points which are Grandstream phones with perfectly 
> working audio in both directions, all the time, consistently.
> 
> I have other Grandstream end points which work for the same type of call 
> every time, with audio in both directions but the same phone has no 
> audio on slightly different types of calls (hard to explain what I mean 
> by "types of calls").
> 
> Ideally, we would not care about this working with Asterisk 1.8.x since 
> we are moving away from it but it's important for it to work as part of 
> our transition/migration.
> 
> I had horrible audio issues at first were it was hardly working at all 
> or one way audio consistently. I fixed this by setting nat=yes in the 
> sip.conf for the context pointing to the OpenSIPs server. I couldn't 
> understand why this fixed it since the OpenSIPs server and the Asterisk 
> server both have static IP's and are NOT behind any NAT of any sort. 
> Only the end points registered to OpenSIPs are behind end points.
> 
> Still I noticed that Asterisk was trying to send calls to the LAN IP of 
> the end points, so I tested nat=yes and it fixed most of the audio 
> issues with only the issues outlined above remaining.
> 
> My next steps are to see if I have good audio if I push calls to the 
> newer Asterisk server then to the end points registered to the OpenSIPs 
> server. Even if that works, it does not solve my current need to make 
> this work with Asterisk 1.8.x at least until the migration is complete.
> 
> Thanks in advance for any assistance with this.
> 
> Regards,
> 
> Todd
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From razvan at opensips.org  Wed Oct 16 03:06:05 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 16 Oct 2019 10:06:05 +0300
Subject: [OpenSIPS-Users] Clustering 2.4 and 3.0
In-Reply-To: <1571200812.18647143@f116.i.mail.ru>
References: <1571200812.18647143@f116.i.mail.ru>
Message-ID: <e2331591-1905-b59c-2cb1-4e4c8db634fc@opensips.org>

Hi, Alexey!

Most likely clustering 2.4 and 3.0 versions would not work, due to 
different replication packets versions.

Best regards,
Răzvan

On 10/16/19 7:40 AM, Alexey Kazantsev via Users wrote:
> Hi list,
> is it a good/bad idea to create a cluster of 2.4 and 3.0 versions?
> Or it’s of vital importance to have the same version of instances in the 
> cluster?
> The idea is to create a federated user location cluster. [1]
> [1] 
> https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation
> -----------------------------------------------
> BR, Alexey
> http://alexeyka.zantsev.com/
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From kurgan-rus at inbox.ru  Wed Oct 16 03:16:59 2019
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Wed, 16 Oct 2019 10:16:59 +0300
Subject: [OpenSIPS-Users] =?utf-8?q?Clustering_2=2E4_and_3=2E0?=
In-Reply-To: <e2331591-1905-b59c-2cb1-4e4c8db634fc@opensips.org>
References: <1571200812.18647143@f116.i.mail.ru>
 <e2331591-1905-b59c-2cb1-4e4c8db634fc@opensips.org>
Message-ID: <1571210219.160332873@f165.i.mail.ru>



Hi Răzvan,
thank you.
 
So, 2.4 will be the best choice as it it LTS.
 
>Hi, Alexey!
>
>Most likely clustering 2.4 and 3.0 versions would not work, due to
>different replication packets versions.
>
>Best regards,
>Răzvan
 
-----------------------------------------------
BR, Alexey
http://alexeyka.zantsev.com/
 
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From todd at firestreamcloud.com  Wed Oct 16 12:12:48 2019
From: todd at firestreamcloud.com (Todd Routhier)
Date: Wed, 16 Oct 2019 11:12:48 -0500
Subject: [OpenSIPS-Users] OpenSIPs as Registration server in front of
	Asterisk
In-Reply-To: <754c40b0-d1ff-7a23-c425-69cf1fa17fc6@opensips.org>
References: <CA+za3LL0_eQWYWqseQUZYdhH6c3aUdb_gENAGofyOKcZ4Ean5A@mail.gmail.com>
 <754c40b0-d1ff-7a23-c425-69cf1fa17fc6@opensips.org>
Message-ID: <CA+za3LKcDQ2ume=wgAJsPHtsU7-1M_Sma48h_ZQUshg4qyAJoA@mail.gmail.com>

Yes the end point phones are behind NAT but reach is behind a different
NAT. Typically one or two phones at each location. NATs are different at
each location so I'm sure this is why some work and some don't.

None of them use STUN but this is because I never had to use STUN for any
of these same runs points when registered directly to Asterisk.

On Wed, Oct 16, 2019, 2:07 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Hi, Todd!
>
> Can you provide a pcap of one of the calls that are not working?
> Also, are these clients behind NAT? Do they use STUN?
>
> Best regards,
> Răzvan
>
> On 10/15/19 9:01 PM, Todd Routhier wrote:
> > Problem: Calls from PSTN provider > Asterisk > OpenSIPs > SIP Endpoint
> > have intermittent audio issues. See below for details.
> >
> > I am a long time Asterisk user but extremely new to OpenSIPs.
> >
> > We are in the process of a migration from an older Asterisk server to a
> > newer version along with some other changes.
> >
> > First order of business is for us to offload all registrations from our
> > current 1.8.x Asterisk server to OpenSIPs 2.4.6.
> >
> > We have a setup that seems to be mostly working but intermittent audio
> > issues are what we are trying to eliminate.
> >
> > When I say intermittent, audio seems to work for a particular end
> > point in certain situations or it doesn't. For example, we have some end
> > points which have no audio at all such as my personal soft-phone. I
> > can't get audio on any of three different soft-phones on my laptop, no
> > audio in either direction. But, I have a Grandstream phone on the same
> > LAN which works perfectly every time, on every call.
> >
> > I have other end points which are Grandstream phones with perfectly
> > working audio in both directions, all the time, consistently.
> >
> > I have other Grandstream end points which work for the same type of call
> > every time, with audio in both directions but the same phone has no
> > audio on slightly different types of calls (hard to explain what I mean
> > by "types of calls").
> >
> > Ideally, we would not care about this working with Asterisk 1.8.x since
> > we are moving away from it but it's important for it to work as part of
> > our transition/migration.
> >
> > I had horrible audio issues at first were it was hardly working at all
> > or one way audio consistently. I fixed this by setting nat=yes in the
> > sip.conf for the context pointing to the OpenSIPs server. I couldn't
> > understand why this fixed it since the OpenSIPs server and the Asterisk
> > server both have static IP's and are NOT behind any NAT of any sort.
> > Only the end points registered to OpenSIPs are behind end points.
> >
> > Still I noticed that Asterisk was trying to send calls to the LAN IP of
> > the end points, so I tested nat=yes and it fixed most of the audio
> > issues with only the issues outlined above remaining.
> >
> > My next steps are to see if I have good audio if I push calls to the
> > newer Asterisk server then to the end points registered to the OpenSIPs
> > server. Even if that works, it does not solve my current need to make
> > this work with Asterisk 1.8.x at least until the migration is complete.
> >
> > Thanks in advance for any assistance with this.
> >
> > Regards,
> >
> > Todd
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From k.galinurov at gmail.com  Thu Oct 17 02:48:04 2019
From: k.galinurov at gmail.com (Kirill Galinurov)
Date: Thu, 17 Oct 2019 09:48:04 +0300
Subject: [OpenSIPS-Users] APT and YUM repositories down
Message-ID: <CAPDUut-f1S7XYy82F7sdvdTOWOrcdDJBHpW8OXcTwQHbf=DB+Q@mail.gmail.com>

And now yum.opensips.org seem down again.
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From nick at altmann.pro  Thu Oct 17 05:34:43 2019
From: nick at altmann.pro (Nick Altmann)
Date: Thu, 17 Oct 2019 12:34:43 +0300
Subject: [OpenSIPS-Users] APT and YUM repositories down
In-Reply-To: <CAPDUut-f1S7XYy82F7sdvdTOWOrcdDJBHpW8OXcTwQHbf=DB+Q@mail.gmail.com>
References: <CAPDUut-f1S7XYy82F7sdvdTOWOrcdDJBHpW8OXcTwQHbf=DB+Q@mail.gmail.com>
Message-ID: <CAJOe1EMuzug5CTZhDdcW6yTmEit80cKMnRTSGXvCyE4h4rpmhg@mail.gmail.com>

The repositories are up again.

--
Nick


чт, 17 окт. 2019 г. в 09:51, Kirill Galinurov <k.galinurov at gmail.com>:

> And now yum.opensips.org seem down again.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From solarmon at one-n.co.uk  Fri Oct 18 05:49:02 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Fri, 18 Oct 2019 10:49:02 +0100
Subject: [OpenSIPS-Users] change_reply_status - dropping SDP from 183?
In-Reply-To: <CAOXfywP_OJF4N5cyi94CVkXEmfCp886pJk17XBQxu1PGQvhK2A@mail.gmail.com>
References: <CAOXfywP_OJF4N5cyi94CVkXEmfCp886pJk17XBQxu1PGQvhK2A@mail.gmail.com>
Message-ID: <CAOXfywMMTJ=HYekkhJFCxD2eP4yKfQtvigNYcwggemvhx48Euw@mail.gmail.com>

Hi,

If found the solution to my question. I used remove_body_part():

remove_body_part("application/sdp");

Thanks you.

On Fri, 18 Oct 2019 at 09:08, solarmon <solarmon at one-n.co.uk> wrote:

> Hi,
>
> I am able to map 183 to 180 using the change_reply_status() function.
>
> However, I would also like to drop SDP if it is present in the 183. How
> could I achieve this?
>
> Thank you.
>
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From ryanembgrets at gmail.com  Sat Oct 19 06:12:39 2019
From: ryanembgrets at gmail.com (Ryan embgrets)
Date: Sat, 19 Oct 2019 15:12:39 +0500
Subject: [OpenSIPS-Users] [Drouting] failed to add rule id 176415 ->
 skipping, add_prefix: is not decimal digit
Message-ID: <CA+7jXCttm6kd=7tfWLy=8EuH0HSm5FHgLG6MvRKYR-ApC7Y_dw@mail.gmail.com>

Good day,

Whenever I do a restart/reload of opensips/drouting records, my opensips
gives below logs.

Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 176407 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 176409 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 176411 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 176413 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 176415 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 176417 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183011 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183013 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183015 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183017 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183019 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183021 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183027 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route
Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
ERROR:drouting:dr_load_routing_info: failed to add rule id 183029 ->
skipping
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
is not decimal digit
Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
failed to add prefix route

I tried to check the prefix at the problematic id but i could not find any
special character in the prefix. I have re-entered the records just to make
sure that nothing goes amiss in the table but to no avail.

Also, above logs does not look correct, because it always complains about
odd ids.

Any hint how i can find the anomaly here.

Ryan.
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From ryanembgrets at gmail.com  Sat Oct 19 15:04:47 2019
From: ryanembgrets at gmail.com (Ryan embgrets)
Date: Sun, 20 Oct 2019 00:04:47 +0500
Subject: [OpenSIPS-Users] [Drouting] failed to add rule id 176415 ->
 skipping, add_prefix: is not decimal digit
In-Reply-To: <CA+7jXCttm6kd=7tfWLy=8EuH0HSm5FHgLG6MvRKYR-ApC7Y_dw@mail.gmail.com>
References: <CA+7jXCttm6kd=7tfWLy=8EuH0HSm5FHgLG6MvRKYR-ApC7Y_dw@mail.gmail.com>
Message-ID: <CA+7jXCsb72Uze3p=GhM_SMy2PrMZoAk3maoT9CEn2OLF5XpdfA@mail.gmail.com>

Please, ignore this. My application was indeed putting special characters
and opensips was correcting pointing it.

Ryan.

On Sat, 19 Oct 2019 at 15:12, Ryan embgrets <ryanembgrets at gmail.com> wrote:

> Good day,
>
> Whenever I do a restart/reload of opensips/drouting records, my opensips
> gives below logs.
>
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 176407 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 176409 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 176411 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 176413 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 176415 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 176417 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183011 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183013 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183015 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183017 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183019 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183021 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183027 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]:
> ERROR:drouting:dr_load_routing_info: failed to add rule id 183029 ->
> skipping
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_prefix:
> is not decimal digit
> Oct 19 09:53:11 home /usr/sbin/opensips[30138]: ERROR:drouting:add_rule:
> failed to add prefix route
>
> I tried to check the prefix at the problematic id but i could not find any
> special character in the prefix. I have re-entered the records just to make
> sure that nothing goes amiss in the table but to no avail.
>
> Also, above logs does not look correct, because it always complains about
> odd ids.
>
> Any hint how i can find the anomaly here.
>
> Ryan.
>
>
>
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From kurgan-rus at inbox.ru  Mon Oct 21 04:01:10 2019
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Mon, 21 Oct 2019 11:01:10 +0300
Subject: [OpenSIPS-Users] =?utf-8?q?=28no_subject=29?=
Message-ID: <1571644870.152212066@f516.i.mail.ru>


Hi list

I’m trying to find the best solution for setting
federated user location cluster.

Key notes:
- geographically distributed OpenSIPS nodes
- the same numbering plan for all devices (no strict
ranges for Europe/Asia and so on)
- ability to be in service (at least inside of each node)
in case of short periodic connectivity problems.

I configured (in VirtualBox test environment) 2 nodes with
one MongoDB instance, using this tutorial:
https://www.opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation

It works fine.

But I’d like to add some extra high-availability for situations if
there will be some problems with connectivity between nodes.
At least to achieve the ability of each node to serve calls among
those users who are registered directly on it.

What is the best architecture? Maybe something else, not MongoDB?
                                                                                 
                                                                                 
                    user1                                 user4                  
                       |                                    |                    
                       |                                    |                    
                 +-----------+                           +-----------+           
                 |  osips1   |                           |  osips2   |           
    user2------- |  Europe   |<------------------------->|   Asia    |----user5  
                 +-----------+                           +-----------+           
                       |                                       |                 
                       |                                       |                 
                       |                                     user6               
                     user3                                                       
                                                                                 
                                                                                 
 
-----------------------------------------------
BR, Alexey
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From solarmon at one-n.co.uk  Mon Oct 21 06:07:17 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 21 Oct 2019 11:07:17 +0100
Subject: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs stats
	synced?
Message-ID: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>

Hi,

Are dialogs stats meant to be totally synced between to openSIPS nodes in a
cluster - i.e. when using clusterer?

I am monitoring/graphing the active_dialogs, failed_dialogs and
processed_dialogs for each node, using 'opensipsctl fifo get_statistics'
and they are not fully synced:

*Node 1:*
active_dialogs = 6
failed_dialogs = 175
processed_dialogs = 605

*Node 2:*
active_dialogs = 6
failed_dialogs = 0
processed_dialogs = 430

It seems active_dialogs is synced, but failed_dialogs and processed_dialogs
are not.

processed_dialogs figures are increasing on both, but they are not synced.

Please can somebody explain and/or confirm this behaviour.

Thanks
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From kurgan-rus at inbox.ru  Mon Oct 21 06:22:38 2019
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Mon, 21 Oct 2019 13:22:38 +0300
Subject: [OpenSIPS-Users]
 =?utf-8?q?opensipsctl_fifo_get=5Fstatistics_-_di?=
 =?utf-8?q?alogs_stats=09synced=3F?=
In-Reply-To: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
References: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
Message-ID: <1571653358.96500561@f146.i.mail.ru>


Hi, solarmon
 
no, there’s no need to sync dialogs. It’s another type of cluster.
 
As for now, the main purpose is just to have a common
user location dataset, when OpenSIPS nodes will know
the location of each registered device. If it’s connected locally,
it will serve the call locally. If the device is registered on the other node,
it will route the call there.
 
Sure I know about clustering with dialogs syncing (we have such cluster),
but this question is about another type of clustering.
 
I emulated network connectivity ptoblems between two nodes
(just dropped packets going to MongoDB instance) and after that
I was unable to make any calls, no matter which device on which node is registered.
 
So, I’d like to find the best solution which will sustain some short-time
disconnections between nodes.
 
I’m reading about MongoDB clustering but not sure if this is what I need.
>  
 
 
-----------------------------------------------
BR, Alexey
 
 
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From solarmon at one-n.co.uk  Mon Oct 21 07:08:14 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 21 Oct 2019 12:08:14 +0100
Subject: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs
	stats synced?
In-Reply-To: <1571653358.96500561@f146.i.mail.ru>
References: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
 <1571653358.96500561@f146.i.mail.ru>
Message-ID: <CAOXfywPRJiTQ44HNPKXL_vXxNTJK3CWtwMoMv3r_0caLwWHF3g@mail.gmail.com>

Alexey,

Sorry, I'm a bit confused with your answer, and why you are using this
thread to ask another question!

I'm just trying to understand whether these statistics figures should be
synced or not, and why they would be different between the two clustered
openSIPS servers (clustered using corosync/pacemaker for the virtual IP,
and using 'clusterer' between the two OpenSIPS nodes).

On Mon, 21 Oct 2019 at 11:24, Alexey Kazantsev via Users <
users at lists.opensips.org> wrote:

> Hi, solarmon
>
> no, there’s no need to sync dialogs. It’s another type of cluster.
>
> As for now, the main purpose is just to have a common
> user location dataset, when OpenSIPS nodes will know
> the location of each registered device. If it’s connected locally,
> it will serve the call locally. If the device is registered on the other
> node,
> it will route the call there.
>
> Sure I know about clustering with dialogs syncing (we have such cluster),
> but this question is about another type of clustering.
>
> I emulated network connectivity ptoblems between two nodes
> (just dropped packets going to MongoDB instance) and after that
> I was unable to make any calls, no matter which device on which node is
> registered.
>
> So, I’d like to find the best solution which will sustain some short-time
> disconnections between nodes.
>
> I’m reading about MongoDB clustering but not sure if this is what I need.
>
>
>
>
>
> -----------------------------------------------
> BR, Alexey
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From vladp at opensips.org  Mon Oct 21 07:49:26 2019
From: vladp at opensips.org (Vlad Patrascu)
Date: Mon, 21 Oct 2019 14:49:26 +0300
Subject: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs
 stats synced?
In-Reply-To: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
References: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
Message-ID: <676182a2-c2ae-0b2f-a9b6-446a2789312c@opensips.org>

Hi solarmon,

'failed_dialogs' and 'processed_dialogs' represent the total number of 
failed/processed dialogs on that OpenSIPS instance since startup. If one 
of the nodes has been running and handling traffic for a longer time, 
it's normal for the stats to be different. On the other hand 
'active_dialogs' should indeed be always in sync.

Regards,

Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com

On 10/21/2019 01:07 PM, solarmon wrote:
> Hi,
>
> Are dialogs stats meant to be totally synced between to openSIPS nodes 
> in a cluster - i.e. when using clusterer?
>
> I am monitoring/graphing the active_dialogs, failed_dialogs and 
> processed_dialogs for each node, using 'opensipsctl fifo 
> get_statistics' and they are not fully synced:
> *Node 1:*
> active_dialogs = 6
> failed_dialogs = 175
> processed_dialogs = 605
>
> *Node 2:*
> active_dialogs = 6
> failed_dialogs = 0
> processed_dialogs = 430
>
> It seems active_dialogs is synced, but failed_dialogs and 
> processed_dialogs are not.
>
> processed_dialogs figures are increasing on both, but they are not synced.
>
> Please can somebody explain and/or confirm this behaviour.
>
> Thanks
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From solarmon at one-n.co.uk  Mon Oct 21 07:57:32 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 21 Oct 2019 12:57:32 +0100
Subject: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs
	stats synced?
In-Reply-To: <676182a2-c2ae-0b2f-a9b6-446a2789312c@opensips.org>
References: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
 <676182a2-c2ae-0b2f-a9b6-446a2789312c@opensips.org>
Message-ID: <CAOXfywOHKbqO-vF=yhChJQHnUpTUK51BcvDNckLBxir75OKLBA@mail.gmail.com>

 Hi Vlad,

That was my understanding too, but why would processed_dialogs be
incrementing on the standby (State=0) node?

On Mon, 21 Oct 2019 at 12:51, Vlad Patrascu <vladp at opensips.org> wrote:

> Hi solarmon,
>
> 'failed_dialogs' and 'processed_dialogs' represent the total number of
> failed/processed dialogs on that OpenSIPS instance since startup. If one of
> the nodes has been running and handling traffic for a longer time, it's
> normal for the stats to be different. On the other hand 'active_dialogs'
> should indeed be always in sync.
>
> Regards,
>
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 10/21/2019 01:07 PM, solarmon wrote:
>
> Hi,
>
> Are dialogs stats meant to be totally synced between to openSIPS nodes in
> a cluster - i.e. when using clusterer?
>
> I am monitoring/graphing the active_dialogs, failed_dialogs and
> processed_dialogs for each node, using 'opensipsctl fifo get_statistics'
> and they are not fully synced:
>
> *Node 1:*
> active_dialogs = 6
> failed_dialogs = 175
> processed_dialogs = 605
>
> *Node 2:*
> active_dialogs = 6
> failed_dialogs = 0
> processed_dialogs = 430
>
> It seems active_dialogs is synced, but failed_dialogs and
> processed_dialogs are not.
>
> processed_dialogs figures are increasing on both, but they are not synced.
>
> Please can somebody explain and/or confirm this behaviour.
>
> Thanks
>
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://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 vladp at opensips.org  Mon Oct 21 08:29:15 2019
From: vladp at opensips.org (Vlad Patrascu)
Date: Mon, 21 Oct 2019 15:29:15 +0300
Subject: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs
 stats synced?
In-Reply-To: <CAOXfywOHKbqO-vF=yhChJQHnUpTUK51BcvDNckLBxir75OKLBA@mail.gmail.com>
References: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
 <676182a2-c2ae-0b2f-a9b6-446a2789312c@opensips.org>
 <CAOXfywOHKbqO-vF=yhChJQHnUpTUK51BcvDNckLBxir75OKLBA@mail.gmail.com>
Message-ID: <bbb64ba6-f126-716a-c92e-688d938973f4@opensips.org>

The term "processed" should be read more generically in this context to 
include to act of receiving the replicated dialog and preparing it to 
become active at any time.

Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com

On 10/21/2019 02:57 PM, solarmon wrote:
>  Hi Vlad,
>
> That was my understanding too, but why would processed_dialogs be 
> incrementing on the standby (State=0) node?
>
> On Mon, 21 Oct 2019 at 12:51, Vlad Patrascu <vladp at opensips.org 
> <mailto:vladp at opensips.org>> wrote:
>
>     Hi solarmon,
>
>     'failed_dialogs' and 'processed_dialogs' represent the total
>     number of failed/processed dialogs on that OpenSIPS instance since
>     startup. If one of the nodes has been running and handling traffic
>     for a longer time, it's normal for the stats to be different. On
>     the other hand 'active_dialogs' should indeed be always in sync.
>
>     Regards,
>
>     Vlad Patrascu
>     OpenSIPS Developer
>     http://www.opensips-solutions.com
>
>     On 10/21/2019 01:07 PM, solarmon wrote:
>>     Hi,
>>
>>     Are dialogs stats meant to be totally synced between to openSIPS
>>     nodes in a cluster - i.e. when using clusterer?
>>
>>     I am monitoring/graphing the active_dialogs, failed_dialogs and
>>     processed_dialogs for each node, using 'opensipsctl fifo
>>     get_statistics' and they are not fully synced:
>>     *Node 1:*
>>     active_dialogs = 6
>>     failed_dialogs = 175
>>     processed_dialogs = 605
>>
>>     *Node 2:*
>>     active_dialogs = 6
>>     failed_dialogs = 0
>>     processed_dialogs = 430
>>
>>     It seems active_dialogs is synced, but failed_dialogs and
>>     processed_dialogs are not.
>>
>>     processed_dialogs figures are increasing on both, but they are
>>     not synced.
>>
>>     Please can somebody explain and/or confirm this behaviour.
>>
>>     Thanks
>>
>>
>>
>>     _______________________________________________
>>     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
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From solarmon at one-n.co.uk  Mon Oct 21 08:43:16 2019
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 21 Oct 2019 13:43:16 +0100
Subject: [OpenSIPS-Users] opensipsctl fifo get_statistics - dialogs
	stats synced?
In-Reply-To: <bbb64ba6-f126-716a-c92e-688d938973f4@opensips.org>
References: <CAOXfywORR0mKoigQb92XgT50C-PferQgf9pKMUAwX8gpgvKq6Q@mail.gmail.com>
 <676182a2-c2ae-0b2f-a9b6-446a2789312c@opensips.org>
 <CAOXfywOHKbqO-vF=yhChJQHnUpTUK51BcvDNckLBxir75OKLBA@mail.gmail.com>
 <bbb64ba6-f126-716a-c92e-688d938973f4@opensips.org>
Message-ID: <CAOXfywNAu0KxuuHq_rsXAw1rzV5Oqid78LmJQ9eC7XtkcicECw@mail.gmail.com>

Hi Vlad,

Thanks for the clarification.

OK - that makes sense.

I suppose this processed_dialogs value incrementing on the standby node is
a good indicator that clustering and dialog syncing is working as expected.

Thank you.

On Mon, 21 Oct 2019 at 13:30, Vlad Patrascu <vladp at opensips.org> wrote:

> The term "processed" should be read more generically in this context to
> include to act of receiving the replicated dialog and preparing it to
> become active at any time.
>
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 10/21/2019 02:57 PM, solarmon wrote:
>
>  Hi Vlad,
>
> That was my understanding too, but why would processed_dialogs be
> incrementing on the standby (State=0) node?
>
> On Mon, 21 Oct 2019 at 12:51, Vlad Patrascu <vladp at opensips.org> wrote:
>
>> Hi solarmon,
>>
>> 'failed_dialogs' and 'processed_dialogs' represent the total number of
>> failed/processed dialogs on that OpenSIPS instance since startup. If one of
>> the nodes has been running and handling traffic for a longer time, it's
>> normal for the stats to be different. On the other hand 'active_dialogs'
>> should indeed be always in sync.
>>
>> Regards,
>>
>> Vlad Patrascu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>> On 10/21/2019 01:07 PM, solarmon wrote:
>>
>> Hi,
>>
>> Are dialogs stats meant to be totally synced between to openSIPS nodes in
>> a cluster - i.e. when using clusterer?
>>
>> I am monitoring/graphing the active_dialogs, failed_dialogs and
>> processed_dialogs for each node, using 'opensipsctl fifo get_statistics'
>> and they are not fully synced:
>>
>> *Node 1:*
>> active_dialogs = 6
>> failed_dialogs = 175
>> processed_dialogs = 605
>>
>> *Node 2:*
>> active_dialogs = 6
>> failed_dialogs = 0
>> processed_dialogs = 430
>>
>> It seems active_dialogs is synced, but failed_dialogs and
>> processed_dialogs are not.
>>
>> processed_dialogs figures are increasing on both, but they are not synced.
>>
>> Please can somebody explain and/or confirm this behaviour.
>>
>> Thanks
>>
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://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 listUsers at lists.opensips.orghttp://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 genesys.com  Mon Oct 21 15:59:41 2019
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Mon, 21 Oct 2019 19:59:41 +0000
Subject: [OpenSIPS-Users] Opensips 3.0 load display
In-Reply-To: <616cc39d-4105-4c26-8bd0-28b33ddcd9cc@opensips.org>
References: <CAG2rjpty8t2x9RnTTi0+3N5LZ6+XUOuNawk7Yd_6XJTvBXyNpg@mail.gmail.com>
 <050D766A-B85C-49A9-BBEF-32E9572EDD92@genesys.com>
 <616cc39d-4105-4c26-8bd0-28b33ddcd9cc@opensips.org>
Message-ID: <7B7AE50E-ED8B-436D-BBC5-A74F49124A15@genesys.com>

Razvan,

I am curious about this. I can see how you can use the ratelimit module to reject calls based on calls per second, or any configured window, but Mehdi was asking about reporting it as a statistic. I can't see any way to get the value out of the ratelimit module for use elsewhere; it only seems possible to perform a query of whether it is over some configured threshold, not get the value itself.

Ben Newlin 

On 10/1/19, 5:57 AM, "Users on behalf of Răzvan Crainea" <users-bounces at lists.opensips.org on behalf of razvan at opensips.org> wrote:

    Hi, Mehdi!
    
    If you want to measure the Calls per second, you will have to use the 
    ratelimit module[1], which is able to count the calls per second, if you 
    choose a window_size & slot_period [2] values that result in 1 second 
    window.
    
    [1] https://opensips.org/html/docs/modules/3.0.x/ratelimit.html
    [2] 
    https://opensips.org/html/docs/modules/3.0.x/ratelimit.html#param_window_size
    
    Best regards,
    Răzvan
    
    On 9/30/19 3:48 PM, Ben Newlin wrote:
    > Shortly, yes. The load statistic does not measure calls per second. Per 
    > the documentation [1], it measures the percentage of OpenSIPS processes 
    > that are actively processing messages.
    > 
    > There is no built in statistic for calls per second that I am aware of.
    > 
    > [1] 
    > https://www.opensips.org/Documentation/Interface-CoreStatistics-3-0#toc15
    > 
    > Ben Newlin
    > 
    > *From: *Users <users-bounces at lists.opensips.org> on behalf of Mehdi 
    > Shirazi <m.shirazi at gmail.com>
    > *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
    > *Date: *Monday, September 30, 2019 at 12:56 AM
    > *To: *OpenSIPS users mailling list <users at lists.opensips.org>
    > *Subject: *[OpenSIPS-Users] Opensips 3.0 load display
    > 
    > Hi
    > 
    > I use Opensips3.0 as simple stateless proxy.I want to see how many 
    > Invites per second was processed. I used :
    > (opensips-cli): mi get_statistics load:
    > 
    > But all results are zero.Am I doing something wrong ?
    > 
    > Regards
    > 
    > M.Shirazi
    > 
    > 
    > _______________________________________________
    > Users mailing list
    > Users at lists.opensips.org
    > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
    > 
    
    -- 
    Răzvan Crainea
    OpenSIPS Core Developer
       http://www.opensips-solutions.com
    
    _______________________________________________
    Users mailing list
    Users at lists.opensips.org
    http://lists.opensips.org/cgi-bin/mailman/listinfo/users
    
    


From razvan at opensips.org  Tue Oct 22 05:13:01 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 22 Oct 2019 12:13:01 +0300
Subject: [OpenSIPS-Users] Opensips 3.0 load display
In-Reply-To: <7B7AE50E-ED8B-436D-BBC5-A74F49124A15@genesys.com>
References: <CAG2rjpty8t2x9RnTTi0+3N5LZ6+XUOuNawk7Yd_6XJTvBXyNpg@mail.gmail.com>
 <050D766A-B85C-49A9-BBEF-32E9572EDD92@genesys.com>
 <616cc39d-4105-4c26-8bd0-28b33ddcd9cc@opensips.org>
 <7B7AE50E-ED8B-436D-BBC5-A74F49124A15@genesys.com>
Message-ID: <5bc12137-1e61-4998-fe9a-cdd272af5c8a@opensips.org>

Hi, Ben!

The rl_list[1] command does show both the CPS of the previous second and 
the threshold. So indeed, if you are looking for an average CPS within 
the last minute for example, that is not possible - all you can get is 
the current second CPS (calls that were received during the previous 
second).

[1] https://opensips.org/html/docs/modules/3.0.x/ratelimit.html#mi_rl_list

Best regards,
Răzvan

On 10/21/19 10:59 PM, Ben Newlin wrote:
> Razvan,
> 
> I am curious about this. I can see how you can use the ratelimit module to reject calls based on calls per second, or any configured window, but Mehdi was asking about reporting it as a statistic. I can't see any way to get the value out of the ratelimit module for use elsewhere; it only seems possible to perform a query of whether it is over some configured threshold, not get the value itself.
> 
> Ben Newlin
> 
> On 10/1/19, 5:57 AM, "Users on behalf of Răzvan Crainea" <users-bounces at lists.opensips.org on behalf of razvan at opensips.org> wrote:
> 
>      Hi, Mehdi!
>      
>      If you want to measure the Calls per second, you will have to use the
>      ratelimit module[1], which is able to count the calls per second, if you
>      choose a window_size & slot_period [2] values that result in 1 second
>      window.
>      
>      [1] https://opensips.org/html/docs/modules/3.0.x/ratelimit.html
>      [2]
>      https://opensips.org/html/docs/modules/3.0.x/ratelimit.html#param_window_size
>      
>      Best regards,
>      Răzvan
>      
>      On 9/30/19 3:48 PM, Ben Newlin wrote:
>      > Shortly, yes. The load statistic does not measure calls per second. Per
>      > the documentation [1], it measures the percentage of OpenSIPS processes
>      > that are actively processing messages.
>      >
>      > There is no built in statistic for calls per second that I am aware of.
>      >
>      > [1]
>      > https://www.opensips.org/Documentation/Interface-CoreStatistics-3-0#toc15
>      >
>      > Ben Newlin
>      >
>      > *From: *Users <users-bounces at lists.opensips.org> on behalf of Mehdi
>      > Shirazi <m.shirazi at gmail.com>
>      > *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
>      > *Date: *Monday, September 30, 2019 at 12:56 AM
>      > *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>      > *Subject: *[OpenSIPS-Users] Opensips 3.0 load display
>      >
>      > Hi
>      >
>      > I use Opensips3.0 as simple stateless proxy.I want to see how many
>      > Invites per second was processed. I used :
>      > (opensips-cli): mi get_statistics load:
>      >
>      > But all results are zero.Am I doing something wrong ?
>      >
>      > Regards
>      >
>      > M.Shirazi
>      >
>      >
>      > _______________________________________________
>      > Users mailing list
>      > Users at lists.opensips.org
>      > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>      >
>      
>      --
>      Răzvan Crainea
>      OpenSIPS Core Developer
>         http://www.opensips-solutions.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
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From jwilkie at usipcom.com  Tue Oct 22 18:04:28 2019
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Tue, 22 Oct 2019 18:04:28 -0400
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.6 Minor Version
In-Reply-To: <617e8142-1a1a-acc7-94c7-19d72d54c336@opensips.org>
References: <617e8142-1a1a-acc7-94c7-19d72d54c336@opensips.org>
Message-ID: <CANFkrT1UgEbzY42woVC1F3S2kYjO4-5BNuUz9GOw0Rh0fwMEDA@mail.gmail.com>

We've been sitting on 2.4.0 in the lab for a while and would like to update
to 2.4.6 or latest LTS.  Originally installed with a tar file, can we
update without having to completely reinstall or migrate?  Can you link the
procedure if so?

Thank you

On Tue, Jun 11, 2019 at 11:29 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Hi, everyone!
>
> We are thrilled to announce you that we've just released a new minor
> version of the OpenSIPS 2.4 branch. The new OpenSIPS 2.4.6[1] contains
> all the bug fixes we've resolved during the last months, ensuring you
> your deployment more stable now!
> This minor release does not need any migration, so feel free to update
> to the latest version anytime! For a full list of changes, consult the
> ChangeLog[2].
>
> [1] https://opensips.org/pub/opensips/2.4.6/opensips-2.4.6.tar.gz
> [2] https://opensips.org/pub/opensips/2.4.6/ChangeLog
>
> Cheers,
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
> _______________________________________________
> 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  Thu Oct 24 04:50:09 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Thu, 24 Oct 2019 11:50:09 +0300
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.6 Minor Version
In-Reply-To: <CANFkrT1UgEbzY42woVC1F3S2kYjO4-5BNuUz9GOw0Rh0fwMEDA@mail.gmail.com>
References: <617e8142-1a1a-acc7-94c7-19d72d54c336@opensips.org>
 <CANFkrT1UgEbzY42woVC1F3S2kYjO4-5BNuUz9GOw0Rh0fwMEDA@mail.gmail.com>
Message-ID: <153519e4-ee28-a49c-4ec1-9456ff849c8d@opensips.org>

Hi, Jeff!

Unfortunately you can't update without a complete reinstall, but most 
likely there's no need of any migration.
If you still have the old src directory, all you have to do is to copy 
the Makefile.conf file in the new root directory of the decompressed 
opensips-2.4.6.tar.gz.
After that, running `make install` should compile the same modules in 
the same paths as the previous install. NOTE that this will overwrite 
the previous install, but you can always rollback by running `make 
install` on the old decompress dir.
If you don't have the old sources, you'll have to reconfigure opensips 
build all over again.
Hope this helps!

Best regards,
Răzvan

On 10/23/19 1:04 AM, Jeff Wilkie wrote:
> 
> We've been sitting on 2.4.0 in the lab for a while and would like to 
> update to 2.4.6 or latest LTS.  Originally installed with a tar file, 
> can we update without having to completely reinstall or migrate?  Can 
> you link the procedure if so?
> 
> Thank you
> 
> On Tue, Jun 11, 2019 at 11:29 AM Răzvan Crainea <razvan at opensips.org 
> <mailto:razvan at opensips.org>> wrote:
> 
>     Hi, everyone!
> 
>     We are thrilled to announce you that we've just released a new minor
>     version of the OpenSIPS 2.4 branch. The new OpenSIPS 2.4.6[1] contains
>     all the bug fixes we've resolved during the last months, ensuring you
>     your deployment more stable now!
>     This minor release does not need any migration, so feel free to update
>     to the latest version anytime! For a full list of changes, consult the
>     ChangeLog[2].
> 
>     [1] https://opensips.org/pub/opensips/2.4.6/opensips-2.4.6.tar.gz
>     [2] https://opensips.org/pub/opensips/2.4.6/ChangeLog
> 
>     Cheers,
>     -- 
>     Răzvan Crainea
>     OpenSIPS Core Developer
>     http://www.opensips-solutions.com
> 
>     _______________________________________________
>     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
> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com


From jwilkie at usipcom.com  Thu Oct 24 22:38:30 2019
From: jwilkie at usipcom.com (Jeff Wilkie)
Date: Thu, 24 Oct 2019 22:38:30 -0400
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.6 Minor Version
In-Reply-To: <153519e4-ee28-a49c-4ec1-9456ff849c8d@opensips.org>
References: <617e8142-1a1a-acc7-94c7-19d72d54c336@opensips.org>
 <CANFkrT1UgEbzY42woVC1F3S2kYjO4-5BNuUz9GOw0Rh0fwMEDA@mail.gmail.com>
 <153519e4-ee28-a49c-4ec1-9456ff849c8d@opensips.org>
Message-ID: <CANFkrT0qDe3EDFjKfX_tGrGK1EFb2JNAX4Q4z-Z_yEOFVhbAcg@mail.gmail.com>

Thanks for that info.  I've extracted 2.4.6 and saved the old Makefile.conf
then moved it back into the 2.4.6 root directory followed by a make
install.  After completing and running I still get the following on
opensips -V:

version: opensips *2.4.0* (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, sigio_rt, select.
main.c compiled on 22:12:42 Oct 24 2019 with gcc 6.3.0

Shouldn't I see 2.4.6 here?


<https://www.usipcom.com/>
Jeff Wilkie
*CTO*

p: 919-297-1057 <(919)%20297-1057> p/f: 919-882-8490 <(919)%20882-8490> m:
919-348-0753 <(919)%20348-0753>
a: 3201 Northside Dr, Suite 109, Raleigh NC, 27615
w: www.usipcom.com e: jwilkie at usipcom.com

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On Thu, Oct 24, 2019 at 11:25 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Hi, Jeff!
>
> Unfortunately you can't update without a complete reinstall, but most
> likely there's no need of any migration.
> If you still have the old src directory, all you have to do is to copy
> the Makefile.conf file in the new root directory of the decompressed
> opensips-2.4.6.tar.gz.
> After that, running `make install` should compile the same modules in
> the same paths as the previous install. NOTE that this will overwrite
> the previous install, but you can always rollback by running `make
> install` on the old decompress dir.
> If you don't have the old sources, you'll have to reconfigure opensips
> build all over again.
> Hope this helps!
>
> Best regards,
> Răzvan
>
> On 10/23/19 1:04 AM, Jeff Wilkie wrote:
> >
> > We've been sitting on 2.4.0 in the lab for a while and would like to
> > update to 2.4.6 or latest LTS.  Originally installed with a tar file,
> > can we update without having to completely reinstall or migrate?  Can
> > you link the procedure if so?
> >
> > Thank you
> >
> > On Tue, Jun 11, 2019 at 11:29 AM Răzvan Crainea <razvan at opensips.org
> > <mailto:razvan at opensips.org>> wrote:
> >
> >     Hi, everyone!
> >
> >     We are thrilled to announce you that we've just released a new minor
> >     version of the OpenSIPS 2.4 branch. The new OpenSIPS 2.4.6[1]
> contains
> >     all the bug fixes we've resolved during the last months, ensuring you
> >     your deployment more stable now!
> >     This minor release does not need any migration, so feel free to
> update
> >     to the latest version anytime! For a full list of changes, consult
> the
> >     ChangeLog[2].
> >
> >     [1] https://opensips.org/pub/opensips/2.4.6/opensips-2.4.6.tar.gz
> >     [2] https://opensips.org/pub/opensips/2.4.6/ChangeLog
> >
> >     Cheers,
> >     --
> >     Răzvan Crainea
> >     OpenSIPS Core Developer
> >     http://www.opensips-solutions.com
> >
> >     _______________________________________________
> >     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
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From artiom.druz at gmail.com  Fri Oct 25 14:27:27 2019
From: artiom.druz at gmail.com (=?UTF-8?B?0JDRgNGC0LXQvCDQlNGA0YPQt9GM?=)
Date: Fri, 25 Oct 2019 23:27:27 +0500
Subject: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology
	with NoSQL
Message-ID: <CAEBOSMQYy690uanmBJj76CaeLvKjdDaotGAia4usodP0DepKaA@mail.gmail.com>

Hello.
I found in the documentation of the usrloc module (for OpenSIPS 2.4) a
link to an empty tutorial:
https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Full-Sharing
Can someone help with examples for this use case?

Best regards,
Artiom Druz


From liviu at opensips.org  Sat Oct 26 07:12:00 2019
From: liviu at opensips.org (Liviu Chircu)
Date: Sat, 26 Oct 2019 14:12:00 +0300
Subject: [OpenSIPS-Users] Example of configuration "Full Sharing"
 Topology with NoSQL
In-Reply-To: <CAEBOSMQYy690uanmBJj76CaeLvKjdDaotGAia4usodP0DepKaA@mail.gmail.com>
References: <CAEBOSMQYy690uanmBJj76CaeLvKjdDaotGAia4usodP0DepKaA@mail.gmail.com>
Message-ID: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org>

On 25.10.2019 21:27, Артем Друзь wrote:
> Hello.
> I found in the documentation of the usrloc module (for OpenSIPS 2.4) a
> link to an empty tutorial:
> https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Full-Sharing
> Can someone help with examples for this use case?

Hey Artiom,

A full sharing setup is easier to set up than a federated one, so if you 
understand
the federation tutorial, you should have an idea about what you have to do.

Some tips: the "sip_addr" column becomes redundant (unused) in full 
sharing, and you
only need 1 seed node for an N-node setup.  Each node will hold the full 
data set, so
any of them can properly serve any incoming call.  If you also want the 
cluster to
push out OPTIONS pings, make sure to enable the desired behavior [1].

Given that the configuration differences are so small, I have kept 
postponing writing
the tutorial.  But I intend to finally complete it after next week's 
AstriCon or
hopefully even during it.

Regards,

[1]: 
https://opensips.org/html/docs/modules/2.4.x/usrloc.html#param_shared_pinging

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



From liviu at opensips.org  Sat Oct 26 07:18:19 2019
From: liviu at opensips.org (Liviu Chircu)
Date: Sat, 26 Oct 2019 14:18:19 +0300
Subject: [OpenSIPS-Users] (no subject)
In-Reply-To: <1571644870.152212066@f516.i.mail.ru>
References: <1571644870.152212066@f516.i.mail.ru>
Message-ID: <68a6d938-778b-89cd-eba1-c0d4029c54a2@opensips.org>

On 21.10.2019 11:01, Alexey Kazantsev via Users wrote:
> But I’d like to add some extra high-availability for situations if
> there will be some problems with connectivity between nodes.
> At least to achieve the ability of each node to serve calls among
> those users who are registered directly on it.
>
> What is the best architecture? Maybe something else, not MongoDB?

Hey Alexey,

MongoDB has worked fine for the federation use-cases I've deployed,
but only as long as you configure proper values for the connect timeout
setting, server selection timeout setting, and any other options which
may cause blocking when a Mongo cluster node is down [1].

PS: With the built-in MongoDB C library settings, the TCP connect
timeout is 10 seconds, which is abysmally large and will lead to
unacceptable PDD values if any of your POPs become isolated.

Regards,

[1]: 
https://docs.mongodb.com/manual/reference/connection-string/#timeout-options

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

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From kurgan-rus at inbox.ru  Mon Oct 28 04:12:04 2019
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Mon, 28 Oct 2019 11:12:04 +0300
Subject: [OpenSIPS-Users] =?utf-8?q?=28no_subject=29?=
In-Reply-To: <68a6d938-778b-89cd-eba1-c0d4029c54a2@opensips.org>
References: <1571644870.152212066@f516.i.mail.ru>
 <68a6d938-778b-89cd-eba1-c0d4029c54a2@opensips.org>
Message-ID: <1572250324.176830662@f422.i.mail.ru>


Hey Liviu,
 
thank you for the informative answer!
 
By the way, I configured a full-sharing usrloc cluster
and it seems to be what I need.
 
-----------------------------------------------
BR, Alexey
http://alexeyka.zantsev.com/
 
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From kurgan-rus at inbox.ru  Mon Oct 28 04:21:35 2019
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Mon, 28 Oct 2019 11:21:35 +0300
Subject: [OpenSIPS-Users]
 =?utf-8?q?Example_of_configuration_=22Full_Shari?=
 =?utf-8?q?ng=22_Topology_with_NoSQL?=
In-Reply-To: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org>
References: <CAEBOSMQYy690uanmBJj76CaeLvKjdDaotGAia4usodP0DepKaA@mail.gmail.com>
 <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org>
Message-ID: <1572250895.648618260@f112.i.mail.ru>


Hey Artiom,
 
the main difference in call-flow between federated and full-sharing
architecture is as follows:
 
 
Federated cluster:                                    
                                                              
                     +--------+            +--------+         
                     |        |            |        |         
                     | Osips1 |----------> | Osips2 |--->UAC_2
                     +--------+            +--------+         
                      ^                                       
                      |                                       
                      |                                       
                      |                                       
                   UAC_1                                      
                                                              
                                                              
                                                              
                                                              
                                                              
Full-sharing cluster:                                  
                                                              
                    +--------+            +---------+         
                    |        |            |         |         
                    |Osips1  |            |Osips2   |         
                    +--------+-\          +---------+         
                     ^          --\                           
                     |             ---\                       
                     |                 --\                    
                  UAC_1                   -> UAC_2            
 
 
 
 
So, as already written in the documention, the full-sharing
architecture requires either an edge SBC or a lack of any
network/IP routing problems between UACs and all OpenSIPS nodes.
 
-----------------------------------------------
BR, Alexey
http://alexeyka.zantsev.com/
 
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From bhattravi4 at gmail.com  Tue Oct 15 05:02:06 2019
From: bhattravi4 at gmail.com (RAVI bhatt)
Date: Tue, 15 Oct 2019 14:32:06 +0530
Subject: [OpenSIPS-Users] Opensips crash while connecting mariadb
Message-ID: <CAKrXSuiGj+_A6OM7OGKZMKWohoK7YaP9k6aRTYi1CstQSBY_fw@mail.gmail.com>

Hello all,
i am using opensips 3.0.1 with mariadb 10.4.8 and i am facing issue  in
starting opensips. As opensips crash with below given message:
ct 15 03:27:17 [22849] DBG:core:find_mod_export: found <db_bind_api> in
module db_mysql [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for db_mysql
Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not
found in pool
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno:
2000

i have attached logs and back  trace from core  so please provide your
suggestions.
please note credentials given in db_url are working while using from msyql
client

Thanks in advance.
Ravindra Bhatt
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Oct 15 03:27:17 [22849] INFO:core:evi_publish_event: Registered event <E_DLG_STATE_CHANGED(11)>
Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found <load_tm> in module tm [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found <load_rr> in module rr [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found <db_bind_api> in module db_mysql [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for db_mysql
Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not found in pool
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000

Got ERROR: "InnoDB: Operating system error number 11 in a file operation." errno: 2000
Got ERROR: "InnoDB: Error number 11 means 'Resource temporarily unavailable'" errno: 2000
Got ERROR: "InnoDB: Cannot open datafile '/var/lib/mysql/ibdata1'" errno: 2000
Got ERROR: "InnoDB: Could not open or create the system tablespace. If you tried to add new data files to the system tablespace, and it failed here, you should now edit innodb_data_file_path in my.cnf back to what it was, and remove the new ibdata files InnoDB created in this failed attempt. InnoDB only wrote those files full of zeros, but did not yet use them in any way. But be careful: do not remove old data files which contain your precious data!" errno: 2000
Got ERROR: "InnoDB: Plugin initialization aborted with error Cannot open a file" errno: 2000
Got ERROR: "Plugin 'InnoDB' init function returned error." errno: 2000
Got ERROR: "Plugin 'InnoDB' registration as a STORAGE ENGINE failed." errno: 2000
Got ERROR: "unknown: Can't lock aria control file '/var/lib/mysql/aria_log_control' for exclusive use, error: 11. Will retry for 30 seconds" errno: 2000



^C^CGot ERROR: "unknown: Got error 'Could not get an exclusive lock; file is probably in use by another process' when trying to use aria control file '/var/lib/mysql/aria_log_control'" errno: 2000
Got ERROR: "Plugin 'Aria' init function returned error." errno: 2000
Got ERROR: "Plugin 'Aria' registration as a STORAGE ENGINE failed." errno: 2000
Got ERROR: "Unknown/unsupported storage engine: InnoDB" errno: 2000
 DBG:db_mysql:db_mysql_connect: opening connection: mysql://xxxx:xxxx at localhost/cc_master
CRITICAL:core:sig_usr: segfault in attendant (starter) process!
DBG:core:restore_segv_handler: restoring SIGSEGV handler...
 DBG:core:restore_segv_handler: successfully restored system SIGSEGV handler
^CSegmentation fault (core dumped)

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(gdb) bt full
#0  intern_plugin_lock (lex=0x0, state_mask=14, rc=0x0)
    at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:948
        pi = 0x0
#1  plugin_thdvar_init (thd=0x28af578) at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:3155
        old_table_plugin = 0x0
        old_tmp_table_plugin = 0x0
        old_enforced_table_plugin = 0x0
#2  0x00007fc7d58722b1 in THD::init (this=this at entry=0x28af578)
    at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:1177
No locals.
#3  0x00007fc7d587310a in THD::THD (this=0x28af578, id=<optimized out>, is_wsrep_applier=<optimized out>)
    at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:798
        tmp = <optimized out>
#4  0x00007fc7d57f2bdc in create_embedded_thd (client_flag=client_flag at entry=-2143837683)
    at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/lib_sql.cc:685
        thd = 0x7fff689c20b0
#5  0x00007fc7d57fa1e4 in mysql_real_connect (mysql=0x7fc7da7190b0, host=<optimized out>, user=<optimized out>, 
    passwd=<optimized out>, db=0x7fc7da717620 "cc_master", port=port at entry=0, unix_socket=unix_socket at entry=0x0, 
    client_flag=2151129613, client_flag at entry=2147549184)
    at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/libmysqld.c:179
        name_buff = "\200\314|\326\307\177\000\000\060!\234h\377\177\000\000 !\234h\377\177\000\000\060!\234h\377\177\000\000\200\314|\326\307\177\000\000\001\000\000\000\000\000\000\000\340r\242\000\000\000\000\000p\251\215\000\000\000\000\000\340r\242\000\000\000\000\000\270\"\234h\377\177\000\000 vq\332\307\177\000\000\362\344s\333\307\177\000\000\000\000\000\000\000\000\000\000\060\000\000\000\060\000\000\000\060\"\234h\377\177\000\000P!\234h\377\177", '\000' <repeats 26 times>, "\224*\207\333\307\177\000\000\005+\207\333\307\177\000\000\300\061\253\333\307\177\000\000\200\363\252\333\307\177\000\000c\226u\333\307\177\000\000 vq\332\307\177\000\000"...
#6  0x00007fc7d70e8a63 in db_mysql_connect (ptr=ptr at entry=0x7fc7da717668) at my_con.c:105
        reconnect = 0 '\000'
---Type <return> to continue, or q <return> to quit---
        __FUNCTION__ = "db_mysql_connect"
#7  0x00007fc7d70e94ff in db_mysql_new_connection (id=0x7fc7da717510) at my_con.c:165
        ptr = 0x7fc7da717668
        __FUNCTION__ = "db_mysql_new_connection"
#8  0x00000000005ec4ba in db_do_init (url=<optimized out>, 
    new_connection=0x7fc7d70e9417 <db_mysql_new_connection>) at db/db.c:338
        id = 0x7fc7da717510
        con = <optimized out>
        res = 0x7fc7da717468
        con_size = <optimized out>
        __FUNCTION__ = "db_do_init"
#9  0x00007fc7d01044cd in dlg_connect_db (db_url=db_url at entry=0x7fc7d0340610 <db_url>) at dlg_db_handler.c:135
        __FUNCTION__ = "dlg_connect_db"
#10 0x00007fc7d0104511 in init_dlg_db (db_url=db_url at entry=0x7fc7d0340610 <db_url>, 
    dlg_hash_size=<optimized out>, db_update_period=60) at dlg_db_handler.c:176
        __FUNCTION__ = "init_dlg_db"
#11 0x00007fc7d0100c27 in mod_init () at dialog.c:900
        n = <optimized out>
        __FUNCTION__ = "mod_init"
#12 0x000000000050bfa7 in init_mod (m=0x7fc7da6face0, skip_others=skip_others at entry=0) at sr_module.c:697
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#13 0x000000000050c028 in init_mod (m=0x7fc7da6fb970, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#14 0x000000000050c028 in init_mod (m=0x7fc7da6fbd28, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
---Type <return> to continue, or q <return> to quit---
#15 0x000000000050c028 in init_mod (m=0x7fc7da6fbeb8, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#16 0x000000000050c028 in init_mod (m=0x7fc7da6fc0c0, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#17 0x000000000050c028 in init_mod (m=0x7fc7da6fc340, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#18 0x000000000050c028 in init_mod (m=0x7fc7da6fc4d8, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#19 0x000000000050c028 in init_mod (m=0x7fc7da6fd488, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#20 0x000000000050c028 in init_mod (m=0x7fc7da6fde20, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#21 0x000000000050f45d in init_modules () at sr_module.c:759
        currentMod = 0x0
        ret = <optimized out>
        __FUNCTION__ = "init_modules"
#22 0x0000000000420351 in main (argc=<optimized out>, argv=<optimized out>) at main.c:1421
        c = <optimized out>
        r = <optimized out>
        tmp = 0x1 <Address 0x1 out of bounds>
        tmp_len = <optimized out>
        port = <optimized out>
---Type <return> to continue, or q <return> to quit---
        proto = <optimized out>
        protos_no = <optimized out>
        options = 0x67e9f8 "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"
        ret = -1
        seed = 961253727
        rfd = <optimized out>
        __FUNCTION__ = "main"


From todd at virtualtas.com  Wed Oct 16 12:04:40 2019
From: todd at virtualtas.com (Todd Routhier)
Date: Wed, 16 Oct 2019 11:04:40 -0500
Subject: [OpenSIPS-Users] OpenSIPs as Registration server in front of
	Asterisk
In-Reply-To: <754c40b0-d1ff-7a23-c425-69cf1fa17fc6@opensips.org>
References: <CA+za3LL0_eQWYWqseQUZYdhH6c3aUdb_gENAGofyOKcZ4Ean5A@mail.gmail.com>
 <754c40b0-d1ff-7a23-c425-69cf1fa17fc6@opensips.org>
Message-ID: <CA+za3LKmg3FGFKi+U9dgz9SG5ezSCzSvoGss_bVspv=qbBMTuA@mail.gmail.com>

Yes the end point phones are behind NAT but reach is behind a different
NAT. Typically one or two phones at each location. NATs are different at
each location so I'm sure this is why some work and some don't.

None of them use STUN but this is because I never had to use STUN for any
of these same runs points when registered directly to Asterisk.


On Wed, Oct 16, 2019, 2:07 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Hi, Todd!
>
> Can you provide a pcap of one of the calls that are not working?
> Also, are these clients behind NAT? Do they use STUN?
>
> Best regards,
> Răzvan
>
> On 10/15/19 9:01 PM, Todd Routhier wrote:
> > Problem: Calls from PSTN provider > Asterisk > OpenSIPs > SIP Endpoint
> > have intermittent audio issues. See below for details.
> >
> > I am a long time Asterisk user but extremely new to OpenSIPs.
> >
> > We are in the process of a migration from an older Asterisk server to a
> > newer version along with some other changes.
> >
> > First order of business is for us to offload all registrations from our
> > current 1.8.x Asterisk server to OpenSIPs 2.4.6.
> >
> > We have a setup that seems to be mostly working but intermittent audio
> > issues are what we are trying to eliminate.
> >
> > When I say intermittent, audio seems to work for a particular end
> > point in certain situations or it doesn't. For example, we have some end
> > points which have no audio at all such as my personal soft-phone. I
> > can't get audio on any of three different soft-phones on my laptop, no
> > audio in either direction. But, I have a Grandstream phone on the same
> > LAN which works perfectly every time, on every call.
> >
> > I have other end points which are Grandstream phones with perfectly
> > working audio in both directions, all the time, consistently.
> >
> > I have other Grandstream end points which work for the same type of call
> > every time, with audio in both directions but the same phone has no
> > audio on slightly different types of calls (hard to explain what I mean
> > by "types of calls").
> >
> > Ideally, we would not care about this working with Asterisk 1.8.x since
> > we are moving away from it but it's important for it to work as part of
> > our transition/migration.
> >
> > I had horrible audio issues at first were it was hardly working at all
> > or one way audio consistently. I fixed this by setting nat=yes in the
> > sip.conf for the context pointing to the OpenSIPs server. I couldn't
> > understand why this fixed it since the OpenSIPs server and the Asterisk
> > server both have static IP's and are NOT behind any NAT of any sort.
> > Only the end points registered to OpenSIPs are behind end points.
> >
> > Still I noticed that Asterisk was trying to send calls to the LAN IP of
> > the end points, so I tested nat=yes and it fixed most of the audio
> > issues with only the issues outlined above remaining.
> >
> > My next steps are to see if I have good audio if I push calls to the
> > newer Asterisk server then to the end points registered to the OpenSIPs
> > server. Even if that works, it does not solve my current need to make
> > this work with Asterisk 1.8.x at least until the migration is complete.
> >
> > Thanks in advance for any assistance with this.
> >
> > Regards,
> >
> > Todd
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>    http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From monideth.pen at one-n.co.uk  Fri Oct 18 04:06:42 2019
From: monideth.pen at one-n.co.uk (Monideth Pen)
Date: Fri, 18 Oct 2019 09:06:42 +0100
Subject: [OpenSIPS-Users] change_reply_status - dropping SDP from 183?
Message-ID: <CAOXfywOsn0JJ8dVpAZmJimnCX5_XKSViH2dizSddyWvastnkSQ@mail.gmail.com>

Hi,

I am able to map 183 to 180 using the change_reply_status() function.

However, I would also like to drop SDP if it is present in the 183. How
could I achieve this?

Thank you.
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From callum.guy at x-on.co.uk  Mon Oct 28 06:14:47 2019
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Mon, 28 Oct 2019 10:14:47 +0000
Subject: [OpenSIPS-Users] Double record route with TLS
Message-ID: <CAFjCFzmNCRgH=C=O4cwt=U6aqmeXnmo9+skx_jkj7r6WRp5AsA@mail.gmail.com>

Hi All,

Using 3.0.1

I'm dealing with a call scenario where a UA is dialling out using TLS and
travelling between two geographic sites, both of which use NAT. TLS is only
required between the UA and the initial registrar, once the initial
requests hit the network it is forwarded using cleartext UDP. Instances
listen on the internal address only

The problem I am facing is that double_rr is not using my defined
advertised address of the public IP, instead it replaces one of the
addresses and not the other causing a routing breakdown later in the
dialog. I'm using set_advertised_address(SITEA-PUB-IP)

To illustrate, the following headers are presented at site B:

INVITE sip:01234567890 at 192.168.153.226 SIP/2.0
Record-Route:
<sip:SITEA-PUB-IP;r2=on;lr;ftag=1779783491;hyth=3a6.e9f72846;nat=yes>
Record-Route:
<sip:SITEA-PRI-IP:5061;transport=tls;r2=on;lr;ftag=1779783491;hyth=3a6.e9f72846;nat=yes>

For correct routing I need both of those to present the public address. In
a typical NAT traversal scenario I have reply routes which re-write the
public and private addresses accordingly and I can certainly intercept
these INVITE messages and do that however I wondered if there was a cleaner
way. Am I missing a trick?

I have attempted to make changes using a site A branch route to rewrite the
RR however the header is not yet formed. I also attempted to use
record_route_preset() however this did not have the desired effect.

Any ideas would be very much appreciated!

Callum

-- 





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From fgast+opensips at only640k.net  Mon Oct 28 07:06:15 2019
From: fgast+opensips at only640k.net (Fabian Gast)
Date: Mon, 28 Oct 2019 12:06:15 +0100 (CET)
Subject: [OpenSIPS-Users] change_reply_status - dropping SDP from 183?
In-Reply-To: <CAOXfywOsn0JJ8dVpAZmJimnCX5_XKSViH2dizSddyWvastnkSQ@mail.gmail.com>
References: <CAOXfywOsn0JJ8dVpAZmJimnCX5_XKSViH2dizSddyWvastnkSQ@mail.gmail.com>
Message-ID: <292888740.2497.1572260774995.JavaMail.zimbra@only640k.net>


You can use something like

loadmodule "sipmsgops.so"

onreply_route[myreply] {
	
	if (t_check_status("183")) {
		change_reply_status("180", "Ringing");
		remove_body_part("application/sdp");
	}
}

Fabian

----- Ursprüngliche Mail -----
Von: "Monideth Pen" <monideth.pen at one-n.co.uk>
An: "OpenSIPS users mailling list" <users at lists.opensips.org>
Gesendet: Freitag, 18. Oktober 2019 10:06:42
Betreff: [OpenSIPS-Users] change_reply_status - dropping SDP from 183?

Hi, 
I am able to map 183 to 180 using the change_reply_status() function. 

However, I would also like to drop SDP if it is present in the 183. How could I achieve this? 

Thank you. 

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



From social at bohboh.info  Mon Oct 28 10:27:15 2019
From: social at bohboh.info (Social Boh)
Date: Mon, 28 Oct 2019 09:27:15 -0500
Subject: [OpenSIPS-Users] Example of configuration "Full Sharing"
 Topology with NoSQL
In-Reply-To: <1572250895.648618260@f112.i.mail.ru>
References: <CAEBOSMQYy690uanmBJj76CaeLvKjdDaotGAia4usodP0DepKaA@mail.gmail.com>
 <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org>
 <1572250895.648618260@f112.i.mail.ru>
Message-ID: <7d4b9348-1366-9e4c-c1d6-053d347c9e39@bohboh.info>

Hello,

if i don't want use a SBC how can I known is a node of cluster is up or
down? which method do you advise?

Thank you

---
I'm SoCIaL, MayBe

El 28/10/2019 a las 03:21, Alexey Kazantsev via Users escribió:
> Hey Artiom,
>  
> the main difference in call-flow between federated and full-sharing
> architecture is as follows:
>  
>  
> Federated cluster:                                    
>                                                               
>                      +--------+            +--------+         
>                      |        |            |        |         
>                      | Osips1 |----------> | Osips2 |--->UAC_2
>                      +--------+            +--------+         
>                       ^                                       
>                       |                                       
>                       |                                       
>                       |                                       
>                    UAC_1                                      
>                                                               
>                                                               
>                                                               
>                                                               
>                                                               
> Full-sharing cluster:                                  
>                                                               
>                     +--------+            +---------+         
>                     |        |            |         |         
>                     |Osips1  |            |Osips2   |         
>                     +--------+-\          +---------+         
>                      ^          --\                           
>                      |             ---\                       
>                      |                 --\                    
>                   UAC_1                   -> UAC_2            
>  
>  
>  
>  
> So, as already written in the documention, the full-sharing
> architecture requires either an edge SBC or a lack of any
> network/IP routing problems between UACs and all OpenSIPS nodes.
>  
> -----------------------------------------------
> BR, Alexey
> http://alexeyka.zantsev.com/
>  
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From fosdem-rtc-admin at freertc.org  Mon Oct 28 13:49:48 2019
From: fosdem-rtc-admin at freertc.org (fosdem-rtc-admin at freertc.org)
Date: Mon, 28 Oct 2019 18:49:48 +0100 (CET)
Subject: [OpenSIPS-Users] Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers,
	volunteers neeeded
References: <55cee853-9e19-4b0d-a86d-d3f96b404265@freertc.org>
Message-ID: <20191028174948.449B5320C4@daniel1.office.readytechnology.co.uk>

FOSDEM - Real Time Communications devroom CfP
=============================================

Overview
--------

[FOSDEM](https://fosdem.org) is one of the world's premier meetings of
free software developers, with over five thousand people attending each
year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium.

This document contains information about:

-   Real-Time Communications developer room (devroom) and lounge
-   speaking opportunities
-   volunteering in the devroom and lounge
-   social events (the legendary FOSDEM Beer Night and Saturday night
    dinners provide endless networking opportunities)
-   the Planet aggregation sites for RTC blogs

**NEW:** Save yourself entering Free-RTC events and CFP deadlines into
your calendar and task list, follow our iCalendar feed:
https://freertc.org/events.ics

Call for participation - Real Time Communications (RTC)
-------------------------------------------------------

The Real-Time devroom and Real-Time lounge are about all things
involving real-time communication, including: XMPP, SIP, WebRTC,
telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption.

**We are looking for speakers for the devroom and volunteers and
participants for the tables in the Real-Time lounge.**

The devroom is only on Sunday, 2nd of February 2020. The lounge will be
present for both days.

To discuss the devroom and lounge, please join the [Free-RTC mailing
list](http://lists.freertc.org/mailman/listinfo/discuss).

### Speaking opportunities

Note: if you used FOSDEM Pentabarf before, please use the same
account/username

Real-Time Communications devroom: deadline 23:59 UTC on 15th of
December. Please use the
[Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to
submit a talk proposal for the devroom. On the "General" tab, please
look for the "Track" option and choose "Real Time Communications
devroom".

Other devrooms and lightning talks: some speakers may find their topic
is in the scope of more than one devroom. It is encouraged to apply to
more than one devroom and also consider proposing a lightning talk, but
please be kind enough to tell us if you do this by filling out the notes
in the form. Here you can find the [full list of
devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can
apply for a [lightning talk](https://fosdem.org/submit).

### First-time speaking?

FOSDEM devrooms are a welcoming environment for people who have never
given a talk before. Please feel free to contact the devroom
administrators personally if you would like to ask any questions about
it.

### Submission guidelines

The Pentabarf system will ask for many of the essential details. Please
remember to re-use your account from previous years if you have one.

In the "Submission notes", please tell us about:

-   The purpose of your talk
-   Any other talk applications (devrooms, lightning talks, main track)
-   Availability constraints and special needs

You can use HTML and links in your bio, abstract and description.

If you maintain a blog, please consider providing us with the URL of a
feed with posts tagged for your RTC-related work.

We will be looking for relevance to the conference and devroom themes,
presentations aimed at developers of free and open source software about
RTC-related topics.

Please feel free to suggest a duration between 20 minutes and 55 minutes
but note that the final decision on talk durations will be made by the
devroom administrators based on the number of received proposals. As the
two previous devrooms have been combined into one, we may decide to give
shorter slots than in previous years so that more speakers can
participate.

Please note FOSDEM aims to record and live-stream all talks. The CC-BY
license is used.

Volunteers needed
-----------------

To make the devroom and lounge run successfully, we are looking for
volunteers:

-   FOSDEM provides video recording equipment and live streaming,
    volunteers are needed to assist in this
-   Organizing one or more restaurant bookings (dependending upon number
    of participants) for the evening of Saturday, 1 February
-   Participation in the Real-Time lounge
-   Circulating this Call for Participation to other mailing lists

Social events and dinners
-------------------------

The traditional FOSDEM beer night occurs on Friday, 31st of January.

On Saturday night, there are usually dinners associated with each of the
devrooms. Most restaurants in Brussels are not so large so these dinners
have space constraints and reservations are essential. Please subscribe
to the [Free-RTC mailing
list](http://lists.freertc.org/mailman/listinfo/discuss) for further
details about the Saturday night dinner options and how you can register
for a seat.

Related events around FOSDEM
----------------------------

As per usual, the [XMPP
Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening
ahead of FOSDEM. This time it will take place on the 30th and 31st of
January also in Brussels. All details are available on the [summit
website](https://wiki.xmpp.org/web/Conferences/Summit_24).

Spread the word and discuss
---------------------------

If you know of any mailing lists where this CfP would be relevant,
please forward this document. If this devroom excites you, please blog
or microblog about it, especially if you are submitting a talk.

If you regularly blog about RTC topics, please send details about your
blog to the planet site administrators:

-   All projects https://planet.freertc.org planet at freertc.org
-   XMPP https://planet.jabber.org ralphm at ik.nu
-   SIP https://planet.sip5060.net planet at sip5060.net
-   (Español) https://planet.sip5060.net/es/ planet at sip5060.net

Please also link to the Planet sites from your own blog or web site as
this helps everybody in the free real-time communications community.

Contact
-------

For any private queries, contact us directly using the address
**fosdem-rtc-admin at freertc.org** and for any other queries please ask on
the [Free-RTC mailing
list](http://lists.freertc.org/mailman/listinfo/discuss).

The devroom administration team:

-   Saúl Ibarra Corretgé <s at saghul.net>
-   Ralph Meijer <ralphm at ik.nu>
-   Daniel-Constantin Mierla <miconda at gmail.com>
-   Daniel Pocock <daniel at pocock.pro>
-   Guus der Kinderen <guus.der.kinderen at gmail.com>



From miha at softnet.si  Tue Oct 29 05:13:07 2019
From: miha at softnet.si (Miha)
Date: Tue, 29 Oct 2019 10:13:07 +0100
Subject: [OpenSIPS-Users] Remove doubled connection information in SDP
Message-ID: <fd7fef34-1306-36ae-a9f2-d1f41d6190bf@softnet.si>

Hello

I get two connection infomrmation in SDP (doubled), which are the same. 
How to remove one?

ps.: i am using rtpproxy.

thank you.
Miha
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From bogdan at opensips.org  Tue Oct 29 06:25:35 2019
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 29 Oct 2019 12:25:35 +0200
Subject: [OpenSIPS-Users] OpenSIPS Summit 2020 - Amsterdam, NL
Message-ID: <48c6a78e-8370-f29a-4ae8-1a50fc4b27f4@opensips.org>

<http://www.opensips.org/events/Summit-2020Amsterdam>


  OpenSIPS Summit 2020

May 5th-8th, 2020

Amsterdam, The Netherlands


*Bridging people, bridging technologies, bridging experiences
*

OpenSIPS <https://www.opensips.org> is one of the most used Open Source 
SIP Servers, if we are to count the number of calls it routes across the 
world. Form complex end-user services to high throughput infrastructure 
components, OpenSIPS is undoubted a SIP Server able to deliver in 
thousands of deployments, for carriers, telcos or ITSPs.

The OpenSIPS Summit 
<https://www.opensips.org/events/Summit-2020Amsterdam/> is the meeting 
place for the OpenSIPS community, for experts, developers and users from 
all over the world, looking to learn and gain knowledge. The OpenSIPS 
Summit is a melting pot for discussion on new technology, for sharing 
experiences, for brainstorming on new trends, for building bridges in 
the Open-Source VoIP & RTC ecosystem.

*Some Great Reasons to Attend*

  * Access the latest news, knowledge and experience in the VoIP & RTC world
  * Learn about upcoming 3.0 OpenSIPS release and how you can leverage it
  * Attend unique presentations and interactive technical workshops
  * Meet FOSS developers and community to share experience and comments
  * Get solutions consultancy during the Free Design Clinics
  * Become an Expert attending the OpenSIPS Advanced Training


*Summit Agenda*

  * Two full days of presentations given by key speakers
  * Open Discussions with key people from OpenSIPS and other OSS projects
  * One full day of Interactive Demos and Showcases
  * One full day of Design Clinics to validate your OpenSIPS deployments
  * One full day OpenSIPS Training (limited seats!)
  * Social events in the amazing Amsterdam


*Be part of it*

Be a part of both OpenSIPS and the Open Source community, be part of the 
OpenSIPS Summit 2020.

*Attend to learn* - the registration process will be open in the 
following days, stay tuned. Nevertheless, pre-registration is available, 
just contact us.

*Speak to share* - the Call for Papers will be announced during next 
week, so you can share your wisdom and experience with the world.

*Sponsor to help* - we welcome any help in making the Summit such a 
great event. Sponsoring is a natural way of saying "Thank you" for the 
Open Source code you are using within your businesses.

Interested? Please contact 
<https://www.opensips.org/events/Summit-2020Amsterdam/#contact-map> our 
team or email <mailto:summit at opensips.org?subject=OpenSIPS%20Summit> us!


<https://www.google.com/maps/place/Radisson+Blu+Hotel,+Amsterdam/@52.3702502,4.8944856,17z/data=%213m1%214b1%214m5%213m4%211s0x47c609bf6cde2ad5:0x764aad4243f5348f%218m2%213d52.3702469%214d4.8966743?hl=en> 


*
*

*Radisson Blu**
**Rusland 17, 1012CK Amsterdam, The Netherlands*

Meet us again at our familiar Venue, with  the usual space and comfort!


**


	
	

	
	


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

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From Ben.Newlin at genesys.com  Tue Oct 29 09:20:05 2019
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Tue, 29 Oct 2019 13:20:05 +0000
Subject: [OpenSIPS-Users] Branch Flags
Message-ID: <4AE92CBD-31B1-46AF-A0B9-2762E4429BA1@genesys.com>

Hey all,

I am trying to use the branch flags functionality described here [1]. I am running into an issue when using the syntax with the branch_idx. The documentation does not provide any examples for this form.

Basically, from my testing in 2.4.6 it appears that the branch_idx can only be a static number. I have tried numerous forms to try to get a variable to be accepted but all result in syntax errors. Is this expected? If so, is anyone successfully using this functionality? I am having trouble thinking of a scenario where the flags could be useful if the branch numbers must be hard-coded in the script.

Works:
setbflag(1, FLAG)

Syntax Error:
setbflag($T_branch_idx, FLAG)
setbflag(“$T_branch_idx”, FLAG)
setbflag($T_branch_idx, “FLAG”)
setbflag(“$T_branch_idx”, “FLAG”)


[1] - https://www.opensips.org/Documentation/Script-Flags-2-4#toc4

Ben Newlin
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From razvan at opensips.org  Tue Oct 29 11:18:10 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 29 Oct 2019 17:18:10 +0200
Subject: [OpenSIPS-Users] Remove doubled connection information in SDP
In-Reply-To: <fd7fef34-1306-36ae-a9f2-d1f41d6190bf@softnet.si>
References: <fd7fef34-1306-36ae-a9f2-d1f41d6190bf@softnet.si>
Message-ID: <ceb9cbbb-1cb6-8c6a-795a-7978cf5a1607@opensips.org>

Hi, Miha!

You can use the subst body[1] to remove the second c= line, but note 
that this will change the "semantics" of your SDP.

[1] 
https://opensips.org/html/docs/modules/3.1.x/textops.html#func_subst_body

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 10/29/19 11:13 AM, Miha via Users wrote:
> Hello
> 
> I get two connection infomrmation in SDP (doubled), which are the same. 
> How to remove one?
> 
> ps.: i am using rtpproxy.
> 
> thank you.
> Miha
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


From razvan at opensips.org  Tue Oct 29 11:24:52 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 29 Oct 2019 17:24:52 +0200
Subject: [OpenSIPS-Users] Branch Flags
In-Reply-To: <4AE92CBD-31B1-46AF-A0B9-2762E4429BA1@genesys.com>
References: <4AE92CBD-31B1-46AF-A0B9-2762E4429BA1@genesys.com>
Message-ID: <fe425512-dac6-5d56-ee8b-1fc796fe987e@opensips.org>

Hi, Ben!

Unfortunately the `setbflag` function only accepts static indexes, 
that's why it doesn't work no matter how you try.
Perhaps the only scenario where a static branch is useful is when you 
"manually" add branches in your script (using append_branch), and you 
know exactly the order those branches were added, and how to index them.

I believe I ran into this issue a while ago, and I was able to solve it 
by using an AVP, and indexing it using the branch index. I know it's not 
the best way to do it, but it worked for me back then :)

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 10/29/19 3:20 PM, Ben Newlin wrote:
> Hey all,
> 
> I am trying to use the branch flags functionality described here [1]. I 
> am running into an issue when using the syntax with the branch_idx. The 
> documentation does not provide any examples for this form.
> 
> Basically, from my testing in 2.4.6 it appears that the branch_idx can 
> only be a static number. I have tried numerous forms to try to get a 
> variable to be accepted but all result in syntax errors. Is this 
> expected? If so, is anyone successfully using this functionality? I am 
> having trouble thinking of a scenario where the flags could be useful if 
> the branch numbers must be hard-coded in the script.
> 
> Works:
> 
> setbflag(1, FLAG)
> 
> Syntax Error:
> 
> setbflag($T_branch_idx, FLAG)
> 
> setbflag(“$T_branch_idx”, FLAG)
> 
> setbflag($T_branch_idx, “FLAG”)
> 
> setbflag(“$T_branch_idx”, “FLAG”)
> 
> [1] - https://www.opensips.org/Documentation/Script-Flags-2-4#toc4
> 
> Ben Newlin
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


From Ben.Newlin at genesys.com  Tue Oct 29 11:32:06 2019
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Tue, 29 Oct 2019 15:32:06 +0000
Subject: [OpenSIPS-Users] Branch Flags
In-Reply-To: <fe425512-dac6-5d56-ee8b-1fc796fe987e@opensips.org>
References: <4AE92CBD-31B1-46AF-A0B9-2762E4429BA1@genesys.com>
 <fe425512-dac6-5d56-ee8b-1fc796fe987e@opensips.org>
Message-ID: <DB0357FF-5221-4F6C-AF75-B03507849C5D@genesys.com>

Razvan,

Thanks for the quick response. I'm testing a workaround using the $bavp variable type exported by the TM module.

I think there was some work planned in 3.x to make the acceptance of variables more uniform in the scripting language. Do you think this could have been resolved already by that effort? If not, do you think it might be a reasonable feature request?

Ben Newlin 

On 10/29/19, 11:24 AM, "Users on behalf of Răzvan Crainea" <users-bounces at lists.opensips.org on behalf of razvan at opensips.org> wrote:

    Hi, Ben!
    
    Unfortunately the `setbflag` function only accepts static indexes, 
    that's why it doesn't work no matter how you try.
    Perhaps the only scenario where a static branch is useful is when you 
    "manually" add branches in your script (using append_branch), and you 
    know exactly the order those branches were added, and how to index them.
    
    I believe I ran into this issue a while ago, and I was able to solve it 
    by using an AVP, and indexing it using the branch index. I know it's not 
    the best way to do it, but it worked for me back then :)
    
    Best regards,
    
    Răzvan Crainea
    OpenSIPS Core Developer
    http://www.opensips-solutions.com
    
    On 10/29/19 3:20 PM, Ben Newlin wrote:
    > Hey all,
    > 
    > I am trying to use the branch flags functionality described here [1]. I 
    > am running into an issue when using the syntax with the branch_idx. The 
    > documentation does not provide any examples for this form.
    > 
    > Basically, from my testing in 2.4.6 it appears that the branch_idx can 
    > only be a static number. I have tried numerous forms to try to get a 
    > variable to be accepted but all result in syntax errors. Is this 
    > expected? If so, is anyone successfully using this functionality? I am 
    > having trouble thinking of a scenario where the flags could be useful if 
    > the branch numbers must be hard-coded in the script.
    > 
    > Works:
    > 
    > setbflag(1, FLAG)
    > 
    > Syntax Error:
    > 
    > setbflag($T_branch_idx, FLAG)
    > 
    > setbflag(“$T_branch_idx”, FLAG)
    > 
    > setbflag($T_branch_idx, “FLAG”)
    > 
    > setbflag(“$T_branch_idx”, “FLAG”)
    > 
    > [1] - https://www.opensips.org/Documentation/Script-Flags-2-4#toc4
    > 
    > Ben Newlin
    > 
    > 
    > _______________________________________________
    > 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  Tue Oct 29 11:49:38 2019
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 29 Oct 2019 17:49:38 +0200
Subject: [OpenSIPS-Users] Branch Flags
In-Reply-To: <DB0357FF-5221-4F6C-AF75-B03507849C5D@genesys.com>
References: <4AE92CBD-31B1-46AF-A0B9-2762E4429BA1@genesys.com>
 <fe425512-dac6-5d56-ee8b-1fc796fe987e@opensips.org>
 <DB0357FF-5221-4F6C-AF75-B03507849C5D@genesys.com>
Message-ID: <94ad895c-7ff2-3789-4f12-872e71cc3436@opensips.org>

Yes, the entire core functions interface has been reworked in 3.1, so 
this should have been solved now.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 10/29/19 5:32 PM, Ben Newlin wrote:
> Razvan,
> 
> Thanks for the quick response. I'm testing a workaround using the $bavp variable type exported by the TM module.
> 
> I think there was some work planned in 3.x to make the acceptance of variables more uniform in the scripting language. Do you think this could have been resolved already by that effort? If not, do you think it might be a reasonable feature request?
> 
> Ben Newlin
> 
> On 10/29/19, 11:24 AM, "Users on behalf of Răzvan Crainea" <users-bounces at lists.opensips.org on behalf of razvan at opensips.org> wrote:
> 
>      Hi, Ben!
>      
>      Unfortunately the `setbflag` function only accepts static indexes,
>      that's why it doesn't work no matter how you try.
>      Perhaps the only scenario where a static branch is useful is when you
>      "manually" add branches in your script (using append_branch), and you
>      know exactly the order those branches were added, and how to index them.
>      
>      I believe I ran into this issue a while ago, and I was able to solve it
>      by using an AVP, and indexing it using the branch index. I know it's not
>      the best way to do it, but it worked for me back then :)
>      
>      Best regards,
>      
>      Răzvan Crainea
>      OpenSIPS Core Developer
>      http://www.opensips-solutions.com
>      
>      On 10/29/19 3:20 PM, Ben Newlin wrote:
>      > Hey all,
>      >
>      > I am trying to use the branch flags functionality described here [1]. I
>      > am running into an issue when using the syntax with the branch_idx. The
>      > documentation does not provide any examples for this form.
>      >
>      > Basically, from my testing in 2.4.6 it appears that the branch_idx can
>      > only be a static number. I have tried numerous forms to try to get a
>      > variable to be accepted but all result in syntax errors. Is this
>      > expected? If so, is anyone successfully using this functionality? I am
>      > having trouble thinking of a scenario where the flags could be useful if
>      > the branch numbers must be hard-coded in the script.
>      >
>      > Works:
>      >
>      > setbflag(1, FLAG)
>      >
>      > Syntax Error:
>      >
>      > setbflag($T_branch_idx, FLAG)
>      >
>      > setbflag(“$T_branch_idx”, FLAG)
>      >
>      > setbflag($T_branch_idx, “FLAG”)
>      >
>      > setbflag(“$T_branch_idx”, “FLAG”)
>      >
>      > [1] - https://www.opensips.org/Documentation/Script-Flags-2-4#toc4
>      >
>      > Ben Newlin
>      >
>      >
>      > _______________________________________________
>      > 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 virendra at cloud-connect.in  Wed Oct 30 03:56:44 2019
From: virendra at cloud-connect.in (Virendra Bhati)
Date: Wed, 30 Oct 2019 13:26:44 +0530
Subject: [OpenSIPS-Users] Opensips crash while connecting mariadb
Message-ID: <CAB=dTEeKNT7w9fDZMAqZmORTSJGOT5QK3RZAcLmpiWU2vHPL1A@mail.gmail.com>

Dear Team,

We are using opensips 3.0.1 with mariadb 10.4.8 . We are facing issue at
starting opensips. As opensips crash with below given message:

Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found <db_bind_api> in
module db_mysql [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for db_mysql
Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not
found in pool
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno:
2000

I have attached logs and backtrace from core  so please provide your
suggestions. Please note credentials given in db_url are working while
using from mysql client


--
Regards
Virendra Bhati
Lead- Architecture and Software Solutions
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Oct 15 03:27:17 [22849] INFO:core:evi_publish_event: Registered event <E_DLG_STATE_CHANGED(11)>
Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found <load_tm> in module tm [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found <load_rr> in module rr [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found <db_bind_api> in module db_mysql [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for db_mysql
Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not found in pool
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000

Got ERROR: "InnoDB: Operating system error number 11 in a file operation." errno: 2000
Got ERROR: "InnoDB: Error number 11 means 'Resource temporarily unavailable'" errno: 2000
Got ERROR: "InnoDB: Cannot open datafile '/var/lib/mysql/ibdata1'" errno: 2000
Got ERROR: "InnoDB: Could not open or create the system tablespace. If you tried to add new data files to the system tablespace, and it failed here, you should now edit innodb_data_file_path in my.cnf back to what it was, and remove the new ibdata files InnoDB created in this failed attempt. InnoDB only wrote those files full of zeros, but did not yet use them in any way. But be careful: do not remove old data files which contain your precious data!" errno: 2000
Got ERROR: "InnoDB: Plugin initialization aborted with error Cannot open a file" errno: 2000
Got ERROR: "Plugin 'InnoDB' init function returned error." errno: 2000
Got ERROR: "Plugin 'InnoDB' registration as a STORAGE ENGINE failed." errno: 2000
Got ERROR: "unknown: Can't lock aria control file '/var/lib/mysql/aria_log_control' for exclusive use, error: 11. Will retry for 30 seconds" errno: 2000



^C^CGot ERROR: "unknown: Got error 'Could not get an exclusive lock; file is probably in use by another process' when trying to use aria control file '/var/lib/mysql/aria_log_control'" errno: 2000
Got ERROR: "Plugin 'Aria' init function returned error." errno: 2000
Got ERROR: "Plugin 'Aria' registration as a STORAGE ENGINE failed." errno: 2000
Got ERROR: "Unknown/unsupported storage engine: InnoDB" errno: 2000
 DBG:db_mysql:db_mysql_connect: opening connection: mysql://xxxx:xxxx at localhost/cc_master
CRITICAL:core:sig_usr: segfault in attendant (starter) process!
DBG:core:restore_segv_handler: restoring SIGSEGV handler...
 DBG:core:restore_segv_handler: successfully restored system SIGSEGV handler
^CSegmentation fault (core dumped)

-------------- next part --------------
(gdb) bt full
#0  intern_plugin_lock (lex=0x0, state_mask=14, rc=0x0)
    at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:948
        pi = 0x0
#1  plugin_thdvar_init (thd=0x28af578) at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:3155
        old_table_plugin = 0x0
        old_tmp_table_plugin = 0x0
        old_enforced_table_plugin = 0x0
#2  0x00007fc7d58722b1 in THD::init (this=this at entry=0x28af578)
    at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:1177
No locals.
#3  0x00007fc7d587310a in THD::THD (this=0x28af578, id=<optimized out>, is_wsrep_applier=<optimized out>)
    at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:798
        tmp = <optimized out>
#4  0x00007fc7d57f2bdc in create_embedded_thd (client_flag=client_flag at entry=-2143837683)
    at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/lib_sql.cc:685
        thd = 0x7fff689c20b0
#5  0x00007fc7d57fa1e4 in mysql_real_connect (mysql=0x7fc7da7190b0, host=<optimized out>, user=<optimized out>, 
    passwd=<optimized out>, db=0x7fc7da717620 "cc_master", port=port at entry=0, unix_socket=unix_socket at entry=0x0, 
    client_flag=2151129613, client_flag at entry=2147549184)
    at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/libmysqld.c:179
        name_buff = "\200\314|\326\307\177\000\000\060!\234h\377\177\000\000 !\234h\377\177\000\000\060!\234h\377\177\000\000\200\314|\326\307\177\000\000\001\000\000\000\000\000\000\000\340r\242\000\000\000\000\000p\251\215\000\000\000\000\000\340r\242\000\000\000\000\000\270\"\234h\377\177\000\000 vq\332\307\177\000\000\362\344s\333\307\177\000\000\000\000\000\000\000\000\000\000\060\000\000\000\060\000\000\000\060\"\234h\377\177\000\000P!\234h\377\177", '\000' <repeats 26 times>, "\224*\207\333\307\177\000\000\005+\207\333\307\177\000\000\300\061\253\333\307\177\000\000\200\363\252\333\307\177\000\000c\226u\333\307\177\000\000 vq\332\307\177\000\000"...
#6  0x00007fc7d70e8a63 in db_mysql_connect (ptr=ptr at entry=0x7fc7da717668) at my_con.c:105
        reconnect = 0 '\000'
---Type <return> to continue, or q <return> to quit---
        __FUNCTION__ = "db_mysql_connect"
#7  0x00007fc7d70e94ff in db_mysql_new_connection (id=0x7fc7da717510) at my_con.c:165
        ptr = 0x7fc7da717668
        __FUNCTION__ = "db_mysql_new_connection"
#8  0x00000000005ec4ba in db_do_init (url=<optimized out>, 
    new_connection=0x7fc7d70e9417 <db_mysql_new_connection>) at db/db.c:338
        id = 0x7fc7da717510
        con = <optimized out>
        res = 0x7fc7da717468
        con_size = <optimized out>
        __FUNCTION__ = "db_do_init"
#9  0x00007fc7d01044cd in dlg_connect_db (db_url=db_url at entry=0x7fc7d0340610 <db_url>) at dlg_db_handler.c:135
        __FUNCTION__ = "dlg_connect_db"
#10 0x00007fc7d0104511 in init_dlg_db (db_url=db_url at entry=0x7fc7d0340610 <db_url>, 
    dlg_hash_size=<optimized out>, db_update_period=60) at dlg_db_handler.c:176
        __FUNCTION__ = "init_dlg_db"
#11 0x00007fc7d0100c27 in mod_init () at dialog.c:900
        n = <optimized out>
        __FUNCTION__ = "mod_init"
#12 0x000000000050bfa7 in init_mod (m=0x7fc7da6face0, skip_others=skip_others at entry=0) at sr_module.c:697
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#13 0x000000000050c028 in init_mod (m=0x7fc7da6fb970, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#14 0x000000000050c028 in init_mod (m=0x7fc7da6fbd28, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
---Type <return> to continue, or q <return> to quit---
#15 0x000000000050c028 in init_mod (m=0x7fc7da6fbeb8, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#16 0x000000000050c028 in init_mod (m=0x7fc7da6fc0c0, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#17 0x000000000050c028 in init_mod (m=0x7fc7da6fc340, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#18 0x000000000050c028 in init_mod (m=0x7fc7da6fc4d8, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#19 0x000000000050c028 in init_mod (m=0x7fc7da6fd488, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#20 0x000000000050c028 in init_mod (m=0x7fc7da6fde20, skip_others=skip_others at entry=0) at sr_module.c:678
        dep = <optimized out>
        __FUNCTION__ = "init_mod"
#21 0x000000000050f45d in init_modules () at sr_module.c:759
        currentMod = 0x0
        ret = <optimized out>
        __FUNCTION__ = "init_modules"
#22 0x0000000000420351 in main (argc=<optimized out>, argv=<optimized out>) at main.c:1421
        c = <optimized out>
        r = <optimized out>
        tmp = 0x1 <Address 0x1 out of bounds>
        tmp_len = <optimized out>
        port = <optimized out>
---Type <return> to continue, or q <return> to quit---
        proto = <optimized out>
        protos_no = <optimized out>
        options = 0x67e9f8 "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"
        ret = -1
        seed = 961253727
        rfd = <optimized out>
        __FUNCTION__ = "main"


From liviu at opensips.org  Wed Oct 30 11:34:19 2019
From: liviu at opensips.org (Liviu Chircu)
Date: Wed, 30 Oct 2019 11:34:19 -0400
Subject: [OpenSIPS-Users] Example of configuration "Full Sharing"
 Topology with NoSQL
In-Reply-To: <7d4b9348-1366-9e4c-c1d6-053d347c9e39@bohboh.info>
References: <CAEBOSMQYy690uanmBJj76CaeLvKjdDaotGAia4usodP0DepKaA@mail.gmail.com>
 <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org>
 <1572250895.648618260@f112.i.mail.ru>
 <7d4b9348-1366-9e4c-c1d6-053d347c9e39@bohboh.info>
Message-ID: <33579d6f-a9ef-6e4d-5ab9-535450c25714@opensips.org>

Can you rather tell us more about the problem you are trying to solve, 
rather than
inquiring about various (random) solutions?  For example:

* will your platform have 1 POP or multiple POPs?
* do you want high availability for the user location nodes?

Regards,

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

On 10/28/19 10:27 AM, Social Boh wrote:
>
> Hello,
>
> if i don't want use a SBC how can I known is a node of cluster is up 
> or down? which method do you advise?
>
> Thank you
>