From a.shabana at cequens.com Sat Jul 1 08:39:31 2017 From: a.shabana at cequens.com (Ahmed Shabana) Date: Sat, 1 Jul 2017 12:39:31 +0000 Subject: [OpenSIPS-Users] Route per user agent instead of prefix Message-ID: Dears at opensips users list, I need to route per authenticated user instead of prefix based routing What is the best way to do this ? Br, AShabana From goup2010 at gmail.com Sun Jul 2 15:20:50 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Sun, 2 Jul 2017 22:20:50 +0300 Subject: [OpenSIPS-Users] Problem: Registration Proxy with WebRTC Message-ID: Hello, I try to setup Registration proxy (opensips 2.3) like this article: https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ 1. When use Zoiper all is OK Zoiper <----> Opensips REGISTRATION Proxy <-----> SBC 2. I have problem when use WebRTC. (*Via sent-by in the response does not match UA Via host value. Dropping the response*): SIP.JS <----->Opensips REGISTRATION Proxy <-----> SBC Here is WebSocket text messages: REGISTER sip: SIP/2.0 Via: SIP/2.0/WSS 192.0.2.148;branch=z9hG4bK8881457 Max-Forwards: 70 To: "Tester" > From: "Tester" >;tag=u6aro6a8mj Call-ID: k5uhq12e1bb93rg9igvpvv CSeq: 83 REGISTER Contact: ;reg-id=1;+sip.instance="";expires=60 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 sip-0.7.8.js:2900 Sun Jul 02 2017 18:41:19 GMT+0300 (FLE Daylight Time) | sip.transport | received WebSocket text message: SIP/2.0 401 Unauthorized Via: SIP/2.0/WSS 192.0.2.148:5060 ;rport=53162;received=;branch=z9hG4bK8881457 From: "Tester" >;tag=u6aro6a8mj To: "Tester" > Call-ID: k5uhq12e1bb93rg9igvpvv CSeq: 83 REGISTER Contact: :53162;transport=wss>;reg-id=1;+sip.instance="";expires=60 WWW-Authenticate: Digest realm="sbc.com", nonce=" ee228f001f1459108000000c2916c1ef at sbc.com" Content-Length: 0 sip-0.7.8.js:2900 Sun Jul 02 2017 18:41:19 GMT+0300 (FLE Daylight Time) | sip.sanitycheck | *Via sent-by in the response does not match UA Via host value. Dropping the response* Best regards, Drgagomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From max.muehlbronner at 42com.com Mon Jul 3 04:38:25 2017 From: max.muehlbronner at 42com.com (=?iso-8859-1?Q?Max_M=FChlbronner?=) Date: Mon, 3 Jul 2017 08:38:25 +0000 Subject: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Message-ID: Hi, I've never noticed this until i came across it recently. I got a weird issue with drouting, it turned out that even though the gatewaylist ("carrier") contains a total of 20 gateways, only 12 are being used. (all gateways got the same weight) E.g. if all gws are rejecting the calls, it will cycle through the gatewaylist but it never tries all of the gateways, only 12. Is there an internal limitation for the number of gateways? I know there is a limitation due to the database scheme, and there is also DR_MAX_GWLIST in the drouting.c module. Any idea why i am only able to "failover" 12 gateways in a carrier/gatewaylist? BR Max Muehlbronner -------------- next part -------------- An HTML attachment was scrubbed... URL: From max.muehlbronner at 42com.com Mon Jul 3 05:19:33 2017 From: max.muehlbronner at 42com.com (=?iso-8859-1?Q?Max_M=FChlbronner?=) Date: Mon, 3 Jul 2017 09:19:33 +0000 Subject: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? In-Reply-To: References: Message-ID: I think i got it, default limit is 12 branches in config.h. Which corresponds to my limit of 12 gateways, I will try and report back. :) #define MAX_BRANCHES 12 /*!< maximum number of branches per transaction */ Max Mühlbronner ------ 42com Telecommunication GmbH Straße der Pariser Kommune 12-16 10243 Berlin Fon: +49-(0)30-2434299-28 Fax: +49-(0)30-2434299-99 E-Mail: mm at 42com.com Web: www.42com.com Firmenangaben/Company information: Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B Umsatzsteuer-ID/VAT-ID: DE223812306 Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese sind möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail. This message is intended only for the use of the individual or entity to which it is addressed. If you have received this message by mistake, please notify us immediately. ________________________________ Von: Users im Auftrag von Max Mühlbronner Gesendet: Montag, 3. Juli 2017 10:38:25 An: OpenSIPS users mailling list Betreff: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Hi, I've never noticed this until i came across it recently. I got a weird issue with drouting, it turned out that even though the gatewaylist ("carrier") contains a total of 20 gateways, only 12 are being used. (all gateways got the same weight) E.g. if all gws are rejecting the calls, it will cycle through the gatewaylist but it never tries all of the gateways, only 12. Is there an internal limitation for the number of gateways? I know there is a limitation due to the database scheme, and there is also DR_MAX_GWLIST in the drouting.c module. Any idea why i am only able to "failover" 12 gateways in a carrier/gatewaylist? BR Max Muehlbronner -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 3 06:08:42 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 3 Jul 2017 13:08:42 +0300 Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway In-Reply-To: References: <8581f584-81ec-0f69-864d-466476d04311@opensips.org> <63f86d1a-518b-ccba-3a67-6cba8f74ce6c@opensips.org> <3277451d-2830-db7b-5da2-cfc955a04c5e@opensips.org> Message-ID: <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> Hi Alex, As suspected, the ACK is not properly routed - see the retransmissions of the 200OK + ACK. SImply based on the logs I cannot see what the problem is - probably some missing fix_nated_contact() for the replies coming from the WS party. Please make a pcap capture + opensips log (level 4) and send them to me *offlist* ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/30/2017 05:37 PM, Alex Megalokonomos wrote: > I have attached the debug log so you get a fuller picture. I hope > that's ok > > (Incoming call to WS client 694 is the WS extension...610 is my normal > desk phone which is connected to OmniPCX) (10.0.1.63-> OpenSIPS > ,10.0.1.200-> OmniPCX) > > > > On Fri, Jun 30, 2017 at 5:20 PM, Bogdan-Andrei Iancu > > wrote: > > Good, there is some progress :). > > On the incoming calls, if the WS get's the call, we can park the > part with the auth (it seems your opensips script is accepting > calls from unknown sources...we can address this security hole later. > > So, if a call drop after 30 secs it usually means there is no ACK. > Can you make a mgrep capture on OpenSIPS to grab the whole call > flow ? (grab 5060 and 80 ports) > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 06/30/2017 04:52 PM, Alex Megalokonomos wrote: >> I think I set up uac_registrant correctly. >> I can dial out from a ws client and the ws extension rings from >> outside calls. >> However: a) on incoming calls, when ws client accepts, there is >> no sound and the line is dropped after 30 secs or so >> b) on outgoing calls, when the called extension accepts the ws >> client immediately responds with 401 Unauthorised and then BYE >> b) I believe is what you mentioned here "In OpenSIPS, when >> receiving calls, you need to authorize (by IP) the calls from >> OmniPCX " >> How do I do this? >> and a) seems to be rtp proxy related since I see the following >> errors in the logs¨ >> ERROR:rtpengine:rtpe_function_call: proxy replied with error: >> Unknown call-id >> and >> no matching transaction >> On Fri, Jun 30, 2017 at 2:27 PM, Bogdan-Andrei Iancu >> > wrote: >> >> I checked the script you mentioned and it does not help you - >> it has only UDP (no WS), it is really basic and it does not >> handle any REGISTER stuff, which is the trickiest - see >> https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ >> >> or >> https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/ >> >> Maybe you can start with handling REGISTERs - what you need >> (on top of the script from the WSS tutorial) is to add this >> uac_registrant, to have the WS extensions registered into >> OmniPCX with a contact URI pointing back to OpenSIPS IP: >> http://www.opensips.org/html/docs/modules/2.3.x/uac_registrant.html >> >> Let me know if you get stuck in this first step. Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> >> On 06/30/2017 12:22 PM, Alex Megalokonomos wrote: >>> Hello Bogdan, >>> First of all, thanks for your time. >>> Unfortunately my SIP/OpensSIPS skills are what I've managed >>> to learn in the last couple of days. I am a programmer but >>> I've never had to work on SIP stuff before. >>> Frankly to me, both solutions sound equally difficult since >>> I have no idea where to start. (And to be honest, I expected >>> the first to be simpler) >>> I found this >>> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/ >>> >>> and tried to port the config to OpenSIPS since from what I >>> understand Kamailio and OpenSIPS share a common codebase to >>> an extent but was unsuccesful. >>> In your second scenario, I am not interested in WS->WS >>> calls so that auth part is not an issue. >>> So I guess I need the uac_registrar, authorize by IP and >>> usrloc parts. >>> Any relevant documentation to get me started since I'm still >>> not clear on what I need to change? >>> Best regards, >>> Alex >>> On Fri, Jun 30, 2017 at 11:29 AM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Alex, To make a kind of WS<>UDP gateway you need a >>> complete rework of the script presented in the tutorial, >>> as it is a completely different SIP scenario. Not sure >>> what are your SIP/OpenSIPS skills. But, there is a >>> simpler alternative . Instead of a GW, you can make >>> OpenSIPS as a sub-server for the WS extensions: >>> Registration handling: 1) WS extensions register only >>> with OpenSIPS (as right now) - authentication is done by >>> OpenSIPS 2) OpenSIPS registers the 3 extensions into >>> OmniPCX using the uac_registrar By this, we simply add >>> the uac_registration and you achieve kind of decoupled 2 >>> steps registration (with a minimum change in the cfg) >>> Inbound calls: 1) OmniPCX will send all the calls (from >>> other extensions) for the WS extension to OpenSIPS (due >>> the registration via uac_registrar) - this is default >>> behavior , so nothing to change 2) In OpenSIPS, when >>> receiving calls, you need to authorize (by IP) the calls >>> from OmniPCX - and as the current script does, you will >>> handle them via the local opensips usrloc -> calls are >>> sent to WS extension Outbound calls: 1) when you receive >>> a call from a WS extension, you have to check if the >>> call is for a local extension (on opensips) or for an >>> extension in OmniPCX 2) if call is local (WS to WS) you >>> will do authentication for the call 3) if the call is to >>> be sent to OmniPCX, simply send the call to OmniPCX >>> without auth - the auth will be done by OmniPCX as for >>> any other extension Hopefully this will work for you :) >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> >>> On 06/29/2017 11:54 AM, Alex Megalokonomos wrote: >>>> Hello Bogdan, >>>> Yes, a gateway from WS to UDP (as well as DTLS-SRTP to >>>> RTP in order for it to work) is exactly what we're >>>> looking for. >>>> Unfortunately our Alcatel OmniPCX call center is a >>>> proprietary system that only allows for a limited >>>> number of SIP extensions (served from what appears to >>>> be an outdated customised Kamailio 3.2.2 from what I >>>> can tell from the headers. >>>> For our normal internal office use it all works fine. >>>> However we have 3 customer support lines that are >>>> currently routed to 3 extensions via OmniPCX. >>>> We want to integrate these to our custom web-based CRM >>>> and the best way for us to do it is to use something >>>> like SIP js to handle and log calls, identify calling >>>> parties, bring up customer details etc. >>>> Since the kamailio version inside OmniPCX does not >>>> support ws/webrtc we are looking to set up Opensips in >>>> exactly the way you described as a gateway/proxy for >>>> everything in order to convert the UDP-only sip >>>> extensions to ws+ webRTC capable ones. >>>> I have used this tutorial >>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 >>>> >>>> to get what I assume is half the work (for RTP >>>> proxying) but I havent figured out the rest yet. >>>> Best regards, >>>> Alex >>>> On Thu, Jun 29, 2017 at 11:43 AM, Bogdan-Andrei Iancu >>>> > wrote: >>>> >>>> Hi Alex, First, some questions regarding the >>>> desired topology: 1) the WS end-points should >>>> register in OpenSIPS or all the way into Kamailio ? >>>> 2) also, the calls from the WS end-points >>>> should be all the time sent to Kamailio ? More or >>>> less, what I'm asking is : is OpenSIPS suppose to >>>> act as a gateway from WS to UDP , but pass all the >>>> resulting traffic to Kamailio ? Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> >>>> On 06/28/2017 12:47 PM, Alex Megalokonomos wrote: >>>>> Hello, >>>>> We have the following scenario: our office call >>>>> center is an Alcatel OmniPCX Office setup. >>>>> This handles most of our needs and also provides 4 >>>>> SIP extensions. >>>>> These are provided by what appears to be a >>>>> Kamailio SIP server v 3.2.2 (no webrtc or >>>>> websockets support) >>>>> What we would like to do is set up an OpenSIPS >>>>> instance to handle WebRTC and proxy everything to >>>>> this Kamailio SIP server. >>>>> The idea is to allow a web client (using sip js or >>>>> something similar) to register / make / receive >>>>> calls as one of the Kamailio extensions. >>>>> I think half of the configuration is this : >>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 >>>>> >>>>> which I've already completed and indeed, clients >>>>> can register to opensips and chat/make calls over >>>>> websockets between them. >>>>> How do I go about proxying >>>>> registrations/invites/etc to the kamailio server >>>>> instead? >>>>> best regards >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 3 06:12:16 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 3 Jul 2017 13:12:16 +0300 Subject: [OpenSIPS-Users] Route per user agent instead of prefix In-Reply-To: References: Message-ID: Hi, Put an extra fields in subscriber table in order to assign to each user an routing group - and after auth (see load_credentialas in auth_db module for loading extra columns from subscriber table), use the routing group value in order to trigger the corresponding routing logic. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/01/2017 03:39 PM, Ahmed Shabana wrote: > Dears at opensips users list, I need to route per authenticated user instead of prefix based routing > > What is the best way to do this ? > > Br, > AShabana > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Jul 3 06:18:42 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 3 Jul 2017 13:18:42 +0300 Subject: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? In-Reply-To: References: Message-ID: Hi Max, Yes, a transaction cannot have more than 12 branches used. But this does not limit how many GWs you can put in Dynamic Routing - you can put as many as you want on DR, but of course only 12 will be tried. But this depends a lot on the GW selection algorithm (with wights, with carriers, etc). To actually end up trying 12 gw for a call is a very rare case, I would say :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/03/2017 12:19 PM, Max Mühlbronner wrote: > > I think i got it, default limit is 12 branches in config.h. Which > corresponds to my limit of 12 gateways, I will try and report back. :) > > > > #define MAX_BRANCHES 12 /*!< maximum number of > branches per transaction */ > > > Max Mühlbronner > ------ > 42com Telecommunication GmbH* > *Straße der Pariser Kommune 12-16 > 10243 Berlin > > Fon: +49-(0)30-2434299-28 > Fax: +49-(0)30-2434299-99 > E-Mail: mm at 42com.com > Web: _www.42com.com _ > > > Firmenangaben/Company information: > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306 > Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. > Diese sind möglicherweise vertraulich und ausschließlich für den > Adressaten bestimmt. Sollten Sie diese elektronische Nachricht > irrtümlicherweise erhalten haben, so informieren Sie uns bitte > unverzüglich telefonisch oder per E-Mail. > > This message is intended only for the use of the individual or entity > to which it is addressed. If you have received this message by > mistake, please notify us immediately. > > ------------------------------------------------------------------------ > *Von:* Users im Auftrag von Max > Mühlbronner > *Gesendet:* Montag, 3. Juli 2017 10:38:25 > *An:* OpenSIPS users mailling list > *Betreff:* [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum > number of gateways? > Hi, > > I've never noticed this until i came across it recently. I got a weird > issue with drouting, it turned out that even though the gatewaylist > ("carrier") contains a total of 20 gateways, only 12 are being used. > (all gateways got the same weight) > > E.g. if all gws are rejecting the calls, it will cycle through the > gatewaylist but it never tries all of the gateways, only 12. > > Is there an internal limitation for the number of gateways? I know > there is a limitation due to the database scheme, and there is also > DR_MAX_GWLIST in the drouting.c module. > > Any idea why i am only able to "failover" 12 gateways in a > carrier/gatewaylist? > > > BR > > Max Muehlbronner > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From max.muehlbronner at 42com.com Mon Jul 3 06:42:17 2017 From: max.muehlbronner at 42com.com (=?iso-8859-1?Q?Max_M=FChlbronner?=) Date: Mon, 3 Jul 2017 10:42:17 +0000 Subject: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? In-Reply-To: References: , Message-ID: Thanks, i will try raising max_branches to allow the use of more gateways. Yeah i know it seems to be unusual to "try" so many gateways, but the setup works with a lot of smaller instances which are immediately rejecting calls when full to allow a failover to the next instances. Load balancing would be a better solution, but is not feasible in my scenario. Also it seems that tm module additionally has a limit of 30 branches. Anyway it should be enough even for me. :) BR ________________________________ Von: Bogdan-Andrei Iancu Gesendet: Montag, 3. Juli 2017 12:18:42 An: OpenSIPS users mailling list; Max Mühlbronner Betreff: Re: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Hi Max, Yes, a transaction cannot have more than 12 branches used. But this does not limit how many GWs you can put in Dynamic Routing - you can put as many as you want on DR, but of course only 12 will be tried. But this depends a lot on the GW selection algorithm (with wights, with carriers, etc). To actually end up trying 12 gw for a call is a very rare case, I would say :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/03/2017 12:19 PM, Max Mühlbronner wrote: I think i got it, default limit is 12 branches in config.h. Which corresponds to my limit of 12 gateways, I will try and report back. :) #define MAX_BRANCHES 12 /*!< maximum number of branches per transaction */ Max Mühlbronner ------ 42com Telecommunication GmbH Straße der Pariser Kommune 12-16 10243 Berlin Fon: +49-(0)30-2434299-28 Fax: +49-(0)30-2434299-99 E-Mail: mm at 42com.com Web: www.42com.com Firmenangaben/Company information: Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B Umsatzsteuer-ID/VAT-ID: DE223812306 Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese sind möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail. This message is intended only for the use of the individual or entity to which it is addressed. If you have received this message by mistake, please notify us immediately. ________________________________ Von: Users im Auftrag von Max Mühlbronner Gesendet: Montag, 3. Juli 2017 10:38:25 An: OpenSIPS users mailling list Betreff: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Hi, I've never noticed this until i came across it recently. I got a weird issue with drouting, it turned out that even though the gatewaylist ("carrier") contains a total of 20 gateways, only 12 are being used. (all gateways got the same weight) E.g. if all gws are rejecting the calls, it will cycle through the gatewaylist but it never tries all of the gateways, only 12. Is there an internal limitation for the number of gateways? I know there is a limitation due to the database scheme, and there is also DR_MAX_GWLIST in the drouting.c module. Any idea why i am only able to "failover" 12 gateways in a carrier/gatewaylist? BR Max Muehlbronner _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 3 06:49:21 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 3 Jul 2017 13:49:21 +0300 Subject: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? In-Reply-To: References: Message-ID: Max, After moving to 30, just check the usage of the shm memory, as TM will start using more (the branches are statically part of the transactions, so the transactions will become larger with 30 branches). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/03/2017 01:42 PM, Max Mühlbronner wrote: > > Thanks, i will try raising max_branches to allow the use of more gateways. > > > Yeah i know it seems to be unusual to "try" so many gateways, but the > setup works with a lot of smaller instances which are immediately > rejecting calls when full to allow a failover to the next instances. > Load balancing would be a better solution, but is not feasible in my > scenario. > > > Also it seems that tm module additionally has a limit of 30 branches. > Anyway it should be enough even for me. :) > > > > BR > > ------------------------------------------------------------------------ > *Von:* Bogdan-Andrei Iancu > *Gesendet:* Montag, 3. Juli 2017 12:18:42 > *An:* OpenSIPS users mailling list; Max Mühlbronner > *Betreff:* Re: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum > number of gateways? > Hi Max, > > Yes, a transaction cannot have more than 12 branches used. But this > does not limit how many GWs you can put in Dynamic Routing - you can > put as many as you want on DR, but of course only 12 will be tried. > But this depends a lot on the GW selection algorithm (with wights, > with carriers, etc). To actually end up trying 12 gw for a call is a > very rare case, I would say :) > > Regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > On 07/03/2017 12:19 PM, Max Mühlbronner wrote: >> >> I think i got it, default limit is 12 branches in config.h. Which >> corresponds to my limit of 12 gateways, I will try and report back. :) >> >> >> >> #define MAX_BRANCHES 12 /*!< maximum number >> of branches per transaction */ >> >> >> Max Mühlbronner >> ------ >> 42com Telecommunication GmbH* >> *Straße der Pariser Kommune 12-16 >> 10243 Berlin >> >> Fon: +49-(0)30-2434299-28 >> Fax: +49-(0)30-2434299-99 >> E-Mail: mm at 42com.com >> Web: _www.42com.com_ >> >> >> Firmenangaben/Company information: >> Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B >> Umsatzsteuer-ID/VAT-ID: DE223812306 >> Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig >> >> Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. >> Diese sind möglicherweise vertraulich und ausschließlich für den >> Adressaten bestimmt. Sollten Sie diese elektronische Nachricht >> irrtümlicherweise erhalten haben, so informieren Sie uns bitte >> unverzüglich telefonisch oder per E-Mail. >> >> This message is intended only for the use of the individual or entity >> to which it is addressed. If you have received this message by >> mistake, please notify us immediately. >> >> ------------------------------------------------------------------------ >> *Von:* Users im Auftrag von Max >> Mühlbronner >> *Gesendet:* Montag, 3. Juli 2017 10:38:25 >> *An:* OpenSIPS users mailling list >> *Betreff:* [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum >> number of gateways? >> Hi, >> >> I've never noticed this until i came across it recently. I got a >> weird issue with drouting, it turned out that even though the >> gatewaylist ("carrier") contains a total of 20 gateways, only 12 are >> being used. (all gateways got the same weight) >> >> E.g. if all gws are rejecting the calls, it will cycle through the >> gatewaylist but it never tries all of the gateways, only 12. >> >> Is there an internal limitation for the number of gateways? I know >> there is a limitation due to the database scheme, and there is also >> DR_MAX_GWLIST in the drouting.c module. >> >> Any idea why i am only able to "failover" 12 gateways in a >> carrier/gatewaylist? >> >> >> BR >> >> Max Muehlbronner >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From max.muehlbronner at 42com.com Mon Jul 3 07:50:05 2017 From: max.muehlbronner at 42com.com (=?Windows-1252?Q?Max_M=FChlbronner?=) Date: Mon, 3 Jul 2017 11:50:05 +0000 Subject: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? In-Reply-To: References: , Message-ID: Thanks for the hint. I will keep an eye on it. BR Max M. ________________________________ Von: Bogdan-Andrei Iancu Gesendet: Montag, 3. Juli 2017 12:49:21 An: Max Mühlbronner; OpenSIPS users mailling list Betreff: Re: AW: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Max, After moving to 30, just check the usage of the shm memory, as TM will start using more (the branches are statically part of the transactions, so the transactions will become larger with 30 branches). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/03/2017 01:42 PM, Max Mühlbronner wrote: Thanks, i will try raising max_branches to allow the use of more gateways. Yeah i know it seems to be unusual to "try" so many gateways, but the setup works with a lot of smaller instances which are immediately rejecting calls when full to allow a failover to the next instances. Load balancing would be a better solution, but is not feasible in my scenario. Also it seems that tm module additionally has a limit of 30 branches. Anyway it should be enough even for me. :) BR ________________________________ Von: Bogdan-Andrei Iancu Gesendet: Montag, 3. Juli 2017 12:18:42 An: OpenSIPS users mailling list; Max Mühlbronner Betreff: Re: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Hi Max, Yes, a transaction cannot have more than 12 branches used. But this does not limit how many GWs you can put in Dynamic Routing - you can put as many as you want on DR, but of course only 12 will be tried. But this depends a lot on the GW selection algorithm (with wights, with carriers, etc). To actually end up trying 12 gw for a call is a very rare case, I would say :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/03/2017 12:19 PM, Max Mühlbronner wrote: I think i got it, default limit is 12 branches in config.h. Which corresponds to my limit of 12 gateways, I will try and report back. :) #define MAX_BRANCHES 12 /*!< maximum number of branches per transaction */ Max Mühlbronner ------ 42com Telecommunication GmbH Straße der Pariser Kommune 12-16 10243 Berlin Fon: +49-(0)30-2434299-28 Fax: +49-(0)30-2434299-99 E-Mail: mm at 42com.com Web: www.42com.com Firmenangaben/Company information: Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B Umsatzsteuer-ID/VAT-ID: DE223812306 Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese sind möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail. This message is intended only for the use of the individual or entity to which it is addressed. If you have received this message by mistake, please notify us immediately. ________________________________ Von: Users im Auftrag von Max Mühlbronner Gesendet: Montag, 3. Juli 2017 10:38:25 An: OpenSIPS users mailling list Betreff: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of gateways? Hi, I've never noticed this until i came across it recently. I got a weird issue with drouting, it turned out that even though the gatewaylist ("carrier") contains a total of 20 gateways, only 12 are being used. (all gateways got the same weight) E.g. if all gws are rejecting the calls, it will cycle through the gatewaylist but it never tries all of the gateways, only 12. Is there an internal limitation for the number of gateways? I know there is a limitation due to the database scheme, and there is also DR_MAX_GWLIST in the drouting.c module. Any idea why i am only able to "failover" 12 gateways in a carrier/gatewaylist? BR Max Muehlbronner _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 3 11:46:59 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 3 Jul 2017 18:46:59 +0300 Subject: [OpenSIPS-Users] SIP URI User Parameters In-Reply-To: References: <208A1714-5990-44FD-AA64-073AC633E249@genesys.com> Message-ID: <131bc2e6-8b7b-071e-8fff-6a97442effcd@opensips.org> Hi Ben, Thank you for your digging and reporting. Following your leads I found some old strange behavior of the parse_uri() function - the function responsible for parsing the URIs in OpenSISP. For some ancient and unknown reasons, a SIP URI with user=phone was automatically converted to a TEL URI. Such conversion, automatically done, is dangerous - there is nothing in the RFC3261 stating something like this. Even more, the conversion is not complete - besides moving the username parameters to URI parameters, the domain is not stripped and the TEL not added. Basically, the existing code was converting: sip:username;bla=foo at host.com;param1=1;param2=2;user=phone to sip:username at host.com;bla=foo I tried to dig around the subject, but not more - there is no reference or recommendation for such a behavior. If you have the time, see these links: * SIP implementer -> https://lists.cs.columbia.edu/pipermail/sip-implementors/2013-February/028837.html * SIP Core -> https://www.ietf.org/mail-archive/web/sipcore/current/msg01783.html * voip info -> https://www.voip-info.org/wiki/view/SIP+URI (Telephone numbers section) On voip-info there is a recommendation on how to compare a SIP uri with a TEL uri (in terms of username and parameters parts), but nothing of a "must" conversion. So, I disabled the guilty code in OpenSIPS, and it should work as expected now. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/30/2017 10:14 PM, Ben Newlin wrote: > > Bogdan, > > I have been able to reproduce this locally now. The piece that was > missing is that the Request URI must already have URI parameters on > it. If it has both URI and user parameters, the call to > route_to_carrier (and possibly do_routing) replaces all of the URI > parameters with the user parameter(s). > > Original $ru: > > sip:+15555551212;npdi=yes at gw2.com;transport=udp;user=phone > > After call to route_to_carrier: > > sip:+15555551212 at gw2.com;npdi=yes > > After further testing, it appears this behavior is not restricted to > the Dynamic Routing module after all. Simply modifying $ru while both > user and URI parameters are present causes the issue. > > Original $ru: > > sip:+15555551212;npdi=yes at gw2.com;transport=udp;user=phone > > Perform this action: > > $rU = $rU + “;rn=+15555550000”; > > Resultant $ru: > > sip:+15555551212;rn=+15555550000 at gw2.com;npdi=yes > > Ben Newlin > > Lead Voice Network Engineer, PureCloud > > ** > > O+1 317.957.1009 > > _ben.newlin at genesys.com _ > > > > > > *From: *Users on behalf of Ben > Newlin > *Reply-To: *OpenSIPS users mailling list > *Date: *Friday, June 30, 2017 at 10:47 AM > *To: *Bogdan-Andrei Iancu , OpenSIPS users > mailling list > *Subject: *Re: [OpenSIPS-Users] SIP URI User Parameters > > Bogdan, > > Sorry for the delayed response, I am having some trouble reproducing > this in a local test environment. Currently it is only occurring in > our live environment. I do have some clarifications and answers to > your questions: > > ·The npdi parameter is not present in $ru in the failure route when > the response is 500. It is present when the response is 503 or 408. I > haven’t tested any other responses. This is not terribly important to > my issue, simply an observation. > > > ·We are sometimes using do_routing to populate a list of carriers, but > other times we get the list from our own DB query. We use > route_to_carrier to send the call to each carrier in sequence. This is > because we don’t always use do_routing, but also because we wish to > skip to the next carrier, not just the next gateway, on certain > response codes and the normal do_routing mechanism doesn’t allow that. > > > ·The issue actually does not happen when use_next_gw is called. I was > wrong about that. You were right that seems to be a straight URI copy. > The issue occurs when we skip use_next_gw or there are no gateways > left and we call route_to_carrier for the next carrier with the > parameter present in $ru. > > > ·I printed out the dr_ruri avp after the call to route_to_carrier and > it shows the npdi parameter moved to the end, not after the user: > “sip:+15555551212 at gw2.com;npdi=yes” > > Also, I should have mentioned that we are running 1.11.11. I’m still > working to try to reproduce locally. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Thursday, June 29, 2017 at 4:38 AM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] SIP URI User Parameters > > Hello Ben > > I understand you add the npdi useraname parameter after performing the > initial do_routing() - if you do it in request or branch route is not > relevant (for RURI changes) as RURI is anyhow a per-branch value. > In failure route, when resuming, you will get the RURI of the winning > branch ( the one which was selected to be sent back to caller), so you > see the npdi param. > > So far so good. And now you do use_next_gw() in failure route and you > get "sip:+15555551212 at gw2.com;npdi" directly, without any another npdi > addition ? I'm asking, as use_next_gw() does a full RURI replacement > (it doesn;t care what is the existing RURI). > > Could you also do an > xlog("DR ruris are <$(avp(___dr_ruri__)[*])>\n"); > right after do_routing() ? > > Regards, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 06/28/2017 11:41 PM, Ben Newlin wrote: > > Hi, > > We have run into an issue with OpenSIPs’ handling of user > parameters in SIP URIs with Dynamic Routing module. When a > parameter is added to a SIP URI user part, any subsequent > modification of the URI by DR module results in the parameter > being moved to be a URI parameter. > > For example, starting with $ru of “sip:+15555551212 at gw1.com” > , if we modify it this way: > > $rU = $rU + “;npdi”; > > then we get a new $ru of “sip:+15555551212;npdi at gw1.com” > . > > We send this call out and if it returns an error we want to use > the next available gateway. > > The Request URI in the failure route is still > “sip:+15555551212;npdi at gw1.com” > . > > Note: this is the case even when the “;npdi” parameter was added > in a branch route, which I didn’t expect. I thought changes made > in a branch route were isolated to that branch. > > Now from the failure route when we call use_next_gw the npdi > parameter is moved and the URI is now > “sip:+15555551212 at gw2.com;npdi” > . This is not correct. > > Is there some other way to properly manipulate SIP URI user > parameters or is this a bug? > > Thanks, > > Ben Newlin > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 4709 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 7162 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2054 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 2071 bytes Desc: not available URL: From bogdan at opensips.org Mon Jul 3 12:55:29 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 3 Jul 2017 19:55:29 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?OpenSIPS_=40_ClueCon_2017_=E2=80=93_Tr?= =?utf-8?q?aining_and_more?= Message-ID: <9e028e28-8085-0e72-d949-246a08fb83c5@opensips.org> This year we will be part of ClueCon once again ! https://blog.opensips.org/2017/07/03/opensips-cluecon-2017-training-and-more/ See you in Chicago, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html From carlos.oliva at numintec.com Tue Jul 4 06:44:54 2017 From: carlos.oliva at numintec.com (Carlos Oliva) Date: Tue, 4 Jul 2017 12:44:54 +0200 Subject: [OpenSIPS-Users] sip related question: via and contact headers with 0.0.0.0 Message-ID: Hi List: This question is SIP related, I'm writing to the list just in case any SIP expert can help. I'm facing random cases where some phones send to my OpenSips proxy a REGISTER with VIA and CONTACT host part with 0.0.0.0 address. As far as I know this is not right, I can not see any similar in RFCs. It happens mainly with Grandstream phones, but I saw this with 3CX softphones the last week. The headers look like: Via: SIP/2.0/TLS 0.0.0.0:30201;branch=z9hG4bK1519166325;rport;alias Contact: ;reg-id=1;+sip.instance="" The issue happens using UDP and TLS protocol, I can discard a firewall ALG The expires header is at configured value (1200 or 120 seconds) and all other headers (from, to, callId, ... ) seems to be OK My questions is: This is right according RFC or is a bug on those phones? Maybe is right and I'm doing something wrong in my OpenSips config? Thanks for your help, Carlos Oliva -------------- next part -------------- An HTML attachment was scrubbed... URL: From monkeilas at gmail.com Tue Jul 4 07:07:47 2017 From: monkeilas at gmail.com (=?UTF-8?Q?Andreas_B=C3=B8ckmann?=) Date: Tue, 4 Jul 2017 13:07:47 +0200 Subject: [OpenSIPS-Users] B2B not relaying 180 in prepaid scenario Message-ID: Hello I am playing around with B2B and running OpenSIPS proxy and B2B on the same VM. I am triggering prepaid scenario on initial INVITEs for authenticated clients. https://www.opensips.org/Documentation/Tutorials-B2BUA#toc13 Now; everything seems to work OK except for the fact that 180 is not relayed and no ringing is ever heard on the A-side after listening to Media and while connecting to B-side. It seems to somehow be swallowed by B2B. It's passed to B2B which seems to not handle 180 while in bridging scenario? DBG:tm:local_reply: Passing provisional reply 180 to FIFO application .... DBG:b2b_logic:b2b_logic_notify_reply: Received a reply [180] while in BRIDGING scenario Even though A-side is connected (after listening to media) it would make sense to let the A-side play ringing while trying to reach the B-side. Any ideas of how I can solve this? The OpenSIPS log for handling 180 can be found here: https://pastebin.com/fPVgLrCG Thanks a lot for your kind support! //Andreas -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Jul 4 09:02:54 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 4 Jul 2017 18:32:54 +0530 Subject: [OpenSIPS-Users] Query regarding forking in opensips-2.2 . Message-ID: Hi All , I am using opensips-2.2 . Case 1 : I have 2 contact registered with different port . When an INVITE comes , opensips is doing parallel forking Its sending INVITE to both the contacts with same Branch header and different request URI . But I am expecting Branch header should be different in forking . I guess according to RFC in forking the Branch header changes . Can anybody confirm whether I am doing something wrong or there is issue in this version . Please assist me . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Jul 4 09:03:42 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 4 Jul 2017 18:33:42 +0530 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 Message-ID: Hi All . While saving contact , I have written save("location","c1fp1") . It means when a new register comes that will get added and the old contact will get de-reg in its expire time . But , if 1 contact is existing , and another 2 register comes at the same time with different contact(same Ip different port ) and same Username then what will happen . According to my observation , opensips is saving both the new contacts . Is that write . While I am forcing a single contact this should not happen . Please assist me if I am doing anything wrong . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 11:07:49 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 18:07:49 +0300 Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway In-Reply-To: <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> References: <8581f584-81ec-0f69-864d-466476d04311@opensips.org> <63f86d1a-518b-ccba-3a67-6cba8f74ce6c@opensips.org> <3277451d-2830-db7b-5da2-cfc955a04c5e@opensips.org> <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> Message-ID: Hi Alex, Thank you for the offlist provided data. Shortly, the ACK received by OpenSIPS from OmniPCX is broken as it is missing all the Route headers. According to the pcap, it looks like: ACK sip:udoioiia at 10.0.1.106:49246;transport=ws SIP/2.0 Record-Route: Contact: "Megalokonomos A." User-Agent: OxO_SPG_103/012.001 Content-Type: application/sdp To: sip:694 at 10.0.1.200;tag=4em4m1ah9r From: "Megalokonomos A." ;tag=d5de999de446df5165d773dac1f369ec Call-ID: af3cc9085db1c8dd86050eb91d747249 at 10.0.1.200 CSeq: 659214613 ACK Via: SIP/2.0/UDP 10.0.1.200:5059;branch=z9hG4bKf3de.2fc1fc65cece765af47f9baf8bf0906e.0;i=c Via: SIP/2.0/TCP 10.0.1.200:5080;rport=45698;branch=z9hG4bK89fca3417cd4e227b4315145d96657c7 Max-Forwards: 69 Content-Length: 2960 v=0 o=default 14 ..... As OpenSIPS does not find the Route (former Record-Route) it inserted into the dialog, the routing logic in the script does not work as expected. According to RFC3261, the RR headers MUST be mirrored back in 2xx replies. Let's try to hack to cope with the broken SIP stack onOmniPCX. In script you have something like: } else { # ACK without matching transaction -> # ignore and discard exit; } Try replacing it with } else { # ACK without matching transaction -> # ignore and discard t_relay(); exit; } Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/03/2017 01:08 PM, Bogdan-Andrei Iancu wrote: > Hi Alex, > > As suspected, the ACK is not properly routed - see the > retransmissions of the 200OK + ACK. SImply based on the logs I cannot > see what the problem is - probably some missing fix_nated_contact() > for the replies coming from the WS party. > > Please make a pcap capture + opensips log (level 4) and send them to > me *offlist* ! > > Best regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > On 06/30/2017 05:37 PM, Alex Megalokonomos wrote: >> I have attached the debug log so you get a fuller picture. I hope >> that's ok >> >> (Incoming call to WS client 694 is the WS extension...610 is my >> normal desk phone which is connected to OmniPCX) (10.0.1.63-> >> OpenSIPS ,10.0.1.200-> OmniPCX) >> >> >> >> On Fri, Jun 30, 2017 at 5:20 PM, Bogdan-Andrei Iancu >> wrote: >> >> Good, there is some progress :). >> >> On the incoming calls, if the WS get's the call, we can park the >> part with the auth (it seems your opensips script is accepting >> calls from unknown sources...we can address this security hole later. >> >> So, if a call drop after 30 secs it usually means there is no >> ACK. Can you make a mgrep capture on OpenSIPS to grab the whole >> call flow ? (grab 5060 and 80 ports) >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> >> On 06/30/2017 04:52 PM, Alex Megalokonomos wrote: >>> I think I set up uac_registrant correctly. >>> I can dial out from a ws client and the ws extension rings from >>> outside calls. >>> However: a) on incoming calls, when ws client accepts, there is >>> no sound and the line is dropped after 30 secs or so >>> b) on outgoing calls, when the called extension accepts the ws >>> client immediately responds with 401 Unauthorised and then BYE >>> b) I believe is what you mentioned here "In OpenSIPS, when >>> receiving calls, you need to authorize (by IP) the calls from >>> OmniPCX " >>> How do I do this? >>> and a) seems to be rtp proxy related since I see the following >>> errors in the logs¨ >>> ERROR:rtpengine:rtpe_function_call: proxy replied with error: >>> Unknown call-id >>> and >>> no matching transaction >>> On Fri, Jun 30, 2017 at 2:27 PM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> I checked the script you mentioned and it does not help you >>> - it has only UDP (no WS), it is really basic and it does >>> not handle any REGISTER stuff, which is the trickiest - see >>> https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ >>> >>> or >>> https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/ >>> >>> Maybe you can start with handling REGISTERs - what you need >>> (on top of the script from the WSS tutorial) is to add this >>> uac_registrant, to have the WS extensions registered into >>> OmniPCX with a contact URI pointing back to OpenSIPS IP: >>> http://www.opensips.org/html/docs/modules/2.3.x/uac_registrant.html >>> >>> Let me know if you get stuck in this first step. Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> >>> On 06/30/2017 12:22 PM, Alex Megalokonomos wrote: >>>> Hello Bogdan, >>>> First of all, thanks for your time. >>>> Unfortunately my SIP/OpensSIPS skills are what I've managed >>>> to learn in the last couple of days. I am a programmer but >>>> I've never had to work on SIP stuff before. >>>> Frankly to me, both solutions sound equally difficult since >>>> I have no idea where to start. (And to be honest, I >>>> expected the first to be simpler) >>>> I found this >>>> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/ >>>> >>>> and tried to port the config to OpenSIPS since from what I >>>> understand Kamailio and OpenSIPS share a common codebase to >>>> an extent but was unsuccesful. >>>> In your second scenario, I am not interested in WS->WS >>>> calls so that auth part is not an issue. >>>> So I guess I need the uac_registrar, authorize by IP and >>>> usrloc parts. >>>> Any relevant documentation to get me started since I'm >>>> still not clear on what I need to change? >>>> Best regards, >>>> Alex >>>> On Fri, Jun 30, 2017 at 11:29 AM, Bogdan-Andrei Iancu >>>> > wrote: >>>> >>>> Hi Alex, To make a kind of WS<>UDP gateway you need a >>>> complete rework of the script presented in the >>>> tutorial, as it is a completely different SIP scenario. >>>> Not sure what are your SIP/OpenSIPS skills. But, there >>>> is a simpler alternative . Instead of a GW, you can >>>> make OpenSIPS as a sub-server for the WS extensions: >>>> Registration handling: 1) WS extensions register only >>>> with OpenSIPS (as right now) - authentication is done >>>> by OpenSIPS 2) OpenSIPS registers the 3 extensions into >>>> OmniPCX using the uac_registrar By this, we simply add >>>> the uac_registration and you achieve kind of decoupled >>>> 2 steps registration (with a minimum change in the cfg) >>>> Inbound calls: 1) OmniPCX will send all the calls (from >>>> other extensions) for the WS extension to OpenSIPS (due >>>> the registration via uac_registrar) - this is default >>>> behavior , so nothing to change 2) In OpenSIPS, when >>>> receiving calls, you need to authorize (by IP) the >>>> calls from OmniPCX - and as the current script does, >>>> you will handle them via the local opensips usrloc -> >>>> calls are sent to WS extension Outbound calls: 1) when >>>> you receive a call from a WS extension, you have to >>>> check if the call is for a local extension (on >>>> opensips) or for an extension in OmniPCX 2) if call is >>>> local (WS to WS) you will do authentication for the >>>> call 3) if the call is to be sent to OmniPCX, simply >>>> send the call to OmniPCX without auth - the auth will >>>> be done by OmniPCX as for any other extension Hopefully >>>> this will work for you :) Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> >>>> On 06/29/2017 11:54 AM, Alex Megalokonomos wrote: >>>>> Hello Bogdan, >>>>> Yes, a gateway from WS to UDP (as well as DTLS-SRTP to >>>>> RTP in order for it to work) is exactly what we're >>>>> looking for. >>>>> Unfortunately our Alcatel OmniPCX call center is a >>>>> proprietary system that only allows for a limited >>>>> number of SIP extensions (served from what appears to >>>>> be an outdated customised Kamailio 3.2.2 from what I >>>>> can tell from the headers. >>>>> For our normal internal office use it all works fine. >>>>> However we have 3 customer support lines that are >>>>> currently routed to 3 extensions via OmniPCX. >>>>> We want to integrate these to our custom web-based CRM >>>>> and the best way for us to do it is to use something >>>>> like SIP js to handle and log calls, identify calling >>>>> parties, bring up customer details etc. >>>>> Since the kamailio version inside OmniPCX does not >>>>> support ws/webrtc we are looking to set up Opensips in >>>>> exactly the way you described as a gateway/proxy for >>>>> everything in order to convert the UDP-only sip >>>>> extensions to ws+ webRTC capable ones. >>>>> I have used this tutorial >>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 >>>>> >>>>> to get what I assume is half the work (for RTP >>>>> proxying) but I havent figured out the rest yet. >>>>> Best regards, >>>>> Alex >>>>> On Thu, Jun 29, 2017 at 11:43 AM, Bogdan-Andrei Iancu >>>>> > wrote: >>>>> >>>>> Hi Alex, First, some questions regarding the >>>>> desired topology: 1) the WS end-points should >>>>> register in OpenSIPS or all the way into Kamailio >>>>> ? 2) also, the calls from the WS end-points >>>>> should be all the time sent to Kamailio ? More or >>>>> less, what I'm asking is : is OpenSIPS suppose to >>>>> act as a gateway from WS to UDP , but pass all the >>>>> resulting traffic to Kamailio ? Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developer >>>>> http://www.opensips-solutions.com >>>>> >>>>> >>>>> OpenSIPS Bootcamp 2017, Houston, US >>>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>>> >>>>> >>>>> On 06/28/2017 12:47 PM, Alex Megalokonomos wrote: >>>>>> Hello, >>>>>> We have the following scenario: our office call >>>>>> center is an Alcatel OmniPCX Office setup. >>>>>> This handles most of our needs and also provides >>>>>> 4 SIP extensions. >>>>>> These are provided by what appears to be a >>>>>> Kamailio SIP server v 3.2.2 (no webrtc or >>>>>> websockets support) >>>>>> What we would like to do is set up an OpenSIPS >>>>>> instance to handle WebRTC and proxy everything to >>>>>> this Kamailio SIP server. >>>>>> The idea is to allow a web client (using sip js >>>>>> or something similar) to register / make / >>>>>> receive calls as one of the Kamailio extensions. >>>>>> I think half of the configuration is this : >>>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 >>>>>> >>>>>> which I've already completed and indeed, clients >>>>>> can register to opensips and chat/make calls over >>>>>> websockets between them. >>>>>> How do I go about proxying >>>>>> registrations/invites/etc to the kamailio server >>>>>> instead? >>>>>> best regards >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at clockwork.gr Tue Jul 4 11:22:04 2017 From: alex at clockwork.gr (Alex Megalokonomos) Date: Tue, 4 Jul 2017 18:22:04 +0300 Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway In-Reply-To: References: <8581f584-81ec-0f69-864d-466476d04311@opensips.org> <63f86d1a-518b-ccba-3a67-6cba8f74ce6c@opensips.org> <3277451d-2830-db7b-5da2-cfc955a04c5e@opensips.org> <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> Message-ID: As you may have noticed in my last reply, I reached that far as well but got stuck later on on what appears to be the rtp engine configuration. Not strictly an Opensips issue but you might be able to help me. On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu wrote: > Hi Alex, > > Thank you for the offlist provided data. Shortly, the ACK received by > OpenSIPS from OmniPCX is broken as it is missing all the Route headers. > According to the pcap, it looks like: > > ACK sip:udoioiia at 10.0.1.106:49246;transport=ws SIP/2.0 > Record-Route: ec;lr=on> > Contact: "Megalokonomos A." > User-Agent: OxO_SPG_103/012.001 > Content-Type: application/sdp > To: sip:694 at 10.0.1.200;tag=4em4m1ah9r > From: "Megalokonomos A." ;tag= > d5de999de446df5165d773dac1f369ec > Call-ID: af3cc9085db1c8dd86050eb91d747249 at 10.0.1.200 > CSeq: 659214613 ACK > Via: SIP/2.0/UDP 10.0.1.200:5059;branch=z9hG4bKf3de. > 2fc1fc65cece765af47f9baf8bf0906e.0;i=c > Via: SIP/2.0/TCP 10.0.1.200:5080;rport=45698;branch= > z9hG4bK89fca3417cd4e227b4315145d96657c7 > Max-Forwards: 69 > Content-Length: 2960 > > v=0 > o=default 14 > ..... > > > As OpenSIPS does not find the Route (former Record-Route) it inserted into > the dialog, the routing logic in the script does not work as expected. > According to RFC3261, the RR headers MUST be mirrored back in 2xx replies. > > Let's try to hack to cope with the broken SIP stack on OmniPCX. In script > you have something like: > > } else { > # ACK without matching transaction -> > # ignore and discard > exit; > } > > Try replacing it with > > } else { > # ACK without matching transaction -> > # ignore and discard > t_relay(); > exit; > } > > Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/03/2017 01:08 PM, Bogdan-Andrei Iancu wrote: > > Hi Alex, > > As suspected, the ACK is not properly routed - see the retransmissions of > the 200OK + ACK. SImply based on the logs I cannot see what the problem is > - probably some missing fix_nated_contact() for the replies coming from the > WS party. > > Please make a pcap capture + opensips log (level 4) and send them to me > *offlist* ! > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 06/30/2017 05:37 PM, Alex Megalokonomos wrote: > > I have attached the debug log so you get a fuller picture. I hope that's > ok > > (Incoming call to WS client 694 is the WS extension...610 is my normal > desk phone which is connected to OmniPCX) (10.0.1.63-> OpenSIPS > ,10.0.1.200-> OmniPCX) > > > > On Fri, Jun 30, 2017 at 5:20 PM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Good, there is some progress :). >> >> On the incoming calls, if the WS get's the call, we can park the part >> with the auth (it seems your opensips script is accepting calls from >> unknown sources...we can address this security hole later. >> >> So, if a call drop after 30 secs it usually means there is no ACK. Can >> you make a mgrep capture on OpenSIPS to grab the whole call flow ? (grab >> 5060 and 80 ports) >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 06/30/2017 04:52 PM, Alex Megalokonomos wrote: >> >> I think I set up uac_registrant correctly. >> I can dial out from a ws client and the ws extension rings from outside >> calls. >> However: a) on incoming calls, when ws client accepts, there is no sound >> and the line is dropped after 30 secs or so >> b) on outgoing calls, when the called extension accepts the ws client >> immediately responds with 401 Unauthorised and then BYE >> b) I believe is what you mentioned here "In OpenSIPS, when receiving >> calls, you need to authorize (by IP) the calls from OmniPCX " >> How do I do this? >> and a) seems to be rtp proxy related since I see the following errors in >> the logs¨ >> ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown >> call-id >> and >> no matching transaction >> On Fri, Jun 30, 2017 at 2:27 PM, Bogdan-Andrei Iancu > > wrote: >>> >>> I checked the script you mentioned and it does not help you - it has >>> only UDP (no WS), it is really basic and it does not handle any REGISTER >>> stuff, which is the trickiest - see https://blog.opensips.org/2016 >>> /12/13/how-to-proxy-sip-registrations/ or >>> https://blog.opensips.org/2016/12/20/mid-registrar-scalable- >>> registration-and-call-forking/ Maybe you can start with handling >>> REGISTERs - what you need (on top of the script from the WSS tutorial) is >>> to add this uac_registrant, to have the WS extensions registered into OmniPCX >>> with a contact URI pointing back to OpenSIPS IP: >>> http://www.opensips.org/html/docs/modules/2.3.x/uac_registrant.html Let >>> me know if you get stuck in this first step. Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> On 06/30/2017 12:22 PM, Alex Megalokonomos wrote: >>> >>> Hello Bogdan, >>> First of all, thanks for your time. >>> Unfortunately my SIP/OpensSIPS skills are what I've managed to learn in >>> the last couple of days. I am a programmer but I've never had to work on >>> SIP stuff before. >>> Frankly to me, both solutions sound equally difficult since I have no >>> idea where to start. (And to be honest, I expected the first to be simpler) >>> I found this https://blog.voipxswitch.com/2015/03/27/kamailio-basic- >>> sip-proxy-all-requests-setup/ and tried to port the config to OpenSIPS >>> since from what I understand Kamailio and OpenSIPS share a common codebase >>> to an extent but was unsuccesful. >>> In your second scenario, I am not interested in WS->WS calls so that >>> auth part is not an issue. >>> So I guess I need the uac_registrar, authorize by IP and usrloc parts. >>> Any relevant documentation to get me started since I'm still not clear >>> on what I need to change? >>> Best regards, >>> Alex >>> On Fri, Jun 30, 2017 at 11:29 AM, Bogdan-Andrei Iancu < >>> bogdan at opensips.org> wrote: >>>> >>>> Hi Alex, To make a kind of WS<>UDP gateway you need a complete rework >>>> of the script presented in the tutorial, as it is a completely different >>>> SIP scenario. Not sure what are your SIP/OpenSIPS skills. But, there is a >>>> simpler alternative . Instead of a GW, you can make OpenSIPS as a >>>> sub-server for the WS extensions: Registration handling: 1) WS extensions >>>> register only with OpenSIPS (as right now) - authentication is done by >>>> OpenSIPS 2) OpenSIPS registers the 3 extensions into OmniPCX using the >>>> uac_registrar By this, we simply add the uac_registration and you achieve >>>> kind of decoupled 2 steps registration (with a minimum change in the cfg) >>>> Inbound calls: 1) OmniPCX will send all the calls (from other >>>> extensions) for the WS extension to OpenSIPS (due the registration via >>>> uac_registrar) - this is default behavior , so nothing to change 2) In >>>> OpenSIPS, when receiving calls, you need to authorize (by IP) the calls >>>> from OmniPCX - and as the current script does, you will handle them >>>> via the local opensips usrloc -> calls are sent to WS extension Outbound >>>> calls: 1) when you receive a call from a WS extension, you have to check if >>>> the call is for a local extension (on opensips) or for an extension in OmniPCX >>>> 2) if call is local (WS to WS) you will do authentication for the call 3) >>>> if the call is to be sent to OmniPCX, simply send the call to OmniPCX >>>> without auth - the auth will be done by OmniPCX as for any other >>>> extension Hopefully this will work for you :) Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> On 06/29/2017 11:54 AM, Alex Megalokonomos wrote: >>>> >>>> Hello Bogdan, >>>> Yes, a gateway from WS to UDP (as well as DTLS-SRTP to RTP in order for >>>> it to work) is exactly what we're looking for. >>>> Unfortunately our Alcatel OmniPCX call center is a proprietary system >>>> that only allows for a limited number of SIP extensions (served from what >>>> appears to be an outdated customised Kamailio 3.2.2 from what I can tell >>>> from the headers. >>>> For our normal internal office use it all works fine. >>>> However we have 3 customer support lines that are currently routed to 3 >>>> extensions via OmniPCX. >>>> We want to integrate these to our custom web-based CRM and the best way >>>> for us to do it is to use something like SIP js to handle and log calls, >>>> identify calling parties, bring up customer details etc. >>>> Since the kamailio version inside OmniPCX does not support ws/webrtc we >>>> are looking to set up Opensips in exactly the way you described as a >>>> gateway/proxy for everything in order to convert the UDP-only sip >>>> extensions to ws+ webRTC capable ones. >>>> I have used this tutorial http://www.opensips.o >>>> rg/Documentation/Tutorials-WebSocket-2-1 to get what I assume is half >>>> the work (for RTP proxying) but I havent figured out the rest yet. >>>> Best regards, >>>> Alex >>>> On Thu, Jun 29, 2017 at 11:43 AM, Bogdan-Andrei Iancu < >>>> bogdan at opensips.org> wrote: >>>>> >>>>> Hi Alex, First, some questions regarding the desired topology: 1) >>>>> the WS end-points should register in OpenSIPS or all the way into Kamailio >>>>> ? 2) also, the calls from the WS end-points should be all the time sent >>>>> to Kamailio ? More or less, what I'm asking is : is OpenSIPS suppose to act >>>>> as a gateway from WS to UDP , but pass all the resulting traffic to >>>>> Kamailio ? Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developer >>>>> http://www.opensips-solutions.com >>>>> >>>>> OpenSIPS Bootcamp 2017, Houston, US >>>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>>> >>>>> On 06/28/2017 12:47 PM, Alex Megalokonomos wrote: >>>>> >>>>> Hello, >>>>> We have the following scenario: our office call center is an Alcatel >>>>> OmniPCX Office setup. >>>>> This handles most of our needs and also provides 4 SIP extensions. >>>>> These are provided by what appears to be a Kamailio SIP server v 3.2.2 >>>>> (no webrtc or websockets support) >>>>> What we would like to do is set up an OpenSIPS instance to handle >>>>> WebRTC and proxy everything to this Kamailio SIP server. >>>>> The idea is to allow a web client (using sip js or something similar) >>>>> to register / make / receive calls as one of the Kamailio extensions. >>>>> I think half of the configuration is this : >>>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 >>>>> which I've already completed and indeed, clients can register to >>>>> opensips and chat/make calls over websockets between them. >>>>> How do I go about proxying registrations/invites/etc to the kamailio >>>>> server instead? >>>>> best regards >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 11:57:09 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 18:57:09 +0300 Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway In-Reply-To: References: <8581f584-81ec-0f69-864d-466476d04311@opensips.org> <63f86d1a-518b-ccba-3a67-6cba8f74ce6c@opensips.org> <3277451d-2830-db7b-5da2-cfc955a04c5e@opensips.org> <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> Message-ID: Yeah, sorry, missed that one . Well, it seems that OmniPCX is doing late SDP negotiation (via 200OK + ACK, instead of INVITE+200OK) and the tutorial script does not handle this case (for simplicity and clarity reasons). So, right now the RTPengine interaction (the offer and answer) are done at INVITE and 200 OK time. You have to change a bit the OpenSIPS script to move the offer and answer on 200 OK and ACK if the INVITE has no SDP attached. Let me know if you need any assistance. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 06:22 PM, Alex Megalokonomos wrote: > As you may have noticed in my last reply, I reached that far as well > but got stuck later on on what appears to be the rtp engine > configuration. > > Not strictly an Opensips issue but you might be able to help me. > > On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Alex, > > Thank you for the offlist provided data. Shortly, the ACK received > by OpenSIPS from OmniPCX is broken as it is missing all the Route > headers. According to the pcap, it looks like: > > ACK sip:udoioiia at 10.0.1.106:49246;transport=ws SIP/2.0 > Record-Route: > > Contact: "Megalokonomos A." > User-Agent: OxO_SPG_103/012.001 > Content-Type: application/sdp > To: sip:694 at 10.0.1.200;tag=4em4m1ah9r > From: "Megalokonomos A." > ;tag=d5de999de446df5165d773dac1f369ec > Call-ID: af3cc9085db1c8dd86050eb91d747249 at 10.0.1.200 > > CSeq: 659214613 ACK > Via: SIP/2.0/UDP > 10.0.1.200:5059;branch=z9hG4bKf3de.2fc1fc65cece765af47f9baf8bf0906e.0;i=c > Via: SIP/2.0/TCP > 10.0.1.200:5080;rport=45698;branch=z9hG4bK89fca3417cd4e227b4315145d96657c7 > Max-Forwards: 69 > Content-Length: 2960 > > v=0 > o=default 14 > ..... > > > As OpenSIPS does not find the Route (former Record-Route) it > inserted into the dialog, the routing logic in the script does not > work as expected. According to RFC3261, the RR headers MUST be > mirrored back in 2xx replies. > > Let's try to hack to cope with the broken SIP stack onOmniPCX. In > script you have something like: > > } else { > # ACK without matching transaction -> > # ignore and discard > exit; > } > > Try replacing it with > > } else { > # ACK without matching transaction -> > # ignore and discard > t_relay(); > exit; > } > > Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/03/2017 01:08 PM, Bogdan-Andrei Iancu wrote: >> Hi Alex, As suspected, the ACK is not properly routed - see the >> retransmissions of the 200OK + ACK. SImply based on the logs I >> cannot see what the problem is - probably some missing >> fix_nated_contact() for the replies coming from the WS party. >> Please make a pcap capture + opensips log (level 4) and send them >> to me *offlist* ! Best regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 06/30/2017 05:37 PM, Alex Megalokonomos wrote: >>> I have attached the debug log so you get a fuller picture. I >>> hope that's ok >>> (Incoming call to WS client 694 is the WS extension...610 is my >>> normal desk phone which is connected to OmniPCX) (10.0.1.63-> >>> OpenSIPS ,10.0.1.200-> OmniPCX) -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at clockwork.gr Tue Jul 4 12:07:03 2017 From: alex at clockwork.gr (Alex Megalokonomos) Date: Tue, 4 Jul 2017 19:07:03 +0300 Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway In-Reply-To: References: <8581f584-81ec-0f69-864d-466476d04311@opensips.org> <63f86d1a-518b-ccba-3a67-6cba8f74ce6c@opensips.org> <3277451d-2830-db7b-5da2-cfc955a04c5e@opensips.org> <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> Message-ID: "You have to change a bit the OpenSIPS script to move the offer and answer on 200 OK and ACK if the INVITE has no SDP attached." If you could provide some pointers on this that would be great. I'm guessing the t_on_reply ("handle_nat") stays as is While the branch_route[handle_nat] logic needs to be moved to ACK. But how do I differentiate this ACK which is in response to the 200 ok to the invite compared to a different one? On Tue, Jul 4, 2017 at 6:57 PM, Bogdan-Andrei Iancu wrote: > Yeah, sorry, missed that one . > > Well, it seems that OmniPCX is doing late SDP negotiation (via 200OK + > ACK, instead of INVITE+200OK) and the tutorial script does not handle this > case (for simplicity and clarity reasons). > > So, right now the RTPengine interaction (the offer and answer) are done at > INVITE and 200 OK time. > > You have to change a bit the OpenSIPS script to move the offer and answer > on 200 OK and ACK if the INVITE has no SDP attached. > > Let me know if you need any assistance. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/04/2017 06:22 PM, Alex Megalokonomos wrote: > > As you may have noticed in my last reply, I reached that far as well but > got stuck later on on what appears to be the rtp engine configuration. > > Not strictly an Opensips issue but you might be able to help me. > > On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > >> Hi Alex, >> >> Thank you for the offlist provided data. Shortly, the ACK received by >> OpenSIPS from OmniPCX is broken as it is missing all the Route headers. >> According to the pcap, it looks like: >> >> ACK sip:udoioiia at 10.0.1.106:49246;transport=ws SIP/2.0 >> Record-Route: > lr=on> >> Contact: "Megalokonomos A." >> User-Agent: OxO_SPG_103/012.001 >> Content-Type: application/sdp >> To: sip:694 at 10.0.1.200;tag=4em4m1ah9r >> From: "Megalokonomos A." ;tag=d5de9 >> 99de446df5165d773dac1f369ec >> Call-ID: af3cc9085db1c8dd86050eb91d747249 at 10.0.1.200 >> CSeq: 659214613 ACK >> Via: SIP/2.0/UDP 10.0.1.200:5059;branch=z9hG4bK >> f3de.2fc1fc65cece765af47f9baf8bf0906e.0;i=c >> Via: SIP/2.0/TCP 10.0.1.200:5080;rport=45698;br >> anch=z9hG4bK89fca3417cd4e227b4315145d96657c7 >> Max-Forwards: 69 >> Content-Length: 2960 >> >> v=0 >> o=default 14 >> ..... >> >> >> As OpenSIPS does not find the Route (former Record-Route) it inserted >> into the dialog, the routing logic in the script does not work as expected. >> According to RFC3261, the RR headers MUST be mirrored back in 2xx replies. >> >> Let's try to hack to cope with the broken SIP stack on OmniPCX. In >> script you have something like: >> >> } else { >> # ACK without matching transaction -> >> # ignore and discard >> exit; >> } >> >> Try replacing it with >> >> } else { >> # ACK without matching transaction -> >> # ignore and discard >> t_relay(); >> exit; >> } >> >> Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/03/2017 01:08 PM, Bogdan-Andrei Iancu wrote: >> >> Hi Alex, As suspected, the ACK is not properly routed - see the >> retransmissions of the 200OK + ACK. SImply based on the logs I cannot see >> what the problem is - probably some missing fix_nated_contact() for the >> replies coming from the WS party. Please make a pcap capture + opensips log >> (level 4) and send them to me *offlist* ! Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 06/30/2017 05:37 PM, Alex Megalokonomos wrote: >> >> I have attached the debug log so you get a fuller picture. I hope that's >> ok >> (Incoming call to WS client 694 is the WS extension...610 is my normal >> desk phone which is connected to OmniPCX) (10.0.1.63-> OpenSIPS >> ,10.0.1.200-> OmniPCX) >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 12:11:58 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 19:11:58 +0300 Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway In-Reply-To: References: <8581f584-81ec-0f69-864d-466476d04311@opensips.org> <63f86d1a-518b-ccba-3a67-6cba8f74ce6c@opensips.org> <3277451d-2830-db7b-5da2-cfc955a04c5e@opensips.org> <341d8491-fa3c-e171-892a-c074a6e23fc9@opensips.org> Message-ID: <01e2646e-c635-1836-6415-72c2dd4c8a59@opensips.org> Send my your cfg offlist. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 07:07 PM, Alex Megalokonomos wrote: > > "You have to change a bit the OpenSIPS script to move the offer and > answer on 200 OK and ACK if the INVITE has no SDP attached." > > If you could provide some pointers on this that would be great. > > I'm guessing the t_on_reply ("handle_nat") stays as is > > While the branch_route[handle_nat] logic needs to be moved to ACK. But > how do I differentiate this ACK which is in response to the 200 ok to > the invite compared to a different one? > > On Tue, Jul 4, 2017 at 6:57 PM, Bogdan-Andrei Iancu > > wrote: > > Yeah, sorry, missed that one . > > Well, it seems that OmniPCX is doing late SDP negotiation (via > 200OK + ACK, instead of INVITE+200OK) and the tutorial script does > not handle this case (for simplicity and clarity reasons). > > So, right now the RTPengine interaction (the offer and answer) are > done at INVITE and 200 OK time. > > You have to change a bit the OpenSIPS script to move the offer and > answer on 200 OK and ACK if the INVITE has no SDP attached. > > Let me know if you need any assistance. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/04/2017 06:22 PM, Alex Megalokonomos wrote: >> As you may have noticed in my last reply, I reached that far as >> well but got stuck later on on what appears to be the rtp engine >> configuration. >> Not strictly an Opensips issue but you might be able to help me. >> On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Alex, Thank you for the offlist provided data. Shortly, >> the ACK received by OpenSIPS from OmniPCX is broken as it is >> missing all the Route headers. According to the pcap, it >> looks like:ACK sip:udoioiia at 10.0.1.106:49246;transport=ws >> SIP/2.0 Record-Route: >> >> Contact: "Megalokonomos A." >> User-Agent: OxO_SPG_103/012.001 Content-Type: application/sdp >> To: sip:694 at 10.0.1.200;tag=4em4m1ah9r From: "Megalokonomos >> A." ;tag=d5de999de446df5165d773dac1f369ec >> Call-ID: af3cc9085db1c8dd86050eb91d747249 at 10.0.1.200 >> CSeq: >> 659214613 ACK Via: SIP/2.0/UDP >> 10.0.1.200:5059;branch=z9hG4bKf3de.2fc1fc65cece765af47f9baf8bf0906e.0;i=c >> Via: SIP/2.0/TCP >> 10.0.1.200:5080;rport=45698;branch=z9hG4bK89fca3417cd4e227b4315145d96657c7 >> Max-Forwards: 69 Content-Length: 2960 v=0 o=default 14 ..... >> As OpenSIPS does not find the Route (former Record-Route) it >> inserted into the dialog, the routing logic in the script >> does not work as expected. According to RFC3261, the RR >> headers MUST be mirrored back in 2xx replies. Let's try to >> hack to cope with the broken SIP stack onOmniPCX. In script >> you have something like: >> >> } else { >> # ACK without matching transaction -> >> # ignore and discard >> exit; >> } >> >> Try replacing it with >> >> } else { >> # ACK without matching transaction -> >> # ignore and discard >> t_relay(); >> exit; >> } >> >> Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> >> On 07/03/2017 01:08 PM, Bogdan-Andrei Iancu wrote: >>> Hi Alex, As suspected, the ACK is not properly routed - see >>> the retransmissions of the 200OK + ACK. SImply based on the >>> logs I cannot see what the problem is - probably some >>> missing fix_nated_contact() for the replies coming from the >>> WS party. Please make a pcap capture + opensips log (level >>> 4) and send them to me *offlist* ! Best regards, >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> On 06/30/2017 05:37 PM, Alex Megalokonomos wrote: >>>> I have attached the debug log so you get a fuller picture. >>>> I hope that's ok >>>> (Incoming call to WS client 694 is the WS extension...610 >>>> is my normal desk phone which is connected to OmniPCX) >>>> (10.0.1.63-> OpenSIPS ,10.0.1.200-> OmniPCX) > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 12:21:43 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 19:21:43 +0300 Subject: [OpenSIPS-Users] sip related question: via and contact headers with 0.0.0.0 In-Reply-To: References: Message-ID: <69c56994-06bc-9a51-3b6f-8197026316da@opensips.org> Hi Carlos, I do not claim to be SIP expert, but it has no sense in my opinion. Even more, I haven't seen anything like that. So I would say it is a bug. My first reaction was to point to an ALG (that's something I came across), but if you do TLS this is out of discussion. Shortly, I'm not aware of any legitimate case when an UAC should send 0.0.0.0 in VIA or Contact. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 01:44 PM, Carlos Oliva wrote: > Hi List: > > This question is SIP related, I'm writing to the list just in case any > SIP expert can help. > > I'm facing random cases where some phones send to my OpenSips proxy a > REGISTER with VIA and CONTACT host part with 0.0.0.0 address. > > As far as I know this is not right, I can not see any similar in RFCs. > > It happens mainly with Grandstream phones, but I saw this with 3CX > softphones the last week. > > The headers look like: > > Via: SIP/2.0/TLS 0.0.0.0:30201;branch=z9hG4bK1519166325;rport;alias > > Contact: > ;reg-id=1;+sip.instance="" > > The issue happens using UDP and TLS protocol, I can discard a firewall ALG > > The expires header is at configured value (1200 or 120 seconds) and > all other headers (from, to, callId, ... ) seems to be OK > > My questions is: This is right according RFC or is a bug on those > phones? Maybe is right and I'm doing something wrong in my OpenSips > config? > > Thanks for your help, > > Carlos Oliva > > > > > > > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 12:43:08 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 19:43:08 +0300 Subject: [OpenSIPS-Users] B2B not relaying 180 in prepaid scenario In-Reply-To: References: Message-ID: <4999fd30-2029-cb7e-0d0b-b5daa3fa364c@opensips.org> Hi Andreas, Yes, in bridging mode (when one of the party was already connected to a previous entity), the provisional replies are not sent anymore (during the re-INVITE) as make no sense (for the already connected party). In your case, once the A side was connected (via 200 OK) to the media server, an incoming 180 on the re-INVITE (while bridging to the B side) will not induce a ringing tone at all. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 02:07 PM, Andreas Bøckmann wrote: > Hello > > I am playing around with B2B and running OpenSIPS proxy and B2B on the > same VM. > I am triggering prepaid scenario on initial INVITEs for authenticated > clients. > > https://www.opensips.org/Documentation/Tutorials-B2BUA#toc13 > > Now; everything seems to work OK except for the fact that 180 is not > relayed and no ringing is ever heard on the A-side after listening to > Media and while connecting to B-side. > > It seems to somehow be swallowed by B2B. It's passed to B2B which > seems to not handle 180 while in bridging scenario? > > DBG:tm:local_reply: Passing provisional reply 180 to FIFO application > .... > DBG:b2b_logic:b2b_logic_notify_reply: Received a reply [180] while in > BRIDGING scenario > > Even though A-side is connected (after listening to media) it would > make sense to let the A-side play ringing while trying to reach the > B-side. > > Any ideas of how I can solve this? > > The OpenSIPS log for handling 180 can be found here: > https://pastebin.com/fPVgLrCG > > Thanks a lot for your kind support! > > //Andreas > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 12:44:35 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 19:44:35 +0300 Subject: [OpenSIPS-Users] Query regarding forking in opensips-2.2 . In-Reply-To: References: Message-ID: Hi Sasmita, Well, you are doing something wrong. While doing forking via TM module, OpenSIPS generated different VIA branch params for each fork. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 04:02 PM, Sasmita Panda wrote: > Hi All , > > I am using opensips-2.2 . > > Case 1 : > I have 2 contact registered with different port . When an INVITE > comes , opensips is doing parallel forking Its sending INVITE to both > the contacts with same Branch header and different request URI . > But I am expecting Branch header should be different in forking > . I guess according to RFC in forking the Branch header changes . > > Can anybody confirm whether I am doing something wrong or there > is issue in this version . > > Please assist me . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 4 12:47:57 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 4 Jul 2017 19:47:57 +0300 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: Message-ID: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> Hi Sasmita, Have you checked via "opensipsctl ul show" to see that you actually have 2 registrations under the same AOR ?? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 04:03 PM, Sasmita Panda wrote: > Hi All . > > > While saving contact , I have written > save("location","c1fp1") . It means when a new register comes that > will get added and the old contact will get de-reg in its expire time . > > But , if 1 contact is existing , and another 2 register comes > at the same time with different contact(same Ip different port ) and > same Username then what will happen . > > According to my observation , opensips is saving both the new > contacts . Is that write . While I am forcing a single contact this > should not happen . > > Please assist me if I am doing anything wrong . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jul 5 01:27:59 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 5 Jul 2017 10:57:59 +0530 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> Message-ID: Actually I am not saving data in memory . It in DB only mode . So I am seeing in database . I get 2 contacts of same Username, From same IP and Different Port . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > Have you checked via "opensipsctl ul show" to see that you actually have 2 > registrations under the same AOR ?? > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/04/2017 04:03 PM, Sasmita Panda wrote: > > Hi All . > > > While saving contact , I have written save("location","c1fp1") . > It means when a new register comes that will get added and the old contact > will get de-reg in its expire time . > > But , if 1 contact is existing , and another 2 register comes at > the same time with different contact(same Ip different port ) and same > Username then what will happen . > > According to my observation , opensips is saving both the new > contacts . Is that write . While I am forcing a single contact this should > not happen . > > Please assist me if I am doing anything wrong . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jul 5 04:22:26 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 11:22:26 +0300 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> Message-ID: Hi Sasmita, Please post (do not attach to the email!) the opensips logs in log_level 4 corresponding to the processing of the second REGISTER requests (which adds the second contact). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 08:27 AM, Sasmita Panda wrote: > Actually I am not saving data in memory . It in DB only mode . So I am > seeing in database . I get 2 contacts of same Username, From same IP > and Different Port . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Sasmita, > > Have you checked via "opensipsctl ul show" to see that you > actually have 2 registrations under the same AOR ?? > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/04/2017 04:03 PM, Sasmita Panda wrote: >> Hi All . >> While saving contact , I have written >> save("location","c1fp1") . It means when a new register comes >> that will get added and the old contact will get de-reg in its >> expire time . >> But , if 1 contact is existing , and another 2 register >> comes at the same time with different contact(same Ip different >> port ) and same Username then what will happen . >> According to my observation , opensips is saving both the >> new contacts . Is that write . While I am forcing a single >> contact this should not happen . >> Please assist me if I am doing anything wrong . >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alameire at openip.fr Wed Jul 5 04:24:45 2017 From: alameire at openip.fr (Lameire Alexis) Date: Wed, 5 Jul 2017 10:24:45 +0200 Subject: [OpenSIPS-Users] Passive call recording using rtpproxy Message-ID: Hello, I would like to acheave call recording on a passive way. My setup will use a port mirroring on the switch on a dedicated port. Despite the presentation on the last summit, I can't properly understand how to properly acheave it. As I get rtpproxy will rewrite the SDP content to fix the media ip to the instance of rtpproxy, but this biavior is not suitable for a passive call recording. As I see on this[1] documentation I don't find a way to inhibate this biavior and just rely on the choiced RTP/RTSP port. In additions I will be faced by the issues related to port mirroring, the destination ip on sig and media packet will not match the local address and rtpproxy and opensips will not be able to bind the datagram socket. So, I think I have lost something, so if you could glow my mind you will be gracefully thanked. In addition of my questions, I would like to strongly thanks the presenter for is call recording experience, mainly due to the keys provided on the post task for files conversion. [1] http://www.opensips.org/html/docs/modules/1.7.x/rtpproxy.html -- Signature Alexis Lameire Ingénieur infrastructure +33 9 70 71 60 01 alameire at openip.fr 37/39, rue de Neuilly, 92110 Clichy http://openip.fr / http://my.openip.fr e-mail -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jul 5 04:30:39 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 11:30:39 +0300 Subject: [OpenSIPS-Users] Problem: Registration Proxy with WebRTC In-Reply-To: References: Message-ID: <77205cc0-6d65-64f9-c1a6-ee4de9c1674d@opensips.org> Hi Dragomir, Do you have any idea if the "Via sent-by" reported by SIP.JS is actually about the "received" parameter returned in VIA by OpenSIPS ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/02/2017 10:20 PM, Dragomir Haralambiev wrote: > Hello, > > I try to setup Registration proxy (opensips 2.3) like this article: > https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ > > 1. When use Zoiper all is OK > Zoiper <----> Opensips REGISTRATION Proxy <-----> SBC > > 2. I have problem when use WebRTC. > (*Via sent-by in the response does not match UA Via host value. > Dropping the response*): > > SIP.JS <----->Opensips REGISTRATION Proxy <-----> SBC > > Here is WebSocket text messages: > > REGISTER sip: SIP/2.0 > Via: SIP/2.0/WSS 192.0.2.148;branch=z9hG4bK8881457 > Max-Forwards: 70 > To: "Tester" > > From: "Tester" >;tag=u6aro6a8mj > Call-ID: k5uhq12e1bb93rg9igvpvv > CSeq: 83 REGISTER > Contact: ;transport=wss>;reg-id=1;+sip.instance="";expires=60 > Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER > Supported: path, gruu, outbound > User-Agent: SIP.js/0.7.8 > Content-Length: 0 > > > > sip-0.7.8.js:2900 Sun Jul 02 2017 18:41:19 GMT+0300 (FLE Daylight > Time) | sip.transport | received WebSocket text message: > > SIP/2.0 401 Unauthorized > Via: SIP/2.0/WSS > 192.0.2.148:5060;rport=53162;received=;branch=z9hG4bK8881457 > From: "Tester" >;tag=u6aro6a8mj > To: "Tester" > > Call-ID: k5uhq12e1bb93rg9igvpvv > CSeq: 83 REGISTER > Contact: > :53162;transport=wss>;reg-id=1;+sip.instance="";expires=60 > WWW-Authenticate: Digest realm="sbc.com ", > nonce="ee228f001f1459108000000c2916c1ef at sbc.com > " > Content-Length: 0 > > > > sip-0.7.8.js:2900 Sun Jul 02 2017 18:41:19 GMT+0300 (FLE Daylight > Time) | sip.sanitycheck | *Via sent-by in the response does not match > UA Via host value. Dropping the response* > > > Best regards, > Drgagomir > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 5 05:43:54 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 5 Jul 2017 12:43:54 +0300 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> Message-ID: Sasmita, could you please post the output of "opensips -V"? A similar bug on 2.2 has been discussed and fixed exactly one month ago [1] [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2017 08:27, Sasmita Panda wrote: > Actually I am not saving data in memory . It in DB only mode . So I am > seeing in database . I get 2 contacts of same Username, From same IP > and Different Port . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Sasmita, > > Have you checked via "opensipsctl ul show" to see that you > actually have 2 registrations under the same AOR ?? > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/04/2017 04:03 PM, Sasmita Panda wrote: >> Hi All . >> >> >> While saving contact , I have written >> save("location","c1fp1") . It means when a new register comes >> that will get added and the old contact will get de-reg in its >> expire time . >> >> But , if 1 contact is existing , and another 2 register >> comes at the same time with different contact(same Ip different >> port ) and same Username then what will happen . >> >> According to my observation , opensips is saving both the >> new contacts . Is that write . While I am forcing a single >> contact this should not happen . >> >> Please assist me if I am doing anything wrong . >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From carlos.oliva at numintec.com Wed Jul 5 05:56:42 2017 From: carlos.oliva at numintec.com (Carlos Oliva) Date: Wed, 5 Jul 2017 11:56:42 +0200 Subject: [OpenSIPS-Users] sip related question: via and contact headers with 0.0.0.0 In-Reply-To: <69c56994-06bc-9a51-3b6f-8197026316da@opensips.org> References: <69c56994-06bc-9a51-3b6f-8197026316da@opensips.org> Message-ID: Thank you very much for your response Bogdan. Probably is a bug in Grandstream Firmware. I will work with the manufacturer about this, but I was not completely sure if there was a corner case where this could be possible. Thanks for your confirmation. Thanks and regards, Carlos Oliva 2017-07-04 18:21 GMT+02:00 Bogdan-Andrei Iancu : > Hi Carlos, > > I do not claim to be SIP expert, but it has no sense in my opinion. Even > more, I haven't seen anything like that. So I would say it is a bug. > > My first reaction was to point to an ALG (that's something I came across), > but if you do TLS this is out of discussion. > > Shortly, I'm not aware of any legitimate case when an UAC should send > 0.0.0.0 in VIA or Contact. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/04/2017 01:44 PM, Carlos Oliva wrote: > > Hi List: > > This question is SIP related, I'm writing to the list just in case any SIP > expert can help. > > I'm facing random cases where some phones send to my OpenSips proxy a > REGISTER with VIA and CONTACT host part with 0.0.0.0 address. > > As far as I know this is not right, I can not see any similar in RFCs. > > It happens mainly with Grandstream phones, but I saw this with 3CX > softphones the last week. > > The headers look like: > > Via: SIP/2.0/TLS 0.0.0.0:30201;branch=z9hG4bK1519166325;rport;alias > > Contact: ;reg-id=1; > +sip.instance="" > > The issue happens using UDP and TLS protocol, I can discard a firewall ALG > > The expires header is at configured value (1200 or 120 seconds) and all > other headers (from, to, callId, ... ) seems to be OK > > My questions is: This is right according RFC or is a bug on those phones? > Maybe is right and I'm doing something wrong in my OpenSips config? > > Thanks for your help, > > Carlos Oliva > > > > > > > > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jul 5 06:14:12 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 5 Jul 2017 15:44:12 +0530 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> Message-ID: [root at localhost opensips]# opensips -V version: opensips 2.2.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 Do you want me to test after applying the patch ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu wrote: > Sasmita, could you please post the output of "opensips -V"? A similar bug > on 2.2 has been discussed and fixed exactly one month ago [1] > > [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 05.07.2017 08:27, Sasmita Panda wrote: > > Actually I am not saving data in memory . It in DB only mode . So I am > seeing in database . I get 2 contacts of same Username, From same IP and > Different Port . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu > wrote: > >> Hi Sasmita, >> >> Have you checked via "opensipsctl ul show" to see that you actually have >> 2 registrations under the same AOR ?? >> >> Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >> >> Hi All . >> >> >> While saving contact , I have written save("location","c1fp1") >> . It means when a new register comes that will get added and the old >> contact will get de-reg in its expire time . >> >> But , if 1 contact is existing , and another 2 register comes at >> the same time with different contact(same Ip different port ) and same >> Username then what will happen . >> >> According to my observation , opensips is saving both the new >> contacts . Is that write . While I am forcing a single contact this should >> not happen . >> >> Please assist me if I am doing anything wrong . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 5 06:18:12 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 5 Jul 2017 13:18:12 +0300 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> Message-ID: <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> Absolutely! It should fix the problem. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2017 13:14, Sasmita Panda wrote: > [root at localhost opensips]# opensips -V > > version: opensips 2.2.2 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 > > > > Do you want me to test after applying the patch ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu > wrote: > > Sasmita, could you please post the output of "opensips -V"? A > similar bug on 2.2 has been discussed and fixed exactly one month > ago [1] > > [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 > > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 05.07.2017 08:27, Sasmita Panda wrote: >> Actually I am not saving data in memory . It in DB only mode . So >> I am seeing in database . I get 2 contacts of same Username, From >> same IP and Different Port . >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Sasmita, >> >> Have you checked via "opensipsctl ul show" to see that you >> actually have 2 registrations under the same AOR ?? >> >> Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> >> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >>> Hi All . >>> >>> >>> While saving contact , I have written >>> save("location","c1fp1") . It means when a new register >>> comes that will get added and the old contact will get >>> de-reg in its expire time . >>> >>> But , if 1 contact is existing , and another 2 >>> register comes at the same time with different contact(same >>> Ip different port ) and same Username then what will happen . >>> >>> According to my observation , opensips is saving both >>> the new contacts . Is that write . While I am forcing a >>> single contact this should not happen . >>> >>> Please assist me if I am doing anything wrong . >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jul 5 07:06:04 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 5 Jul 2017 16:36:04 +0530 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> Message-ID: Even If I will apply the changes . I dont think so my issue will get fixed . My scenario : I have webclient in which webRTC is integrated . when my webclient starts , webRTC device get register in Opensips . When I am opensips multi tab another device getting register . I mean , if there is 2 tab then there will be 2 contact get registered . Even if I had written save("location", "c1fp1") , what will happen if the Refresh for 1st Register request will come . Will that discard that request or refresh the 1st contact ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Jul 5, 2017 at 3:48 PM, Liviu Chircu wrote: > Absolutely! It should fix the problem. > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 05.07.2017 13:14, Sasmita Panda wrote: > > [root at localhost opensips]# opensips -V > > version: opensips 2.2.2 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 > > > > Do you want me to test after applying the patch ? > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu wrote: > >> Sasmita, could you please post the output of "opensips -V"? A similar bug >> on 2.2 has been discussed and fixed exactly one month ago [1] >> >> [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 05.07.2017 08:27, Sasmita Panda wrote: >> >> Actually I am not saving data in memory . It in DB only mode . So I am >> seeing in database . I get 2 contacts of same Username, From same IP and >> Different Port . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu > > wrote: >> >>> Hi Sasmita, >>> >>> Have you checked via "opensipsctl ul show" to see that you actually have >>> 2 registrations under the same AOR ?? >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >>> >>> Hi All . >>> >>> >>> While saving contact , I have written save("location","c1fp1") >>> . It means when a new register comes that will get added and the old >>> contact will get de-reg in its expire time . >>> >>> But , if 1 contact is existing , and another 2 register comes >>> at the same time with different contact(same Ip different port ) and same >>> Username then what will happen . >>> >>> According to my observation , opensips is saving both the new >>> contacts . Is that write . While I am forcing a single contact this should >>> not happen . >>> >>> Please assist me if I am doing anything wrong . >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 5 07:17:28 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 5 Jul 2017 14:17:28 +0300 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> Message-ID: <0a502916-e0fb-3b58-99cb-7ee3022146fc@opensips.org> "Refresh the 1st contact" Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2017 14:06, Sasmita Panda wrote: > Even If I will apply the changes . I dont think so my issue will get > fixed . > > > My scenario : > I have webclient in which webRTC is integrated . when my webclient > starts , webRTC device get register in Opensips . > > When I am opensips multi tab another device getting register . I mean > , if there is 2 tab then there will be 2 contact get registered . > > Even if I had written save("location", "c1fp1") , what will happen if > the Refresh for 1st Register request will come . Will that discard > that request or refresh the 1st contact ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Wed, Jul 5, 2017 at 3:48 PM, Liviu Chircu > wrote: > > Absolutely! It should fix the problem. > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 05.07.2017 13:14, Sasmita Panda wrote: >> [root at localhost opensips]# opensips -V >> >> version: opensips 2.2.2 (x86_64/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN >> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 >> >> >> >> Do you want me to test after applying the patch ? >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu > > wrote: >> >> Sasmita, could you please post the output of "opensips -V"? A >> similar bug on 2.2 has been discussed and fixed exactly one >> month ago [1] >> >> [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 >> >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> >> On 05.07.2017 08:27, Sasmita Panda wrote: >>> Actually I am not saving data in memory . It in DB only mode >>> . So I am seeing in database . I get 2 contacts of same >>> Username, From same IP and Different Port . >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Sasmita, >>> >>> Have you checked via "opensipsctl ul show" to see that >>> you actually have 2 registrations under the same AOR ?? >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> >>> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >>>> Hi All . >>>> >>>> >>>> While saving contact , I have written >>>> save("location","c1fp1") . It means when a new register >>>> comes that will get added and the old contact will get >>>> de-reg in its expire time . >>>> >>>> But , if 1 contact is existing , and another 2 >>>> register comes at the same time with different >>>> contact(same Ip different port ) and same Username then >>>> what will happen . >>>> >>>> According to my observation , opensips is saving both >>>> the new contacts . Is that write . While I am forcing a >>>> single contact this should not happen . >>>> >>>> Please assist me if I am doing anything wrong . >>>> >>>> */Thanks & Regards/* >>>> /Sasmita Panda/ >>>> /Network Testing and Software Engineer/ >>>> /3CLogic , ph:07827611765/ >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jul 5 07:39:30 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 5 Jul 2017 17:09:30 +0530 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: <0a502916-e0fb-3b58-99cb-7ee3022146fc@opensips.org> References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> <0a502916-e0fb-3b58-99cb-7ee3022146fc@opensips.org> Message-ID: Thanks . My problem was solved . I have taken the latest branch opensips-2.2.4 . I get this issue fixed . Just for confirmation , Is this branch has all latest updates without any open issue ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Jul 5, 2017 at 4:47 PM, Liviu Chircu wrote: > "Refresh the 1st contact" > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 05.07.2017 14:06, Sasmita Panda wrote: > > Even If I will apply the changes . I dont think so my issue will get fixed > . > > > My scenario : > I have webclient in which webRTC is integrated . when my webclient starts > , webRTC device get register in Opensips . > > When I am opensips multi tab another device getting register . I mean , if > there is 2 tab then there will be 2 contact get registered . > > Even if I had written save("location", "c1fp1") , what will happen if the > Refresh for 1st Register request will come . Will that discard that request > or refresh the 1st contact ? > > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Wed, Jul 5, 2017 at 3:48 PM, Liviu Chircu wrote: > >> Absolutely! It should fix the problem. >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 05.07.2017 13:14, Sasmita Panda wrote: >> >> [root at localhost opensips]# opensips -V >> >> version: opensips 2.2.2 (x86_64/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 >> >> >> >> Do you want me to test after applying the patch ? >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu wrote: >> >>> Sasmita, could you please post the output of "opensips -V"? A similar >>> bug on 2.2 has been discussed and fixed exactly one month ago [1] >>> >>> [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 >>> >>> Liviu Chircu >>> OpenSIPS Developerhttp://www.opensips-solutions.com >>> >>> On 05.07.2017 08:27, Sasmita Panda wrote: >>> >>> Actually I am not saving data in memory . It in DB only mode . So I am >>> seeing in database . I get 2 contacts of same Username, From same IP and >>> Different Port . >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu < >>> bogdan at opensips.org> wrote: >>> >>>> Hi Sasmita, >>>> >>>> Have you checked via "opensipsctl ul show" to see that you actually >>>> have 2 registrations under the same AOR ?? >>>> >>>> Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >>>> >>>> Hi All . >>>> >>>> >>>> While saving contact , I have written >>>> save("location","c1fp1") . It means when a new register comes that will get >>>> added and the old contact will get de-reg in its expire time . >>>> >>>> But , if 1 contact is existing , and another 2 register comes >>>> at the same time with different contact(same Ip different port ) and same >>>> Username then what will happen . >>>> >>>> According to my observation , opensips is saving both the new >>>> contacts . Is that write . While I am forcing a single contact this should >>>> not happen . >>>> >>>> Please assist me if I am doing anything wrong . >>>> >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> 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 >> >> > > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 5 07:46:25 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 5 Jul 2017 14:46:25 +0300 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> <0a502916-e0fb-3b58-99cb-7ee3022146fc@opensips.org> Message-ID: All the latest fixes and improvements worth backporting. As we are very careful when backporting commits (fixes only!), they almost never cause additional open issues, which is rightfully the case with 2.2, as well as 2.3. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2017 14:39, Sasmita Panda wrote: > Thanks . My problem was solved . > > I have taken the latest branch opensips-2.2.4 . I get this issue fixed . > > Just for confirmation , Is this branch has all latest updates without > any open issue ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Wed, Jul 5, 2017 at 4:47 PM, Liviu Chircu > wrote: > > "Refresh the 1st contact" > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 05.07.2017 14:06, Sasmita Panda wrote: >> Even If I will apply the changes . I dont think so my issue will >> get fixed . >> >> >> My scenario : >> I have webclient in which webRTC is integrated . when my >> webclient starts , webRTC device get register in Opensips . >> >> When I am opensips multi tab another device getting register . I >> mean , if there is 2 tab then there will be 2 contact get >> registered . >> >> Even if I had written save("location", "c1fp1") , what will >> happen if the Refresh for 1st Register request will come . Will >> that discard that request or refresh the 1st contact ? >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Wed, Jul 5, 2017 at 3:48 PM, Liviu Chircu > > wrote: >> >> Absolutely! It should fix the problem. >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> >> On 05.07.2017 13:14, Sasmita Panda wrote: >>> [root at localhost opensips]# opensips -V >>> >>> version: opensips 2.2.2 (x86_64/linux) >>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, >>> PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >>> MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 >>> >>> >>> >>> Do you want me to test after applying the patch ? >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu >>> > wrote: >>> >>> Sasmita, could you please post the output of "opensips >>> -V"? A similar bug on 2.2 has been discussed and fixed >>> exactly one month ago [1] >>> >>> [1]: >>> https://github.com/OpenSIPS/opensips/commit/af07e238 >>> >>> >>> Liviu Chircu >>> OpenSIPS Developer >>> http://www.opensips-solutions.com >>> >>> >>> On 05.07.2017 08:27, Sasmita Panda wrote: >>>> Actually I am not saving data in memory . It in DB only >>>> mode . So I am seeing in database . I get 2 contacts of >>>> same Username, From same IP and Different Port . >>>> >>>> */Thanks & Regards/* >>>> /Sasmita Panda/ >>>> /Network Testing and Software Engineer/ >>>> /3CLogic , ph:07827611765/ >>>> >>>> On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu >>>> > wrote: >>>> >>>> Hi Sasmita, >>>> >>>> Have you checked via "opensipsctl ul show" to see >>>> that you actually have 2 registrations under the >>>> same AOR ?? >>>> >>>> Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> >>>> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >>>>> Hi All . >>>>> >>>>> >>>>> While saving contact , I have written >>>>> save("location","c1fp1") . It means when a new >>>>> register comes that will get added and the old >>>>> contact will get de-reg in its expire time . >>>>> >>>>> But , if 1 contact is existing , and another 2 >>>>> register comes at the same time with different >>>>> contact(same Ip different port ) and same Username >>>>> then what will happen . >>>>> >>>>> According to my observation , opensips is saving >>>>> both the new contacts . Is that write . While I am >>>>> forcing a single contact this should not happen . >>>>> >>>>> Please assist me if I am doing anything wrong . >>>>> >>>>> */Thanks & Regards/* >>>>> /Sasmita Panda/ >>>>> /Network Testing and Software Engineer/ >>>>> /3CLogic , ph:07827611765/ >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jul 5 07:53:38 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 5 Jul 2017 17:23:38 +0530 Subject: [OpenSIPS-Users] Contact saving in opensips-2.2 In-Reply-To: References: <7648194e-bf1e-611a-652b-2abb5c31a94f@opensips.org> <31a0d20b-f363-9e56-e381-b3c792dba5a6@opensips.org> <0a502916-e0fb-3b58-99cb-7ee3022146fc@opensips.org> Message-ID: Thank you so much . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Jul 5, 2017 at 5:16 PM, Liviu Chircu wrote: > All the latest fixes and improvements worth backporting. As we are very > careful when backporting commits (fixes only!), they almost never cause > additional open issues, which is rightfully the case with 2.2, as well as > 2.3. > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 05.07.2017 14:39, Sasmita Panda wrote: > > Thanks . My problem was solved . > > I have taken the latest branch opensips-2.2.4 . I get this issue fixed . > > Just for confirmation , Is this branch has all latest updates without any > open issue ? > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Wed, Jul 5, 2017 at 4:47 PM, Liviu Chircu wrote: > >> "Refresh the 1st contact" >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 05.07.2017 14:06, Sasmita Panda wrote: >> >> Even If I will apply the changes . I dont think so my issue will get >> fixed . >> >> >> My scenario : >> I have webclient in which webRTC is integrated . when my webclient starts >> , webRTC device get register in Opensips . >> >> When I am opensips multi tab another device getting register . I mean , >> if there is 2 tab then there will be 2 contact get registered . >> >> Even if I had written save("location", "c1fp1") , what will happen if the >> Refresh for 1st Register request will come . Will that discard that request >> or refresh the 1st contact ? >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> On Wed, Jul 5, 2017 at 3:48 PM, Liviu Chircu wrote: >> >>> Absolutely! It should fix the problem. >>> >>> Liviu Chircu >>> OpenSIPS Developerhttp://www.opensips-solutions.com >>> >>> On 05.07.2017 13:14, Sasmita Panda wrote: >>> >>> [root at localhost opensips]# opensips -V >>> >>> version: opensips 2.2.2 (x86_64/linux) >>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> main.c compiled on 12:17:33 May 29 2017 with gcc 4.8.5 >>> >>> >>> >>> Do you want me to test after applying the patch ? >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> On Wed, Jul 5, 2017 at 3:13 PM, Liviu Chircu wrote: >>> >>>> Sasmita, could you please post the output of "opensips -V"? A similar >>>> bug on 2.2 has been discussed and fixed exactly one month ago [1] >>>> >>>> [1]: https://github.com/OpenSIPS/opensips/commit/af07e238 >>>> >>>> Liviu Chircu >>>> OpenSIPS Developerhttp://www.opensips-solutions.com >>>> >>>> On 05.07.2017 08:27, Sasmita Panda wrote: >>>> >>>> Actually I am not saving data in memory . It in DB only mode . So I am >>>> seeing in database . I get 2 contacts of same Username, From same IP and >>>> Different Port . >>>> >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> >>>> On Tue, Jul 4, 2017 at 10:17 PM, Bogdan-Andrei Iancu < >>>> bogdan at opensips.org> wrote: >>>> >>>>> Hi Sasmita, >>>>> >>>>> Have you checked via "opensipsctl ul show" to see that you actually >>>>> have 2 registrations under the same AOR ?? >>>>> >>>>> Best regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> OpenSIPS Founder and Developer >>>>> http://www.opensips-solutions.com >>>>> >>>>> OpenSIPS Bootcamp 2017, Houston, US >>>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>>> >>>>> On 07/04/2017 04:03 PM, Sasmita Panda wrote: >>>>> >>>>> Hi All . >>>>> >>>>> >>>>> While saving contact , I have written >>>>> save("location","c1fp1") . It means when a new register comes that will get >>>>> added and the old contact will get de-reg in its expire time . >>>>> >>>>> But , if 1 contact is existing , and another 2 register comes >>>>> at the same time with different contact(same Ip different port ) and same >>>>> Username then what will happen . >>>>> >>>>> According to my observation , opensips is saving both the new >>>>> contacts . Is that write . While I am forcing a single contact this should >>>>> not happen . >>>>> >>>>> Please assist me if I am doing anything wrong . >>>>> >>>>> *Thanks & Regards* >>>>> *Sasmita Panda* >>>>> *Network Testing and Software Engineer* >>>>> *3CLogic , ph:07827611765* >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> 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 >>> >>> >> >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Wed Jul 5 08:32:03 2017 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Wed, 5 Jul 2017 15:32:03 +0300 Subject: [OpenSIPS-Users] Problem: Registration Proxy with WebRTC In-Reply-To: References: Message-ID: Here is answer from JsSIP but I do not know how to setup Opensips: =================== answer =============== The error log does tell you what is happening: ``` JsSIP | SANITY CHECK | Via host in the response does not match UA Via host value. Dropping the respons ``` Have a look at the chapter 18.1.2 from RFC 3261: ``` 18.1.2 Receiving Responses When a response is received, the client transport examines the top Via header field value. If the value of the "sent-by" parameter in that header field value does not correspond to a value that the client transport is configured to insert into requests, the response MUST be silently discarded. ``` The "sent-by" parameter in the response has been altered from the one sent out from JsSIP. Your server is manipulating such a value and that makes the response to be discardes as per RFC 3261. 2017-07-02 22:20 GMT+03:00 Dragomir Haralambiev : > Hello, > > I try to setup Registration proxy (opensips 2.3) like this article: > https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ > > 1. When use Zoiper all is OK > Zoiper <----> Opensips REGISTRATION Proxy <-----> SBC > > 2. I have problem when use WebRTC. > (*Via sent-by in the response does not match UA Via host value. Dropping > the response*): > > SIP.JS <----->Opensips REGISTRATION Proxy <-----> SBC > > Here is WebSocket text messages: > > REGISTER sip: SIP/2.0 > Via: SIP/2.0/WSS 192.0.2.148;branch=z9hG4bK8881457 > Max-Forwards: 70 > To: "Tester" > > From: "Tester" >;tag=u6aro6a8mj > Call-ID: k5uhq12e1bb93rg9igvpvv > CSeq: 83 REGISTER > Contact: ;reg-id=1;+sip. > instance="";expires=60 > Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER > Supported: path, gruu, outbound > User-Agent: SIP.js/0.7.8 > Content-Length: 0 > > > > sip-0.7.8.js:2900 Sun Jul 02 2017 18:41:19 GMT+0300 (FLE Daylight Time) | > sip.transport | received WebSocket text message: > > SIP/2.0 401 Unauthorized > Via: SIP/2.0/WSS 192.0.2.148:5060;rport=53162;received=;branch= > z9hG4bK8881457 > From: "Tester" >;tag=u6aro6a8mj > To: "Tester" > > Call-ID: k5uhq12e1bb93rg9igvpvv > CSeq: 83 REGISTER > Contact: :53162;transport=wss>;reg-id=1; > +sip.instance="";expires=60 > WWW-Authenticate: Digest realm="sbc.com", nonce="ee228f001f1459108000000 > c2916c1ef at sbc.com" > Content-Length: 0 > > > > sip-0.7.8.js:2900 Sun Jul 02 2017 18:41:19 GMT+0300 (FLE Daylight Time) | > sip.sanitycheck | *Via sent-by in the response does not match UA Via host > value. Dropping the response* > > > Best regards, > Drgagomir > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liran.aknin at vonage.com Wed Jul 5 09:23:40 2017 From: liran.aknin at vonage.com (Aknin, Liran) Date: Wed, 5 Jul 2017 16:23:40 +0300 Subject: [OpenSIPS-Users] Using usrloc module with Redis DB Message-ID: Hi Bogdan and all, We would like to cluster few opensips servers, in terms of sharing usrloc data between them. We already use Redis on these servers and we want to take advantage of that, as well as enjoining faster access comparing to SQL. Is it possible to use usrloc module with Redis? If so, can you please provide us with an example of defining usrloc's db_url parameter to do so? Thanks and regards, Liran -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Jul 5 10:32:36 2017 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 5 Jul 2017 14:32:36 +0000 Subject: [OpenSIPS-Users] SIP URI User Parameters In-Reply-To: <131bc2e6-8b7b-071e-8fff-6a97442effcd@opensips.org> References: <208A1714-5990-44FD-AA64-073AC633E249@genesys.com> <131bc2e6-8b7b-071e-8fff-6a97442effcd@opensips.org> Message-ID: Bogdan, Thanks for your work to find the issue. I do agree that the usage of the “user=phone” parameter is not well defined and a bit ambiguous. However, I think your action is correct as it should be the responsibility of the end gateway to do any necessary SIP -> Tel conversion, not the proxy. And especially not a partial conversion. :) Just to clarify, where was the change you made submitted? I know 1.11 is no longer supported, but we are still using it and are not ready to upgrade yet due to the many script changes necessary to use 2.X. If this change cannot be added to 1.11, do you have any suggestions for a workaround? I haven’t found anything yet, but I’ve yet to try using revert_uri in the failure route to remove the user params before any other processing. Do you think this will work? Ben Newlin Lead Voice Network Engineer, PureCloud [cid:image001.png at 01D2F57A.03C1BF50] O +1 317.957.1009 ben.newlin at genesys.com [cid:image001.png at 01D2F57A.03C1BF50] [cid:image002.png at 01D2F57A.03C1BF50] [cid:image003.png at 01D2F57A.03C1BF50][cid:image004.png at 01D2F57A.03C1BF50][cid:image005.png at 01D2F57A.03C1BF50][cid:image006.png at 01D2F57A.03C1BF50][cid:image007.png at 01D2F57A.03C1BF50][cid:image008.png at 01D2F57A.03C1BF50] From: Bogdan-Andrei Iancu Date: Monday, July 3, 2017 at 11:46 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] SIP URI User Parameters Hi Ben, Thank you for your digging and reporting. Following your leads I found some old strange behavior of the parse_uri() function - the function responsible for parsing the URIs in OpenSISP. For some ancient and unknown reasons, a SIP URI with user=phone was automatically converted to a TEL URI. Such conversion, automatically done, is dangerous - there is nothing in the RFC3261 stating something like this. Even more, the conversion is not complete - besides moving the username parameters to URI parameters, the domain is not stripped and the TEL not added. Basically, the existing code was converting: sip:username;bla=foo at host.com;param1=1;param2=2;user=phone to sip:username at host.com;bla=foo I tried to dig around the subject, but not more - there is no reference or recommendation for such a behavior. If you have the time, see these links: * SIP implementer -> https://lists.cs.columbia.edu/pipermail/sip-implementors/2013-February/028837.html * SIP Core -> https://www.ietf.org/mail-archive/web/sipcore/current/msg01783.html * voip info -> https://www.voip-info.org/wiki/view/SIP+URI (Telephone numbers section) On voip-info there is a recommendation on how to compare a SIP uri with a TEL uri (in terms of username and parameters parts), but nothing of a "must" conversion. So, I disabled the guilty code in OpenSIPS, and it should work as expected now. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/30/2017 10:14 PM, Ben Newlin wrote: Bogdan, I have been able to reproduce this locally now. The piece that was missing is that the Request URI must already have URI parameters on it. If it has both URI and user parameters, the call to route_to_carrier (and possibly do_routing) replaces all of the URI parameters with the user parameter(s). Original $ru: sip:+15555551212;npdi=yes at gw2.com;transport=udp;user=phone After call to route_to_carrier: sip:+15555551212 at gw2.com;npdi=yes After further testing, it appears this behavior is not restricted to the Dynamic Routing module after all. Simply modifying $ru while both user and URI parameters are present causes the issue. Original $ru: sip:+15555551212;npdi=yes at gw2.com;transport=udp;user=phone Perform this action: $rU = $rU + “;rn=+15555550000”; Resultant $ru: sip:+15555551212;rn=+15555550000 at gw2.com;npdi=yes Ben Newlin Lead Voice Network Engineer, PureCloud [cid:image009.png at 01D2F57A.03C1BF50] O +1 317.957.1009 ben.newlin at genesys.com [cid:image010.png at 01D2F57A.03C1BF50] [cid:image011.png at 01D2F57A.03C1BF50] [cid:image012.png at 01D2F57A.03C1BF50][cid:image013.png at 01D2F57A.03C1BF50][cid:image014.png at 01D2F57A.03C1BF50][cid:image015.png at 01D2F57A.03C1BF50][cid:image016.png at 01D2F57A.03C1BF50][cid:image017.png at 01D2F57A.03C1BF50] From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Friday, June 30, 2017 at 10:47 AM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] SIP URI User Parameters Bogdan, Sorry for the delayed response, I am having some trouble reproducing this in a local test environment. Currently it is only occurring in our live environment. I do have some clarifications and answers to your questions: · The npdi parameter is not present in $ru in the failure route when the response is 500. It is present when the response is 503 or 408. I haven’t tested any other responses. This is not terribly important to my issue, simply an observation. · We are sometimes using do_routing to populate a list of carriers, but other times we get the list from our own DB query. We use route_to_carrier to send the call to each carrier in sequence. This is because we don’t always use do_routing, but also because we wish to skip to the next carrier, not just the next gateway, on certain response codes and the normal do_routing mechanism doesn’t allow that. · The issue actually does not happen when use_next_gw is called. I was wrong about that. You were right that seems to be a straight URI copy. The issue occurs when we skip use_next_gw or there are no gateways left and we call route_to_carrier for the next carrier with the parameter present in $ru. · I printed out the dr_ruri avp after the call to route_to_carrier and it shows the npdi parameter moved to the end, not after the user: “sip:+15555551212 at gw2.com;npdi=yes” Also, I should have mentioned that we are running 1.11.11. I’m still working to try to reproduce locally. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, June 29, 2017 at 4:38 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] SIP URI User Parameters Hello Ben I understand you add the npdi useraname parameter after performing the initial do_routing() - if you do it in request or branch route is not relevant (for RURI changes) as RURI is anyhow a per-branch value. In failure route, when resuming, you will get the RURI of the winning branch ( the one which was selected to be sent back to caller), so you see the npdi param. So far so good. And now you do use_next_gw() in failure route and you get "sip:+15555551212 at gw2.com;npdi" directly, without any another npdi addition ? I'm asking, as use_next_gw() does a full RURI replacement (it doesn;t care what is the existing RURI). Could you also do an xlog("DR ruris are <$(avp(___dr_ruri__)[*])>\n"); right after do_routing() ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/28/2017 11:41 PM, Ben Newlin wrote: Hi, We have run into an issue with OpenSIPs’ handling of user parameters in SIP URIs with Dynamic Routing module. When a parameter is added to a SIP URI user part, any subsequent modification of the URI by DR module results in the parameter being moved to be a URI parameter. For example, starting with $ru of “sip:+15555551212 at gw1.com”, if we modify it this way: $rU = $rU + “;npdi”; then we get a new $ru of “sip:+15555551212;npdi at gw1.com”. We send this call out and if it returns an error we want to use the next available gateway. The Request URI in the failure route is still “sip:+15555551212;npdi at gw1.com”. Note: this is the case even when the “;npdi” parameter was added in a branch route, which I didn’t expect. I thought changes made in a branch route were isolated to that branch. Now from the failure route when we call use_next_gw the npdi parameter is moved and the URI is now “sip:+15555551212 at gw2.com;npdi”. This is not correct. Is there some other way to properly manipulate SIP URI user parameters or is this a bug? Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 1242 bytes Desc: image001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 7162 bytes Desc: image002.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.png Type: image/png Size: 2054 bytes Desc: image003.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.png Type: image/png Size: 2041 bytes Desc: image004.png URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: image017.png Type: image/png Size: 2072 bytes Desc: image017.png URL: From bogdan at opensips.org Wed Jul 5 10:38:40 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 17:38:40 +0300 Subject: [OpenSIPS-Users] Passive call recording using rtpproxy In-Reply-To: References: Message-ID: <841b5ac5-a7b5-039a-3276-583030c84368@opensips.org> Hi Alexis, To record the RTP on the your platform, you need first to get the RTP to flow through your servers, right ? So you have to do RTP pinning - to be transparent for the users (if recorded or not), you will have to do it for all calls I guess. And here you can use RTPproxy to pin the media. Now, once you have the RTP on your servers (via rtpproxy), you have 2 options: 1) get the recording directly from RTPProxy 2) get the RTP via port mirroring Let me know if ok so far. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 11:24 AM, Lameire Alexis wrote: > > Hello, > > I would like to acheave call recording on a passive way. My setup will > use a port mirroring on the switch on a dedicated port. > > Despite the presentation on the last summit, I can't properly > understand how to properly acheave it. As I get rtpproxy will rewrite > the SDP content to fix the media ip to the instance of rtpproxy, but > this biavior is not suitable for a passive call recording. > > As I see on this[1] documentation I don't find a way to inhibate this > biavior and just rely on the choiced RTP/RTSP port. > > In additions I will be faced by the issues related to port mirroring, > the destination ip on sig and media packet will not match the local > address and rtpproxy and opensips will not be able to bind the > datagram socket. > > So, I think I have lost something, so if you could glow my mind you > will be gracefully thanked. > > In addition of my questions, I would like to strongly thanks the > presenter for is call recording experience, mainly due to the keys > provided on the post task for files conversion. > > [1] http://www.opensips.org/html/docs/modules/1.7.x/rtpproxy.html > > -- > Signature > Alexis Lameire > Ingénieur infrastructure > > +33 9 70 71 60 01 > alameire at openip.fr 37/39, rue de Neuilly, 92110 Clichy > http://openip.fr / http://my.openip.fr e-mail > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jul 5 10:46:28 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 17:46:28 +0300 Subject: [OpenSIPS-Users] SIP URI User Parameters In-Reply-To: References: <208A1714-5990-44FD-AA64-073AC633E249@genesys.com> <131bc2e6-8b7b-071e-8fff-6a97442effcd@opensips.org> Message-ID: <43fee1c8-eb73-ca10-4e10-8ec408f4fd52@opensips.org> Hi Ben, The fix is present on trunk (2.4), 2.3 and 2.2 (the currently maintained versions). Indeed, the 1.11 does not have the fix. You can easily apply the fix onyour 1.11 code via this patch - https://github.com/OpenSIPS/opensips/commit/91c14ce679f80c8b4888769004c08039da2fc805.patch . It should be 100% compatible. Otherwise, you can try to get rid of user=phone in the URI by doing changes over the full RURI (to avoid its parsing)- like a subst over the full RURI. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 05:32 PM, Ben Newlin wrote: > > Bogdan, > > Thanks for your work to find the issue. I do agree that the usage of > the “user=phone” parameter is not well defined and a bit ambiguous. > However, I think your action is correct as it should be the > responsibility of the end gateway to do any necessary SIP -> Tel > conversion, not the proxy. And especially not a partial conversion. :) > > Just to clarify, where was the change you made submitted? I know 1.11 > is no longer supported, but we are still using it and are not ready to > upgrade yet due to the many script changes necessary to use 2.X. If > this change cannot be added to 1.11, do you have any suggestions for a > workaround? I haven’t found anything yet, but I’ve yet to try using > revert_uri in the failure route to remove the user params before any > other processing. Do you think this will work? > > Ben Newlin > > Lead Voice Network Engineer, PureCloud > > ** > > O+1 317.957.1009 > > _ben.newlin at genesys.com _ > > > > > > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, July 3, 2017 at 11:46 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] SIP URI User Parameters > > Hi Ben, > > Thank you for your digging and reporting. Following your leads I found > some old strange behavior of the parse_uri() function - the function > responsible for parsing the URIs in OpenSISP. > > > For some ancient and unknown reasons, a SIP URI with user=phone was automatically converted to a TEL URI. Such conversion, automatically done, is dangerous - there is nothing in the RFC3261 stating something like this. Even more, the conversion is not complete - besides moving the username parameters to URI parameters, the domain is not stripped and the TEL not added. > Basically, the existing code was converting: > sip:username;bla=foo at host.com;param1=1;param2=2;user=phone > to > sip:username at host.com;bla=foo > I tried to dig around the subject, but not more - there is no reference or recommendation for such a behavior. If you have the time, see these links: > * SIP implementer ->https://lists.cs.columbia.edu/pipermail/sip-implementors/2013-February/028837.html > * SIP Core ->https://www.ietf.org/mail-archive/web/sipcore/current/msg01783.html > * voip info ->https://www.voip-info.org/wiki/view/SIP+URI (Telephone numbers section) > On voip-info there is a recommendation on how to compare a SIP uri with a TEL uri (in terms of username and parameters parts), but nothing of a "must" conversion. > So, I disabled the guilty code in OpenSIPS, and it should work as expected now. > > > Best regards, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 2071 bytes Desc: not available URL: From bogdan at opensips.org Wed Jul 5 10:53:14 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 17:53:14 +0300 Subject: [OpenSIPS-Users] Using usrloc module with Redis DB In-Reply-To: References: Message-ID: <797906ba-f4f1-2196-196d-7d8bc1ce4e15@opensips.org> Hi Liran, Right now, due the complexity of the usrloc data, there is no straight way to use an SQL backend (there is a need for list support with atomic ops). We are working for such support for 2.4 release. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 04:23 PM, Aknin, Liran via Users wrote: > Hi Bogdan and all, > > We would like to cluster few opensips servers, in terms of sharing > usrloc data between them. We already use Redis on these servers and we > want to take advantage of that, as well as enjoining faster access > comparing to SQL. > > Is it possible to use usrloc module with Redis? > If so, can you please provide us with an example of defining usrloc's > db_url parameter to do so? > > Thanks and regards, > Liran > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jul 5 11:24:26 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 18:24:26 +0300 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 7.2.3 is released (for OpenSIPS 2.3) Message-ID: <6adaf2af-a46b-4335-ffd5-5516479802e7@opensips.org> Hi all, Proudly I announce the release of the OpenSIPS Control Panel 7.2.3 (according to the new versioning schema) matching the OpenSIPS 2.3 version. Besides aligning the Control Panel tools to the changes from OpenSIPS 2.3 (like for Clusterer, Dialog, Dispatcher and Load Balancer), the 7.2.3 brings also some news: * Realtime Validation and Tool Tips for the input forms - this is work in progress, not all tools benefit yet, only Clusterer, Dispatcher, Domain and Load Balancer) * update the postgres support - all queries performed via MDB2 were PGSQL validated * improve user experience in filling in input forms (do not present ANY as input selection) * drop FIFO and XMLRPC as options for MI connector and support only JSON (mi_json) * added GET-based authentication against HOMER interface, for a more flexible and simple integration * CSS merging and php code sharing - internal work to simplify the development work when creating new tools Of course, this new version contains a lot of fixes and cleanup work, layout beautification and optimizations. The full ChangeLog is available here: https://github.com/OpenSIPS/opensips-cp/blob/7.2.3/ChangeLog To download OpenSIPS Control Panel, please refer to the project web site: http://controlpanel.opensips.org/ or to the github project https://github.com/OpenSIPS/opensips-cp/tree/7.2.3 Enjoy, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html From me at nevian.org Wed Jul 5 11:57:27 2017 From: me at nevian.org (Serge S. Yuriev) Date: Wed, 5 Jul 2017 18:57:27 +0300 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 7.2.3 is released (for OpenSIPS 2.3) In-Reply-To: <6adaf2af-a46b-4335-ffd5-5516479802e7@opensips.org> References: <6adaf2af-a46b-4335-ffd5-5516479802e7@opensips.org> Message-ID: <10da8322-70be-cd25-287d-d0094252324c@nevian.org> Hi, Will it work with OpenSIPS trunk version? On 05/07/17 18:24, Bogdan-Andrei Iancu wrote: > Hi all, > > Proudly I announce the release of the OpenSIPS Control Panel 7.2.3 > (according to the new versioning schema) matching the OpenSIPS 2.3 version. -- Serge S. Yuriev Lead VoIP engineer From samusenko at msm.ru Wed Jul 5 12:03:34 2017 From: samusenko at msm.ru (Andrey) Date: Wed, 05 Jul 2017 19:03:34 +0300 Subject: [OpenSIPS-Users] Cluesterer and ul_add permanent Message-ID: <4647681.e5x6FOWdfS@samusenko> OpenSIPS 2.2.4 Subject does not work. Slave: WARNING:usrloc:receive_binary_packet: received bin packet from unknown source: 192.168.1.2:4414 and MI process shutdown. > select url from clusterer: bin:192.168.1.2:8026 bin:192.168.1.3:8026 From bogdan at opensips.org Wed Jul 5 12:42:09 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 5 Jul 2017 19:42:09 +0300 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 7.2.3 is released (for OpenSIPS 2.3) In-Reply-To: <10da8322-70be-cd25-287d-d0094252324c@nevian.org> References: <6adaf2af-a46b-4335-ffd5-5516479802e7@opensips.org> <10da8322-70be-cd25-287d-d0094252324c@nevian.org> Message-ID: <1b04ca58-4e70-4705-1fef-0e168aee0348@opensips.org> Hi Serge, Yes, for now it works with OpenSIPS trunk too. Still, in time, due the development progress, this compatibility with the may get lost. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 06:57 PM, Serge S. Yuriev wrote: > Hi, > > Will it work with OpenSIPS trunk version? > > On 05/07/17 18:24, Bogdan-Andrei Iancu wrote: >> Hi all, >> >> Proudly I announce the release of the OpenSIPS Control Panel 7.2.3 >> (according to the new versioning schema) matching the OpenSIPS 2.3 >> version. > From razvan at opensips.org Wed Jul 5 12:59:00 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 5 Jul 2017 19:59:00 +0300 Subject: [OpenSIPS-Users] OpenSIPS 2.2.5 and 2.3.1 releases are out! Message-ID: <66607880-5238-f920-d59a-051e178d7bf0@opensips.org> Hello! I am happy to announce two new releases of OpenSIPS - versions 2.2.5 and 2.3.1. These new releases bring a series of improvements and important fixes to the core TCP networking layer, as well as to OpenSIPS modules (acc, dialog, clusterer, pua,exec, etc). Note that OpenSIPS 2.3.1 is the first minor release of the latest 2.3 version released in April 2017, thus it includes a lot of feedback from the "early users" of this branch. You can find the latest ChangeLog here: * OpenSIPS 2.2.5: http://opensips.org/pub/opensips/2.2.5/ChangeLog * OpenSIPS 2.3.1: http://opensips.org/pub/opensips/2.3.1/ChangeLog You can get the latest versions from here: * OpenSIPS 2.2.5: http://opensips.org/pub/opensips/2.2.5/opensips-2.2.5.tar.gz * OpenSIPS 2.3.1: http://opensips.org/pub/opensips/2.3.1/opensips-2.3.1.tar.gz Thank you everyone for the hard work invested in the OpenSIPS project! Cheers, -- Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com From daniel.zanutti at gmail.com Wed Jul 5 22:43:33 2017 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 5 Jul 2017 23:43:33 -0300 Subject: [OpenSIPS-Users] RTPENGINE (sipwise) working with opensips? Message-ID: Question: Is RTPENGINE (sipwise) working with opensips? I saw only Kamailio on rtpengine page at github, but there is a module for opensips 2.1 ( http://www.opensips.org/html/docs/modules/2.1.x/rtpengine). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Wed Jul 5 23:34:15 2017 From: johan at democon.be (Johan De Clercq) Date: Thu, 6 Jul 2017 05:34:15 +0200 Subject: [OpenSIPS-Users] RTPENGINE (sipwise) working with opensips? In-Reply-To: References: Message-ID: Yes it is. On 06 Jul 2017 4:44 AM, "Daniel Zanutti" wrote: > Question: > > Is RTPENGINE (sipwise) working with opensips? I saw only Kamailio on > rtpengine page at github, but there is a module for opensips 2.1 ( > http://www.opensips.org/html/docs/modules/2.1.x/rtpengine). > > Thanks > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Jul 5 13:16:13 2017 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 5 Jul 2017 17:16:13 +0000 Subject: [OpenSIPS-Users] SIP URI User Parameters In-Reply-To: <43fee1c8-eb73-ca10-4e10-8ec408f4fd52@opensips.org> References: <208A1714-5990-44FD-AA64-073AC633E249@genesys.com> <131bc2e6-8b7b-071e-8fff-6a97442effcd@opensips.org> <43fee1c8-eb73-ca10-4e10-8ec408f4fd52@opensips.org> Message-ID: <27C4FA7A-AD02-46A2-B633-E0FDD8F425F5@genesys.com> Bogdan, Thanks for your suggestion to modify the whole RURI. While this will work for our modifications, we have no control over how modules like Dynamic Routing access the RURI and that module’s functions are also causing the error to occur. So this will not work. However, I have found that using revert_uri within failure_route will remove the user params prior to performing routing and so prevents the issue. We can then add the user params back before sending out the next branch. This workaround is of course specific to our current use case, where we are always injecting the user params ourselves. It would not work if the user params were present in the original received RURI. We will continue with this workaround until we can upgrade to version 2. Thanks again for your help. Ben Newlin From: Bogdan-Andrei Iancu Date: Wednesday, July 5, 2017 at 10:46 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] SIP URI User Parameters Hi Ben, The fix is present on trunk (2.4), 2.3 and 2.2 (the currently maintained versions). Indeed, the 1.11 does not have the fix. You can easily apply the fix on your 1.11 code via this patch - https://github.com/OpenSIPS/opensips/commit/91c14ce679f80c8b4888769004c08039da2fc805.patch . It should be 100% compatible. Otherwise, you can try to get rid of user=phone in the URI by doing changes over the full RURI (to avoid its parsing) - like a subst over the full RURI. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 05:32 PM, Ben Newlin wrote: Bogdan, Thanks for your work to find the issue. I do agree that the usage of the “user=phone” parameter is not well defined and a bit ambiguous. However, I think your action is correct as it should be the responsibility of the end gateway to do any necessary SIP -> Tel conversion, not the proxy. And especially not a partial conversion. :) Just to clarify, where was the change you made submitted? I know 1.11 is no longer supported, but we are still using it and are not ready to upgrade yet due to the many script changes necessary to use 2.X. If this change cannot be added to 1.11, do you have any suggestions for a workaround? I haven’t found anything yet, but I’ve yet to try using revert_uri in the failure route to remove the user params before any other processing. Do you think this will work? Ben Newlin Lead Voice Network Engineer, PureCloud [cid:image001.png at 01D2F590.DDA52890] O +1 317.957.1009 ben.newlin at genesys.com [cid:image001.png at 01D2F590.DDA52890] [cid:image002.png at 01D2F590.DDA52890][cid:image003.png at 01D2F590.DDA52890][cid:image004.png at 01D2F590.DDA52890][cid:image005.png at 01D2F590.DDA52890][cid:image006.png at 01D2F590.DDA52890][cid:image007.png at 01D2F590.DDA52890] From: Bogdan-Andrei Iancu Date: Monday, July 3, 2017 at 11:46 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] SIP URI User Parameters Hi Ben, Thank you for your digging and reporting. Following your leads I found some old strange behavior of the parse_uri() function - the function responsible for parsing the URIs in OpenSISP. For some ancient and unknown reasons, a SIP URI with user=phone was automatically converted to a TEL URI. Such conversion, automatically done, is dangerous - there is nothing in the RFC3261 stating something like this. Even more, the conversion is not complete - besides moving the username parameters to URI parameters, the domain is not stripped and the TEL not added. Basically, the existing code was converting: sip:username;bla=foo at host.com;param1=1;param2=2;user=phone to sip:username at host.com;bla=foo I tried to dig around the subject, but not more - there is no reference or recommendation for such a behavior. If you have the time, see these links: * SIP implementer -> https://lists.cs.columbia.edu/pipermail/sip-implementors/2013-February/028837.html * SIP Core -> https://www.ietf.org/mail-archive/web/sipcore/current/msg01783.html * voip info -> https://www.voip-info.org/wiki/view/SIP+URI (Telephone numbers section) On voip-info there is a recommendation on how to compare a SIP uri with a TEL uri (in terms of username and parameters parts), but nothing of a "must" conversion. So, I disabled the guilty code in OpenSIPS, and it should work as expected now. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 1243 bytes Desc: image001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 2055 bytes Desc: image002.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.png Type: image/png Size: 2042 bytes Desc: image003.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.png Type: image/png Size: 2045 bytes Desc: image004.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image005.png Type: image/png Size: 1921 bytes Desc: image005.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image006.png Type: image/png Size: 2058 bytes Desc: image006.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image007.png Type: image/png Size: 2072 bytes Desc: image007.png URL: From liran.aknin at vonage.com Thu Jul 6 01:29:00 2017 From: liran.aknin at vonage.com (Aknin, Liran) Date: Thu, 6 Jul 2017 08:29:00 +0300 Subject: [OpenSIPS-Users] Using usrloc module with Redis DB In-Reply-To: <797906ba-f4f1-2196-196d-7d8bc1ce4e15@opensips.org> References: <797906ba-f4f1-2196-196d-7d8bc1ce4e15@opensips.org> Message-ID: Hi Bogdan, Thanks for your prompt reply. Just to make sure I understood you correctly, since I was actually asking about Redis - you mean that there is support for *NO-SQL* DB, right? Is usrloc over Redis going to be introduced in 2.4? thanks, Liran On Wed, Jul 5, 2017 at 5:53 PM, Bogdan-Andrei Iancu wrote: > Hi Liran, > > Right now, due the complexity of the usrloc data, there is no straight way > to use an SQL backend (there is a need for list support with atomic ops). > We are working for such support for 2.4 release. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/05/2017 04:23 PM, Aknin, Liran via Users wrote: > > Hi Bogdan and all, > > We would like to cluster few opensips servers, in terms of sharing usrloc > data between them. We already use Redis on these servers and we want to > take advantage of that, as well as enjoining faster access comparing to SQL. > > Is it possible to use usrloc module with Redis? > If so, can you please provide us with an example of defining usrloc's > db_url parameter to do so? > > Thanks and regards, > Liran > > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 6 06:35:55 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 6 Jul 2017 13:35:55 +0300 Subject: [OpenSIPS-Users] Using usrloc module with Redis DB In-Reply-To: References: <797906ba-f4f1-2196-196d-7d8bc1ce4e15@opensips.org> Message-ID: <9923591e-8cb7-1220-35a4-9289143ec7b0@opensips.org> Hi Liran, yes, typo, it should have been "there is no straight way to use an noSQL backends" But this is for location. Otherwise, OpenSIPS has a cachedb_redis module you can use from script as a key-value cache . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/06/2017 08:29 AM, Aknin, Liran wrote: > Hi Bogdan, > > Thanks for your prompt reply. > > Just to make sure I understood you correctly, since I was actually > asking about Redis - you mean that there is support for *NO-SQL* DB, > right? > Is usrloc over Redis going to be introduced in 2.4? > > thanks, > Liran > > On Wed, Jul 5, 2017 at 5:53 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Liran, > > Right now, due the complexity of the usrloc data, there is no > straight way to use an SQL backend (there is a need for list > support with atomic ops). We are working for such support for 2.4 > release. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/05/2017 04:23 PM, Aknin, Liran via Users wrote: >> Hi Bogdan and all, >> >> We would like to cluster few opensips servers, in terms of >> sharing usrloc data between them. We already use Redis on these >> servers and we want to take advantage of that, as well as >> enjoining faster access comparing to SQL. >> >> Is it possible to use usrloc module with Redis? >> If so, can you please provide us with an example of defining >> usrloc's db_url parameter to do so? >> >> Thanks and regards, >> Liran >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 6 06:43:31 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 6 Jul 2017 13:43:31 +0300 Subject: [OpenSIPS-Users] SIP URI User Parameters In-Reply-To: <27C4FA7A-AD02-46A2-B633-E0FDD8F425F5@genesys.com> References: <208A1714-5990-44FD-AA64-073AC633E249@genesys.com> <131bc2e6-8b7b-071e-8fff-6a97442effcd@opensips.org> <43fee1c8-eb73-ca10-4e10-8ec408f4fd52@opensips.org> <27C4FA7A-AD02-46A2-B633-E0FDD8F425F5@genesys.com> Message-ID: Ben, Exploring the "whole RURI modification" option - once you get rid of the user=phone param, you are safe for the rest of the processing. My idea was to strip that parameter in the very beginning of the processing. After that, any DR processing will be safe. Something like: subst_uri('/^(.+);user=phone([^@]*)$/\1\2/i'); Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 08:16 PM, Ben Newlin wrote: > > Bogdan, > > Thanks for your suggestion to modify the whole RURI. While this will > work for our modifications, we have no control over how modules like > Dynamic Routing access the RURI and that module’s functions are also > causing the error to occur. So this will not work. > > However, I have found that using revert_uri within failure_route will > remove the user params prior to performing routing and so prevents the > issue. We can then add the user params back before sending out the > next branch. This workaround is of course specific to our current use > case, where we are always injecting the user params ourselves. It > would not work if the user params were present in the original > received RURI. > > We will continue with this workaround until we can upgrade to version > 2. Thanks again for your help. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Wednesday, July 5, 2017 at 10:46 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] SIP URI User Parameters > > Hi Ben, > > The fix is present on trunk (2.4), 2.3 and 2.2 (the currently > maintained versions). Indeed, the 1.11 does not have the fix. You can > easily apply the fix on your 1.11 code via this patch - > https://github.com/OpenSIPS/opensips/commit/91c14ce679f80c8b4888769004c08039da2fc805.patch > . It should be 100% compatible. > > Otherwise, you can try to get rid of user=phone in the URI by doing > changes over the full RURI (to avoid its parsing) - like a subst over > the full RURI. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/05/2017 05:32 PM, Ben Newlin wrote: > > Bogdan, > > Thanks for your work to find the issue. I do agree that the usage > of the “user=phone” parameter is not well defined and a bit > ambiguous. However, I think your action is correct as it should be > the responsibility of the end gateway to do any necessary SIP -> > Tel conversion, not the proxy. And especially not a partial > conversion. :) > > Just to clarify, where was the change you made submitted? I know > 1.11 is no longer supported, but we are still using it and are not > ready to upgrade yet due to the many script changes necessary to > use 2.X. If this change cannot be added to 1.11, do you have any > suggestions for a workaround? I haven’t found anything yet, but > I’ve yet to try using revert_uri in the failure route to remove > the user params before any other processing. Do you think this > will work? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 6 07:56:17 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 6 Jul 2017 14:56:17 +0300 Subject: [OpenSIPS-Users] B2B not relaying 180 in prepaid scenario In-Reply-To: References: <4999fd30-2029-cb7e-0d0b-b5daa3fa364c@opensips.org> Message-ID: <9ab18f43-09e8-878d-319f-21a5845d1b0f@opensips.org> Hi Andreas, As I don't know of any mechanism to force a ringing in an ongoing call by simply using SIP signaling, an alternative is to do inject the ringing as media stream - you can use rtpproxy to play a ringing to the A-side between the re-INVITE and its 200 OK. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/04/2017 08:00 PM, Andreas Bøckmann wrote: > Hello Bogdan > > It makes sense; to not send them out that is. The A-side has sent 200 > in reply to re-INVITE and is already "in a call". > > Is it braking RFC3261 to send them to the client and is there anything > said in regards of how the client must/shall act when receiving 180 > after 200? > > It would make sense to me to let the A-party know the status of > "call-progress" during the call (actually both towards mediaserver and > later B-party). > > How would you suggest solving the case of actually playing back 180 > for the A-side after hearing the first media? > > Using the prepaid scenario example it's waiting for BYE from the > mediaserver before re-INVITE and connecting the actual B-party. > This allows me to force the mediafile to finish playing - something I > want it to (not inject media in 183 or similar during ringing). > > > //Andreas > > > > On Tue, Jul 4, 2017 at 6:43 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Andreas, > > Yes, in bridging mode (when one of the party was already connected > to a previous entity), the provisional replies are not sent > anymore (during the re-INVITE) as make no sense (for the already > connected party). > > In your case, once the A side was connected (via 200 OK) to the > media server, an incoming 180 on the re-INVITE (while bridging to > the B side) will not induce a ringing tone at all. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/04/2017 02:07 PM, Andreas Bøckmann wrote: >> Hello >> >> I am playing around with B2B and running OpenSIPS proxy and B2B >> on the same VM. >> I am triggering prepaid scenario on initial INVITEs for >> authenticated clients. >> >> https://www.opensips.org/Documentation/Tutorials-B2BUA#toc13 >> >> >> Now; everything seems to work OK except for the fact that 180 is >> not relayed and no ringing is ever heard on the A-side after >> listening to Media and while connecting to B-side. >> >> It seems to somehow be swallowed by B2B. It's passed to B2B which >> seems to not handle 180 while in bridging scenario? >> >> DBG:tm:local_reply: Passing provisional reply 180 to FIFO application >> .... >> DBG:b2b_logic:b2b_logic_notify_reply: Received a reply [180] >> while in BRIDGING scenario >> >> Even though A-side is connected (after listening to media) it >> would make sense to let the A-side play ringing while trying to >> reach the B-side. >> >> Any ideas of how I can solve this? >> >> The OpenSIPS log for handling 180 can be found here: >> https://pastebin.com/fPVgLrCG >> >> Thanks a lot for your kind support! >> >> //Andreas >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alameire at openip.fr Thu Jul 6 09:16:36 2017 From: alameire at openip.fr (Lameire Alexis) Date: Thu, 6 Jul 2017 15:16:36 +0200 Subject: [OpenSIPS-Users] Passive call recording using rtpproxy In-Reply-To: <841b5ac5-a7b5-039a-3276-583030c84368@opensips.org> References: <841b5ac5-a7b5-039a-3276-583030c84368@opensips.org> Message-ID: <129a4251-7147-d169-0d54-35fe4fe9f86e@openip.fr> Ok, As I understand you, to be able to get the RTP stream I need to place an inline component between client and IPBX or between SBC and upstream to copy and redirect with the correct ip the RTP to the dedicated server. I wouldn't like to do it this way. I would like to be able to do it by only port mirror traffic on the inbound of SBC/PBX to the server, mainly to avoid side effect to the service on a fail of the recording service. Let me know if I definitively not understand you. Regards Le 05/07/2017 à 16:38, Bogdan-Andrei Iancu a écrit : > Hi Alexis, > > To record the RTP on the your platform, you need first to get the RTP > to flow through your servers, right ? So you have to do RTP pinning - > to be transparent for the users (if recorded or not), you will have to > do it for all calls I guess. And here you can use RTPproxy to pin the > media. > > Now, once you have the RTP on your servers (via rtpproxy), you have 2 > options: > 1) get the recording directly from RTPProxy > 2) get the RTP via port mirroring > > Let me know if ok so far. > > Best regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > On 07/05/2017 11:24 AM, Lameire Alexis wrote: >> >> Hello, >> >> I would like to acheave call recording on a passive way. My setup >> will use a port mirroring on the switch on a dedicated port. >> >> Despite the presentation on the last summit, I can't properly >> understand how to properly acheave it. As I get rtpproxy will rewrite >> the SDP content to fix the media ip to the instance of rtpproxy, but >> this biavior is not suitable for a passive call recording. >> >> As I see on this[1] documentation I don't find a way to inhibate this >> biavior and just rely on the choiced RTP/RTSP port. >> >> In additions I will be faced by the issues related to port mirroring, >> the destination ip on sig and media packet will not match the local >> address and rtpproxy and opensips will not be able to bind the >> datagram socket. >> >> So, I think I have lost something, so if you could glow my mind you >> will be gracefully thanked. >> >> In addition of my questions, I would like to strongly thanks the >> presenter for is call recording experience, mainly due to the keys >> provided on the post task for files conversion. >> >> [1] http://www.opensips.org/html/docs/modules/1.7.x/rtpproxy.html >> >> -- >> Signature >> Alexis Lameire >> Ingénieur infrastructure >> >> +33 9 70 71 60 01 >> alameire at openip.fr 37/39, rue de Neuilly, 92110 Clichy >> http://openip.fr / http://my.openip.fr e-mail >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 6 09:27:41 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 6 Jul 2017 16:27:41 +0300 Subject: [OpenSIPS-Users] Passive call recording using rtpproxy In-Reply-To: <129a4251-7147-d169-0d54-35fe4fe9f86e@openip.fr> References: <841b5ac5-a7b5-039a-3276-583030c84368@opensips.org> <129a4251-7147-d169-0d54-35fe4fe9f86e@openip.fr> Message-ID: Hi Alexis, IF you have access to the SBC/PBX boxes/network and able to do port mirroring for the passing RTP, then it is great. You cannot use RTPproxy, but other tools that are able to do RTP capturing via port mirroring - take a look as sipcapture.org, they have some tools for this. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/06/2017 04:16 PM, Lameire Alexis wrote: > > Ok, > > As I understand you, to be able to get the RTP stream I need to place > an inline component between client and IPBX or between SBC and > upstream to copy and redirect with the correct ip the RTP to the > dedicated server. > > I wouldn't like to do it this way. I would like to be able to do it by > only port mirror traffic on the inbound of SBC/PBX to the server, > mainly to avoid side effect to the service on a fail of the recording > service. > > Let me know if I definitively not understand you. > > Regards > > > Le 05/07/2017 à 16:38, Bogdan-Andrei Iancu a écrit : >> Hi Alexis, >> >> To record the RTP on the your platform, you need first to get the RTP >> to flow through your servers, right ? So you have to do RTP pinning - >> to be transparent for the users (if recorded or not), you will have >> to do it for all calls I guess. And here you can use RTPproxy to pin >> the media. >> >> Now, once you have the RTP on your servers (via rtpproxy), you have 2 >> options: >> 1) get the recording directly from RTPProxy >> 2) get the RTP via port mirroring >> >> Let me know if ok so far. >> >> Best regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> On 07/05/2017 11:24 AM, Lameire Alexis wrote: >>> >>> Hello, >>> >>> I would like to acheave call recording on a passive way. My setup >>> will use a port mirroring on the switch on a dedicated port. >>> >>> Despite the presentation on the last summit, I can't properly >>> understand how to properly acheave it. As I get rtpproxy will >>> rewrite the SDP content to fix the media ip to the instance of >>> rtpproxy, but this biavior is not suitable for a passive call recording. >>> >>> As I see on this[1] documentation I don't find a way to inhibate >>> this biavior and just rely on the choiced RTP/RTSP port. >>> >>> In additions I will be faced by the issues related to port >>> mirroring, the destination ip on sig and media packet will not match >>> the local address and rtpproxy and opensips will not be able to bind >>> the datagram socket. >>> >>> So, I think I have lost something, so if you could glow my mind you >>> will be gracefully thanked. >>> >>> In addition of my questions, I would like to strongly thanks the >>> presenter for is call recording experience, mainly due to the keys >>> provided on the post task for files conversion. >>> >>> [1] http://www.opensips.org/html/docs/modules/1.7.x/rtpproxy.html >>> >>> -- >>> Signature >>> Alexis Lameire >>> Ingénieur infrastructure >>> >>> +33 9 70 71 60 01 >>> alameire at openip.fr 37/39, rue de Neuilly, 92110 Clichy >>> http://openip.fr / http://my.openip.fr e-mail >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 6 09:28:53 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 6 Jul 2017 16:28:53 +0300 Subject: [OpenSIPS-Users] Cluesterer and ul_add permanent In-Reply-To: <4647681.e5x6FOWdfS@samusenko> References: <4647681.e5x6FOWdfS@samusenko> Message-ID: Hi, Could you be more specific on what you are trying to do ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/05/2017 07:03 PM, Andrey wrote: > OpenSIPS 2.2.4 > Subject does not work. > Slave: WARNING:usrloc:receive_binary_packet: received bin packet from unknown > source: 192.168.1.2:4414 > and MI process shutdown. > >> select url from clusterer: > bin:192.168.1.2:8026 > bin:192.168.1.3:8026 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From pimenta at inatel.br Thu Jul 6 16:57:15 2017 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 6 Jul 2017 20:57:15 +0000 Subject: [OpenSIPS-Users] How sequential forking works with OpenSIPS. Message-ID: Dear OpenSIPS users, When sequential forking is used by means of OpenSIS, 1 - can we have every callee ringing simultaneously in sometime, when nobody answers the call? Or 2 - each callee will ring only after a previous one stop ringing if the call was not answered until that moment? If the answer is number 2, is it possible to change this behavior to put all callees ringing at same time in some moment? I would like to experiment sequential forking, just to see if helps me to avoid an UAC to mute dialogs (vide RFC 3960) when it receives multiple SIP 183 due to a parallel forking. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Thu Jul 6 17:21:44 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 7 Jul 2017 02:21:44 +0500 Subject: [OpenSIPS-Users] Could not start cdrtool 9.5.0 Message-ID: Good Day. I am trying to start cdrtool 9.5.0 after installation on Debian Jessie 8.3 having php 5.6.24 but it is failing with below errors. root at debian:~# /etc/init.d/cdrtool start [....] Starting cdrtool (via systemctl): cdrtool.serviceJun 5 02:47:30 debian ratingEngine.php[13021]: init_ok) { Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,'Error: Cannot start Rating Engine, fix the errors and try again'); Jun 5 02:47:30 debian ratingEngine.php[13021]: exit; Jun 5 02:47:30 debian ratingEngine.php[13021]: } Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating Engine started sucesfully, going to background..."); Jun 5 02:47:30 debian ratingEngine.php[13021]: // Go to the background Jun 5 02:47:30 debian ratingEngine.php[13021]: $d = new Daemon('/var/run/ratingEngine.pid'); Jun 5 02:47:30 debian ratingEngine.php[13021]: $d->start(); Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon = new socketDaemon(); Jun 5 02:47:30 debian ratingEngine.php[13021]: $server = $daemon->create_server('ratingEngineServer', 'ratingEngineClient', $RatingEngine['socketIP'], $RatingEngine['socketPort']); Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating Engine is now ready to serve network requests"); Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon->process(); Jun 5 02:47:30 debian ratingEngine.php[13021]: ?> Job for cdrtool.service failed. See 'systemctl status cdrtool.service' and 'journalctl -xn' for details. Jun 5 02:47:30 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. Jun 5 02:47:30 debian systemd[1]: Unit cdrtool.service entered failed state. failed! Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state. Jun 5 02:47:31 debian systemd[1]: cdrtool.service start request repeated too quickly, refusing to start. Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state. Even by manually running* /var/www/CDRTool/scripts/importRatingTables.php *script is printing its contents on console. root at debian:~# php /var/www/CDRTool/scripts/importRatingTables.php ImportCSVFiles(); if ($RatingTables->mustReload) { if (!reloadRatingEngineTables()) { print "Error: cannot connect to network rating engine\n"; } } ?> root at debian:~# php -v PHP 5.6.24-0+deb8u1 (cli) (built: Jul 26 2016 08:17:07) Copyright (c) 1997-2016 The PHP Group Zend Engine v2.6.0, Copyright (c) 1998-2016 Zend Technologies with Zend OPcache v7.0.6-dev, Copyright (c) 1999-2016, by Zend Technologies root at debian:~# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 8.3 (jessie) Release: 8.3 Codename: jessie Any pointer would be much appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Jul 6 20:10:38 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 07 Jul 2017 00:10:38 +0000 Subject: [OpenSIPS-Users] Could not start cdrtool 9.5.0 In-Reply-To: References: Message-ID: I didn't see any errors, in fact it says the engine is ready... Seems to me like a startup script problem, the startup has changed in debian, maybe cdrtool hasn't been updated? Have you tried starting it up manually instead of using the startup script? On Thu, Jul 6, 2017 at 11:25 PM Aqs Younas wrote: > Good Day. > > I am trying to start cdrtool 9.5.0 after installation on Debian Jessie 8.3 > having php 5.6.24 but it is failing with below errors. > > root at debian:~# /etc/init.d/cdrtool start > [....] Starting cdrtool (via systemctl): cdrtool.serviceJun 5 02:47:30 > debian ratingEngine.php[13021]: Jun 5 02:47:30 debian ratingEngine.php[13021]: set_time_limit (0); > Jun 5 02:47:30 debian ratingEngine.php[13021]: > ini_set('mbstring.func_overload', '0'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: ini_set('output_handler', > ''); > Jun 5 02:47:30 debian ratingEngine.php[13021]: @ob_end_flush(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: > require('/etc/cdrtool/global.inc'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: require('cdr_generic.php'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: require('rating.php'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: > require('rating_server.php'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: // Init Rating Engine > Jun 5 02:47:30 debian ratingEngine.php[13021]: > syslog(LOG_NOTICE,"Starting CDRTool Rating Engine..."); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $RatingEngineServer = new > RatingEngine(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: if > (!$RatingEngineServer->init_ok) { > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,'Error: > Cannot start Rating Engine, fix the errors and try again'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: exit; > Jun 5 02:47:30 debian ratingEngine.php[13021]: } > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating > Engine started sucesfully, going to background..."); > Jun 5 02:47:30 debian ratingEngine.php[13021]: // Go to the background > Jun 5 02:47:30 debian ratingEngine.php[13021]: $d = new > Daemon('/var/run/ratingEngine.pid'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $d->start(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon = new > socketDaemon(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $server = > $daemon->create_server('ratingEngineServer', 'ratingEngineClient', > $RatingEngine['socketIP'], $RatingEngine['socketPort']); > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating > Engine is now ready to serve network requests"); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon->process(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: ?> > Job for cdrtool.service failed. See 'systemctl status cdrtool.service' and > 'journalctl -xn' for details. > Jun 5 02:47:30 debian systemd[1]: Failed to start CDR mediation and > rating engine for Call Details Records.. > Jun 5 02:47:30 debian systemd[1]: Unit cdrtool.service entered failed > state. > failed! > Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and > rating engine for Call Details Records.. > Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed > state. > Jun 5 02:47:31 debian systemd[1]: cdrtool.service start request repeated > too quickly, refusing to start. > Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and > rating engine for Call Details Records.. > Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed > state. > > > Even by manually running* /var/www/CDRTool/scripts/importRatingTables.php > *script is printing its contents on console. > > root at debian:~# php /var/www/CDRTool/scripts/importRatingTables.php > require("/etc/cdrtool/global.inc"); > require('cdr_generic.php'); > require("rating.php"); > > set_time_limit(0); > > $lockFile=sprintf("/var/lock/CDRTool_import_rates.lock"); > $abort_text="Another import operation is in progress. Try again later.\n"; > > $f=fopen($lockFile,"w"); > if (flock($f, LOCK_EX + LOCK_NB, $w)) { > if ($w) { > print $abort_text; > syslog(LOG_NOTICE,$abort_text); > exit(2); > } > } else { > print $abort_text; > syslog(LOG_NOTICE,$abort_text); > exit(1); > } > > $RatingTables= new RatingTables(); > $RatingTables->ImportCSVFiles(); > > if ($RatingTables->mustReload) { > > if (!reloadRatingEngineTables()) { > print "Error: cannot connect to network rating engine\n"; > } > } > > ?> > > > root at debian:~# php -v > PHP 5.6.24-0+deb8u1 (cli) (built: Jul 26 2016 08:17:07) > Copyright (c) 1997-2016 The PHP Group > Zend Engine v2.6.0, Copyright (c) 1998-2016 Zend Technologies > with Zend OPcache v7.0.6-dev, Copyright (c) 1999-2016, by Zend > Technologies > > root at debian:~# lsb_release -a > No LSB modules are available. > Distributor ID: Debian > Description: Debian GNU/Linux 8.3 (jessie) > Release: 8.3 > Codename: jessie > > > Any pointer would be much appreciated. > > Thanks > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Fri Jul 7 05:22:09 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 7 Jul 2017 14:22:09 +0500 Subject: [OpenSIPS-Users] Could not start cdrtool 9.5.0 In-Reply-To: References: Message-ID: Thanks for reply, David. I think you might missed the part of log. *Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records..* *Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state.* *Jun 5 02:47:31 debian systemd[1]: cdrtool.service start request repeated too quickly, refusing to start.* *Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records..* *Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state.* Well by manually running the Cdrtool it is failing too. root at debian:/var/www/CDRTool/scripts# *php ratingEngine.php* init_ok) { syslog(LOG_NOTICE,'Error: Cannot start Rating Engine, fix the errors and try again'); exit; } syslog(LOG_NOTICE,"Rating Engine started sucesfully, going to background..."); // Go to the background $d = new Daemon('/var/run/ratingEngine.pid'); $d->start(); $daemon = new socketDaemon(); $server = $daemon->create_server('ratingEngineServer', 'ratingEngineClient', $RatingEngine['socketIP'], $RatingEngine['socketPort']); syslog(LOG_NOTICE,"Rating Engine is now ready to serve network requests"); $daemon->process(); ?> *It is simply printing the contents of script. * I think it is due to php version compatibility issue. It is having compatible issues with php 5.6 even though release announcements proclaimed this. Any pointer? Best Regards. On 7 July 2017 at 05:10, David Villasmil wrote: > I didn't see any errors, in fact it says the engine is ready... Seems to > me like a startup script problem, the startup has changed in debian, maybe > cdrtool hasn't been updated? Have you tried starting it up manually instead > of using the startup script? > On Thu, Jul 6, 2017 at 11:25 PM Aqs Younas wrote: > >> Good Day. >> >> I am trying to start cdrtool 9.5.0 after installation on Debian Jessie >> 8.3 having php 5.6.24 but it is failing with below errors. >> >> root at debian:~# /etc/init.d/cdrtool start >> [....] Starting cdrtool (via systemctl): cdrtool.serviceJun 5 02:47:30 >> debian ratingEngine.php[13021]: > Jun 5 02:47:30 debian ratingEngine.php[13021]: set_time_limit (0); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: ini_set('mbstring.func_overload', >> '0'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: ini_set('output_handler', >> ''); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: @ob_end_flush(); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: >> require('/etc/cdrtool/global.inc'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: >> require('cdr_generic.php'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: require('rating.php'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: >> require('rating_server.php'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: // Init Rating Engine >> Jun 5 02:47:30 debian ratingEngine.php[13021]: >> syslog(LOG_NOTICE,"Starting CDRTool Rating Engine..."); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: $RatingEngineServer = new >> RatingEngine(); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: if >> (!$RatingEngineServer->init_ok) { >> Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,'Error: >> Cannot start Rating Engine, fix the errors and try again'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: exit; >> Jun 5 02:47:30 debian ratingEngine.php[13021]: } >> Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating >> Engine started sucesfully, going to background..."); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: // Go to the background >> Jun 5 02:47:30 debian ratingEngine.php[13021]: $d = new >> Daemon('/var/run/ratingEngine.pid'); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: $d->start(); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon = new >> socketDaemon(); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: $server = >> $daemon->create_server('ratingEngineServer', 'ratingEngineClient', >> $RatingEngine['socketIP'], $RatingEngine['socketPort']); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating >> Engine is now ready to serve network requests"); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon->process(); >> Jun 5 02:47:30 debian ratingEngine.php[13021]: ?> >> Job for cdrtool.service failed. See 'systemctl status cdrtool.service' >> and 'journalctl -xn' for details. >> Jun 5 02:47:30 debian systemd[1]: Failed to start CDR mediation and >> rating engine for Call Details Records.. >> Jun 5 02:47:30 debian systemd[1]: Unit cdrtool.service entered failed >> state. >> failed! >> Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and >> rating engine for Call Details Records.. >> Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed >> state. >> Jun 5 02:47:31 debian systemd[1]: cdrtool.service start request repeated >> too quickly, refusing to start. >> Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and >> rating engine for Call Details Records.. >> Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed >> state. >> >> >> Even by manually running* /var/www/CDRTool/scripts/importRatingTables.php >> *script is printing its contents on console. >> >> root at debian:~# php /var/www/CDRTool/scripts/importRatingTables.php >> > require("/etc/cdrtool/global.inc"); >> require('cdr_generic.php'); >> require("rating.php"); >> >> set_time_limit(0); >> >> $lockFile=sprintf("/var/lock/CDRTool_import_rates.lock"); >> $abort_text="Another import operation is in progress. Try again later.\n"; >> >> $f=fopen($lockFile,"w"); >> if (flock($f, LOCK_EX + LOCK_NB, $w)) { >> if ($w) { >> print $abort_text; >> syslog(LOG_NOTICE,$abort_text); >> exit(2); >> } >> } else { >> print $abort_text; >> syslog(LOG_NOTICE,$abort_text); >> exit(1); >> } >> >> $RatingTables= new RatingTables(); >> $RatingTables->ImportCSVFiles(); >> >> if ($RatingTables->mustReload) { >> >> if (!reloadRatingEngineTables()) { >> print "Error: cannot connect to network rating engine\n"; >> } >> } >> >> ?> >> >> >> root at debian:~# php -v >> PHP 5.6.24-0+deb8u1 (cli) (built: Jul 26 2016 08:17:07) >> Copyright (c) 1997-2016 The PHP Group >> Zend Engine v2.6.0, Copyright (c) 1998-2016 Zend Technologies >> with Zend OPcache v7.0.6-dev, Copyright (c) 1999-2016, by Zend >> Technologies >> >> root at debian:~# lsb_release -a >> No LSB modules are available. >> Distributor ID: Debian >> Description: Debian GNU/Linux 8.3 (jessie) >> Release: 8.3 >> Codename: jessie >> >> >> Any pointer would be much appreciated. >> >> Thanks >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jul 7 05:26:47 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 7 Jul 2017 12:26:47 +0300 Subject: [OpenSIPS-Users] How sequential forking works with OpenSIPS. In-Reply-To: References: Message-ID: Hi Rodrigo, Your scenario 1) is not sequential forking, but parallel forking (if you have multiple callee reached in the same time). And 2), yes, it is serial / sequential forking. Both are supported by OpenSIPS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/06/2017 11:57 PM, Rodrigo Pimenta Carvalho wrote: > > > Dear OpenSIPS users, > > > When sequential forking is used by means of OpenSIS, > > > 1 - can we have every callee ringing simultaneously in sometime, when > nobody answers the call? > > > Or > > > 2 - each callee will ring only after a previous one stop ringing if > the call was not answered until that moment? > > > If the answer is number 2, is it possible to change this behavior to > put all callees ringing at same time in some moment? > > > I would like to experiment sequential forking, just to see if helps me > to avoid an UAC to mute dialogs (vide RFC 3960) when it receives > multiple SIP 183 due to a parallel forking. > > > Best regards! > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liran.aknin at vonage.com Thu Jul 6 07:27:14 2017 From: liran.aknin at vonage.com (Aknin, Liran) Date: Thu, 6 Jul 2017 14:27:14 +0300 Subject: [OpenSIPS-Users] Using usrloc module with Redis DB In-Reply-To: <9923591e-8cb7-1220-35a4-9289143ec7b0@opensips.org> References: <797906ba-f4f1-2196-196d-7d8bc1ce4e15@opensips.org> <9923591e-8cb7-1220-35a4-9289143ec7b0@opensips.org> Message-ID: Thanks again Bogdan, I assumed that what you meant. Using SQL, preferably PostgreSQL, is it possible to implement clustering of some opensips servers, in terms of shared read-write of userloc? Regards, Liran On Thu, Jul 6, 2017 at 1:35 PM, Bogdan-Andrei Iancu wrote: > Hi Liran, > > yes, typo, it should have been "there is no straight way to use an noSQL > backends" > > But this is for location. Otherwise, OpenSIPS has a cachedb_redis module > you can use from script as a key-value cache . > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/06/2017 08:29 AM, Aknin, Liran wrote: > > Hi Bogdan, > > Thanks for your prompt reply. > > Just to make sure I understood you correctly, since I was actually asking > about Redis - you mean that there is support for *NO-SQL* DB, right? > Is usrloc over Redis going to be introduced in 2.4? > > thanks, > Liran > > On Wed, Jul 5, 2017 at 5:53 PM, Bogdan-Andrei Iancu > wrote: > >> Hi Liran, >> >> Right now, due the complexity of the usrloc data, there is no straight >> way to use an SQL backend (there is a need for list support with atomic >> ops). We are working for such support for 2.4 release. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/05/2017 04:23 PM, Aknin, Liran via Users wrote: >> >> Hi Bogdan and all, >> >> We would like to cluster few opensips servers, in terms of sharing usrloc >> data between them. We already use Redis on these servers and we want to >> take advantage of that, as well as enjoining faster access comparing to SQL. >> >> Is it possible to use usrloc module with Redis? >> If so, can you please provide us with an example of defining usrloc's >> db_url parameter to do so? >> >> Thanks and regards, >> Liran >> >> >> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From achalkov at yandex.ru Fri Jul 7 09:58:04 2017 From: achalkov at yandex.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=) Date: Fri, 07 Jul 2017 16:58:04 +0300 Subject: [OpenSIPS-Users] event-rabbitmq message transferring Message-ID: <624491499435884@web49j.yandex.ru> An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jul 7 10:55:36 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 7 Jul 2017 17:55:36 +0300 Subject: [OpenSIPS-Users] event-rabbitmq message transferring In-Reply-To: <624491499435884@web49j.yandex.ru> References: <624491499435884@web49j.yandex.ru> Message-ID: <2663bec8-4dd0-ec49-2a60-488471696405@opensips.org> Unfortunately there is currently no way to control which messages should be delivered or which are not. It would be nice to be able to configure a windowfor this, but this is currently not implemented. Please open a feature request for this so we can keep track of these features. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/07/2017 04:58 PM, Чалков Артём wrote: > Hi all. > I use opensips-2.1/2.2/2.3 (all with the same behavior) and > event-rabbitmq for publishing some events. > If rabbitmq-server is unreachable, opensips start caching outgoing > rabbitmq-messages and after rabbitmq-server goes up again - send all > cached messages to it. If event rate is high, it could lead to > devivery outdated data to rabbitmq. > Is there some way to drop these messages while rabbitmq-server is > unreachable? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From samusenko at msm.ru Fri Jul 7 11:18:43 2017 From: samusenko at msm.ru (=?utf-8?B?0KHQsNC80YPRgdC10L3QutC+INCQ0L3QtNGA0LXQuQ==?=) Date: Fri, 07 Jul 2017 18:18:43 +0300 Subject: [OpenSIPS-Users] Cluesterer and ul_add permanent In-Reply-To: References: <4647681.e5x6FOWdfS@samusenko> Message-ID: <2068901.zWUMlDRjOt@samusenko> I try to setup HA OpenSIPS 2.2.4 Using modules clusterer and usrloc. Master management IP 192.168.1.2 Slave management IP 192.168.1.3 SIP REGISTER replication OK. REGISTER -> master master -> slave Error with MI: on master run opensipsctl ul add ............ introduce a permanent usrloc entry or XMLRPC ul_add('location', ...) Brrr, today no warning:usrloc... but on master there is DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by the remote peer! CRITICAL:core:receive_fd: EOF on 11 DBG:core:handle_worker: dead child 2, pid 7562 (shutting down?) On четверг, 6 июля 2017 г. 16:28:53 MSK Bogdan-Andrei Iancu wrote: > Hi, > > Could you be more specific on what you are trying to do ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/05/2017 07:03 PM, Andrey wrote: > > OpenSIPS 2.2.4 > > Subject does not work. > > Slave: WARNING:usrloc:receive_binary_packet: received bin packet from > > unknown source: 192.168.1.2:4414 > > and MI process shutdown. > > > >> select url from clusterer: > > bin:192.168.1.2:8026 > > bin:192.168.1.3:8026 > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From tac.voice at gmail.com Sat Jul 8 02:21:52 2017 From: tac.voice at gmail.com (Voice TAC) Date: Sat, 8 Jul 2017 09:21:52 +0300 Subject: [OpenSIPS-Users] UDP and TCP Listener Message-ID: Hello, Can I know based on what I can set the values of children in the following: children=4 tcp_children=10 I found that the values are: Number of processes per UDP listener. Total number of processes for all TCP listeners. Is there a relation between number of calls and listeners? I tried to search about listeners meaning but I did not find a clear explanation. -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Jul 10 05:59:44 2017 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 10 Jul 2017 12:59:44 +0300 Subject: [OpenSIPS-Users] UDP and TCP Listener In-Reply-To: References: Message-ID: <3122098d-52cf-4b1a-a4b1-2911ac1a5bef@opensips.org> Here is a similar thread [1] TL;DR: the more blocking I/O operations (DB/HTTP/ENUM queries) you include in your script, the more children will be required if you still want your OpenSIPS to be able to ingest thousands of CPS. [1]: http://lists.opensips.org/pipermail/users/2013-December/027530.html Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 08.07.2017 09:21, Voice TAC wrote: > Hello, > > Can I know based on what I can set the values of children in the > following: > > children=4 > > tcp_children=10 > > > I found that the values are: > > Number of processes per UDP listener. > > Total number of processes for all TCP listeners. > > > Is there a relation between number of calls and listeners? > I tried to search about listeners meaning but I did not find a clear > explanation. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Mon Jul 10 06:55:25 2017 From: xaled at web.de (xaled) Date: Mon, 10 Jul 2017 12:55:25 +0200 Subject: [OpenSIPS-Users] negative values in math_rpn Message-ID: <01dd01d2f96b$09530ba0$1bf922e0$@web.de> Hi, am I doing something wrong, or can math_rpn not handle negative values? $var(neg) = "-6.135"; $var(neg_int) = $(var(neg){s.int}); math_rpn("$var(neg_int) abs", "$avp(result)"); 2017-07-10T12:47:59.804008+02:00 fra-ivr01 /usr/local/sbin/opensips[10971]: DBG:mathops:w_evaluate_rpn: Evaluating expression: -6 abs 2017-07-10T12:47:59.804339+02:00 fra-ivr01 /usr/local/sbin/opensips[10971]: WARNING:mathops:get_rpn_op: Parse expr error: Invalid operator! <-6> 2017-07-10T12:47:59.804669+02:00 fra-ivr01 /usr/local/sbin/opensips[10971]: ERROR:mathops:evaluate_rpn: Failed to parse RPN! Thanks, xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Jul 10 07:05:55 2017 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 10 Jul 2017 14:05:55 +0300 Subject: [OpenSIPS-Users] negative values in math_rpn In-Reply-To: <01dd01d2f96b$09530ba0$1bf922e0$@web.de> References: <01dd01d2f96b$09530ba0$1bf922e0$@web.de> Message-ID: <3b591099-8dd0-ac39-e556-43bbde5d0881@opensips.org> I suggest you use math_eval() - it's much more human friendly. The following should work: $var(neg) = "-6.135"; $var(neg_int) = $(var(neg){s.int}); math_eval("abs($var(neg_int))", "$var(result)"); OTOH, if you really want to use RPN, this will fix it: math_rpn("abs $var(neg_int)", "$var(result)"); Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 10.07.2017 13:55, xaled wrote: > > Hi, > > am I doing something wrong, or can math_rpn not handle negative values? > > $var(neg) = "-6.135"; > > $var(neg_int) = $(var(neg){s.int}); > > math_rpn("$var(neg_int) abs", "$avp(result)"); > > 2017-07-10T12:47:59.804008+02:00 fra-ivr01 > /usr/local/sbin/opensips[10971]: DBG:mathops:w_evaluate_rpn: > Evaluating expression: -6 abs > > 2017-07-10T12:47:59.804339+02:00 fra-ivr01 > /usr/local/sbin/opensips[10971]: WARNING:mathops:get_rpn_op: Parse > expr error: Invalid operator! <-6> > > 2017-07-10T12:47:59.804669+02:00 fra-ivr01 > /usr/local/sbin/opensips[10971]: ERROR:mathops:evaluate_rpn: Failed to > parse RPN! > > Thanks, > > xaled > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From tijmen at ag-projects.com Mon Jul 10 08:49:15 2017 From: tijmen at ag-projects.com (Tijmen de Mes) Date: Mon, 10 Jul 2017 14:49:15 +0200 Subject: [OpenSIPS-Users] Could not start cdrtool 9.5.0 In-Reply-To: References: Message-ID: Hi, Unfortunately some code still uses short open tags in PHP. Please check if short_open_tag is set to On in /etc/php5/apache2/php.ini and /etc/php5/cli/php.ini. Best regards, Tijmen de Mes — AG Projects > On 7 jul. 2017, at 11:22, Aqs Younas wrote: > > Thanks for reply, David. > > I think you might missed the part of log. > > Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. > Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state. > Jun 5 02:47:31 debian systemd[1]: cdrtool.service start request repeated too quickly, refusing to start. > Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. > Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state. > > Well by manually running the Cdrtool it is failing too. > > root at debian:/var/www/CDRTool/scripts# php ratingEngine.php > set_time_limit (0); > ini_set('mbstring.func_overload', '0'); > ini_set('output_handler', ''); > @ob_end_flush(); > > require('/etc/cdrtool/global.inc'); > require('cdr_generic.php'); > require('rating.php'); > require('rating_server.php'); > > // Init Rating Engine > syslog(LOG_NOTICE,"Starting CDRTool Rating Engine..."); > > $RatingEngineServer = new RatingEngine(); > > if (!$RatingEngineServer->init_ok) { > syslog(LOG_NOTICE,'Error: Cannot start Rating Engine, fix the errors and try again'); > exit; > } > > syslog(LOG_NOTICE,"Rating Engine started sucesfully, going to background..."); > > // Go to the background > $d = new Daemon('/var/run/ratingEngine.pid'); > $d->start(); > > $daemon = new socketDaemon(); > $server = $daemon->create_server('ratingEngineServer', 'ratingEngineClient', $RatingEngine['socketIP'], $RatingEngine['socketPort']); > > syslog(LOG_NOTICE,"Rating Engine is now ready to serve network requests"); > > $daemon->process(); > > ?> > > It is simply printing the contents of script. > > I think it is due to php version compatibility issue. It is having compatible issues with php 5.6 even though release announcements proclaimed this. > > Any pointer? > Best Regards. > > On 7 July 2017 at 05:10, David Villasmil > wrote: > I didn't see any errors, in fact it says the engine is ready... Seems to me like a startup script problem, the startup has changed in debian, maybe cdrtool hasn't been updated? Have you tried starting it up manually instead of using the startup script? > On Thu, Jul 6, 2017 at 11:25 PM Aqs Younas > wrote: > Good Day. > > I am trying to start cdrtool 9.5.0 after installation on Debian Jessie 8.3 having php 5.6.24 but it is failing with below errors. > > root at debian:~# /etc/init.d/cdrtool start > [....] Starting cdrtool (via systemctl): cdrtool.serviceJun 5 02:47:30 debian ratingEngine.php[13021]: Jun 5 02:47:30 debian ratingEngine.php[13021]: set_time_limit (0); > Jun 5 02:47:30 debian ratingEngine.php[13021]: ini_set('mbstring.func_overload', '0'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: ini_set('output_handler', ''); > Jun 5 02:47:30 debian ratingEngine.php[13021]: @ob_end_flush(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: require('/etc/cdrtool/global.inc'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: require('cdr_generic.php'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: require('rating.php'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: require('rating_server.php'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: // Init Rating Engine > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Starting CDRTool Rating Engine..."); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $RatingEngineServer = new RatingEngine(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: if (!$RatingEngineServer->init_ok) { > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,'Error: Cannot start Rating Engine, fix the errors and try again'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: exit; > Jun 5 02:47:30 debian ratingEngine.php[13021]: } > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating Engine started sucesfully, going to background..."); > Jun 5 02:47:30 debian ratingEngine.php[13021]: // Go to the background > Jun 5 02:47:30 debian ratingEngine.php[13021]: $d = new Daemon('/var/run/ratingEngine.pid'); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $d->start(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon = new socketDaemon(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $server = $daemon->create_server('ratingEngineServer', 'ratingEngineClient', $RatingEngine['socketIP'], $RatingEngine['socketPort']); > Jun 5 02:47:30 debian ratingEngine.php[13021]: syslog(LOG_NOTICE,"Rating Engine is now ready to serve network requests"); > Jun 5 02:47:30 debian ratingEngine.php[13021]: $daemon->process(); > Jun 5 02:47:30 debian ratingEngine.php[13021]: ?> > Job for cdrtool.service failed. See 'systemctl status cdrtool.service' and 'journalctl -xn' for details. > Jun 5 02:47:30 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. > Jun 5 02:47:30 debian systemd[1]: Unit cdrtool.service entered failed state. > failed! > Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. > Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state. > Jun 5 02:47:31 debian systemd[1]: cdrtool.service start request repeated too quickly, refusing to start. > Jun 5 02:47:31 debian systemd[1]: Failed to start CDR mediation and rating engine for Call Details Records.. > Jun 5 02:47:31 debian systemd[1]: Unit cdrtool.service entered failed state. > > > Even by manually running /var/www/CDRTool/scripts/importRatingTables.php script is printing its contents on console. > > root at debian:~# php /var/www/CDRTool/scripts/importRatingTables.php > require("/etc/cdrtool/global.inc"); > require('cdr_generic.php'); > require("rating.php"); > > set_time_limit(0); > > $lockFile=sprintf("/var/lock/CDRTool_import_rates.lock"); > $abort_text="Another import operation is in progress. Try again later.\n"; > > $f=fopen($lockFile,"w"); > if (flock($f, LOCK_EX + LOCK_NB, $w)) { > if ($w) { > print $abort_text; > syslog(LOG_NOTICE,$abort_text); > exit(2); > } > } else { > print $abort_text; > syslog(LOG_NOTICE,$abort_text); > exit(1); > } > > $RatingTables= new RatingTables(); > $RatingTables->ImportCSVFiles(); > > if ($RatingTables->mustReload) { > > if (!reloadRatingEngineTables()) { > print "Error: cannot connect to network rating engine\n"; > } > } > > ?> > > > root at debian:~# php -v > PHP 5.6.24-0+deb8u1 (cli) (built: Jul 26 2016 08:17:07) > Copyright (c) 1997-2016 The PHP Group > Zend Engine v2.6.0, Copyright (c) 1998-2016 Zend Technologies > with Zend OPcache v7.0.6-dev, Copyright (c) 1999-2016, by Zend Technologies > > root at debian:~# lsb_release -a > No LSB modules are available. > Distributor ID: Debian > Description: Debian GNU/Linux 8.3 (jessie) > Release: 8.3 > Codename: jessie > > > Any pointer would be much appreciated. > > Thanks > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 10 10:09:22 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 10 Jul 2017 17:09:22 +0300 Subject: [OpenSIPS-Users] Using usrloc module with Redis DB In-Reply-To: References: <797906ba-f4f1-2196-196d-7d8bc1ce4e15@opensips.org> <9923591e-8cb7-1220-35a4-9289143ec7b0@opensips.org> Message-ID: <3c5a4dba-6e0a-1d89-3b51-87d2f069355e@opensips.org> Hi Liran, Keep in mind that in the USRLOC module, the primary data storage is memory, while the DB is just for periodic flushing (write only) for restart persistence - the only moment when the DB is read is at startup, when the memory is populated. So, in order to make multiple opensips to share usrloc content via DB is to have the DB as primary data storage and this happens exclusively for DB ONLY mode (where there is no memory cache). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/06/2017 02:27 PM, Aknin, Liran wrote: > Thanks again Bogdan, I assumed that what you meant. > > Using SQL, preferably PostgreSQL, is it possible to implement > clustering of some opensips servers, in terms of shared read-write of > userloc? > > Regards, > Liran > > On Thu, Jul 6, 2017 at 1:35 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Liran, > > yes, typo, it should have been "there is no straight way to use an > noSQL backends" > > But this is for location. Otherwise, OpenSIPS has a cachedb_redis > module you can use from script as a key-value cache . > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/06/2017 08:29 AM, Aknin, Liran wrote: >> Hi Bogdan, >> >> Thanks for your prompt reply. >> >> Just to make sure I understood you correctly, since I was >> actually asking about Redis - you mean that there is support for >> *NO-SQL* DB, right? >> Is usrloc over Redis going to be introduced in 2.4? >> >> thanks, >> Liran >> >> On Wed, Jul 5, 2017 at 5:53 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Liran, >> >> Right now, due the complexity of the usrloc data, there is no >> straight way to use an SQL backend (there is a need for list >> support with atomic ops). We are working for such support for >> 2.4 release. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> >> On 07/05/2017 04:23 PM, Aknin, Liran via Users wrote: >>> Hi Bogdan and all, >>> >>> We would like to cluster few opensips servers, in terms of >>> sharing usrloc data between them. We already use Redis on >>> these servers and we want to take advantage of that, as well >>> as enjoining faster access comparing to SQL. >>> >>> Is it possible to use usrloc module with Redis? >>> If so, can you please provide us with an example of defining >>> usrloc's db_url parameter to do so? >>> >>> Thanks and regards, >>> Liran >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 10 10:14:51 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 10 Jul 2017 17:14:51 +0300 Subject: [OpenSIPS-Users] Forking Non-INVITE Requests In-Reply-To: References: <657253FA-AD35-4CCD-8FE2-53EB88350983@gmail.com> Message-ID: <49698cbe-8e07-0838-f89a-bfecc14ff10d@opensips.org> Hi Chad, No, the functionality, as it is now, does not provide any feedback on the reply codes you received on the replicated forks. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/07/2017 06:49 PM, Chad Attermann wrote: > > Thanks Bogdan, I had seen that function and was wondering if it might > be applicable. I don’t suppose there wold be a way to track the > responses from each destination, and have some logic like if all > designations fail, then return failure code to orig side, otherwise if > at least one destination succeeds then send success to orig? > > Thanks again, > > Chad. > >> On Jun 30, 2017, at 2:06 AM, Bogdan-Andrei Iancu > > wrote: >> >> Hi Chad, >> >> I would say the t_replicate() is what you are looking for : >> http://www.opensips.org/html/docs/modules/2.3.x/tm.html#treplicate >> >> Regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> On 06/29/2017 07:31 PM, Chad Attermann wrote: >>> Hi All, >>> >>> I would like to be able to “broadcast” some non-INVITE requests >>> (like NOTIFY and MESSAGE) to multiple registered endpoints. The >>> problem with using parallel forking is that the request is not >>> reliably delivered to all registered endpoints since retransmission >>> of the request stops after the first success response is received. >>> What I need is some hybrid of parallel and serial forking, where >>> requests to all registered are sent at once, but each is treated as >>> a separate transaction so that the message is sent (or at least >>> attempted) reliably to each endpoint. >>> >>> This sort of situation is mentioned in the docs for the TM module… >>> >>> "UAC--it is a very simplistic code which allows you to generate your >>> own transactions. Particularly useful for things like NOTIFYs or >>> IM gateways.” >>> >>> … but there is no mention of *how* to use it. I’m sure there is >>> probably a simple script for this, but I haven’t had any success >>> searching the mailing list or the Internet at large for details. I >>> would appreciate if anybody could provide details or a sample script >>> demonstrating how to create a separate transaction for each branch. >>> >>> Thanks! >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Mon Jul 10 11:07:02 2017 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Mon, 10 Jul 2017 12:07:02 -0300 Subject: [OpenSIPS-Users] Looking for Mediaproxy developer Message-ID: I'm looking for help to customize some things on mediaproxy software. Need to: 1) Fix some bugs 2) Implement new features Please contact me for details. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Mon Jul 10 12:24:17 2017 From: xaled at web.de (xaled) Date: Mon, 10 Jul 2017 18:24:17 +0200 Subject: [OpenSIPS-Users] negative values in math_rpn In-Reply-To: <3b591099-8dd0-ac39-e556-43bbde5d0881@opensips.org> References: <01dd01d2f96b$09530ba0$1bf922e0$@web.de> <3b591099-8dd0-ac39-e556-43bbde5d0881@opensips.org> Message-ID: <02ce01d2f998$fb548050$f1fd80f0$@web.de> Hi Liviu, thanks for pointing to the current state of math_eval. It is definitely much more user friendly. Googling “opensips math_eval” gave me only the 1.10 version of math_eval documentation with only the binary operations at that time. As usual - one should have looked better. I was obviously stuck with the assumption that a value should always come first in an RPN expression. Greetings, xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu Sent: Montag, 10. Juli 2017 13:06 To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] negative values in math_rpn I suggest you use math_eval() - it's much more human friendly. The following should work: $var(neg) = "-6.135"; $var(neg_int) = $(var(neg){s.int}); math_eval("abs($var(neg_int))", "$var(result)"); OTOH, if you really want to use RPN, this will fix it: math_rpn("abs $var(neg_int)", "$var(result)"); Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 10.07.2017 13:55, xaled wrote: Hi, am I doing something wrong, or can math_rpn not handle negative values? $var(neg) = "-6.135"; $var(neg_int) = $(var(neg){s.int}); math_rpn("$var(neg_int) abs", "$avp(result)"); 2017-07-10T12:47:59.804008+02:00 fra-ivr01 /usr/local/sbin/opensips[10971]: DBG:mathops:w_evaluate_rpn: Evaluating expression: -6 abs 2017-07-10T12:47:59.804339+02:00 fra-ivr01 /usr/local/sbin/opensips[10971]: WARNING:mathops:get_rpn_op: Parse expr error: Invalid operator! <-6> 2017-07-10T12:47:59.804669+02:00 fra-ivr01 /usr/local/sbin/opensips[10971]: ERROR:mathops:evaluate_rpn: Failed to parse RPN! Thanks, xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Jul 10 12:21:52 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 10 Jul 2017 19:21:52 +0300 Subject: [OpenSIPS-Users] Cluesterer and ul_add permanent In-Reply-To: <2068901.zWUMlDRjOt@samusenko> References: <4647681.e5x6FOWdfS@samusenko> <2068901.zWUMlDRjOt@samusenko> Message-ID: <9ea53d0a-8112-ae55-b6dc-0eafdf87270f@opensips.org> Hi, Ok, so you configured a two nodes cluster (using clusterer) and configured the USRLOC replication via the clusterer support. And when you add a static contact via "ul_add", you get an error with 2.2.4 and possible a crash on trunk ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/07/2017 06:18 PM, Самусенко Андрей wrote: > I try to setup HA OpenSIPS 2.2.4 > Using modules clusterer and usrloc. > > Master management IP 192.168.1.2 > Slave management IP 192.168.1.3 > > SIP REGISTER replication OK. > REGISTER -> master > master -> slave > > Error with MI: > on master run opensipsctl ul add ............ introduce a > permanent usrloc entry > or XMLRPC ul_add('location', ...) > > Brrr, today no warning:usrloc... but on master there is > DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by > the remote peer! > CRITICAL:core:receive_fd: EOF on 11 > DBG:core:handle_worker: dead child 2, pid 7562 (shutting down?) > > On четверг, 6 июля 2017 г. 16:28:53 MSK Bogdan-Andrei Iancu wrote: >> Hi, >> >> Could you be more specific on what you are trying to do ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/05/2017 07:03 PM, Andrey wrote: >>> OpenSIPS 2.2.4 >>> Subject does not work. >>> Slave: WARNING:usrloc:receive_binary_packet: received bin packet from >>> unknown source: 192.168.1.2:4414 >>> and MI process shutdown. >>> >>>> select url from clusterer: >>> bin:192.168.1.2:8026 >>> bin:192.168.1.3:8026 >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From karthickrm at sardonyx.in Mon Jul 10 03:11:55 2017 From: karthickrm at sardonyx.in (KARTHICKRM) Date: Mon, 10 Jul 2017 12:41:55 +0530 Subject: [OpenSIPS-Users] 20170710-Support Needed to install OpenSIPS Control Panel Message-ID: <000601d2f94b$d1ea1c90$75be55b0$@sardonyx.in> Hi, This is Karthick. I am using Ubuntu 17.04 version. I had installed OpenSIPS 2.3.1 version. And I installed Control Panel 7.2.3 . To install the OpenSIPS, I just referred and followed the link https://www.linuxhelp.com/how-to-install-opensips-on-ubuntu-17-04/ To install the Control Panel 7.2.3 I just followed the link https://www.vultr.com/docs/how-to-install-opensips-control-panel-on-ubuntu-1 6-04 It gives the OpenSIPS Control Panel Login Screen. But at the time of login it gives the error message Error while connecting : MDB2 Error: not found How to solve this issue? Please give the solution ASAP. Note: Latest MDB2 Driver is installed for MySQL. (MDB2-Driver-MYSQL 1.5.0b4) and MDB2 Version is 2.5.0b5 Thanks, Karthick.R.M +91 8124774480 +91 9698289216 -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Jul 11 03:07:19 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 11 Jul 2017 10:07:19 +0300 Subject: [OpenSIPS-Users] 20170710-Support Needed to install OpenSIPS Control Panel In-Reply-To: <000601d2f94b$d1ea1c90$75be55b0$@sardonyx.in> References: <000601d2f94b$d1ea1c90$75be55b0$@sardonyx.in> Message-ID: <130d7667-09ed-b6f1-c26b-1988d2ea34b9@opensips.org> Hi, Karthick! Are you using perl 7.0? If so, OpenSIPS Control Panel is not yet compatible with perl 7.0, so you will have to downgrade perl to the 5.x version. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/10/2017 10:11 AM, KARTHICKRM wrote: > > Hi, > > This is Karthick. I am using *Ubuntu 17.04 version.* > > I had installed *OpenSIPS 2.3.1 version*. And I installed *Control > Panel 7.2.3* . > > To install the OpenSIPS, I just referred and followed the link > _https://www.linuxhelp.com/how-to-install-opensips-on-ubuntu-17-04/_ > > To install the Control Panel 7.2.3 I just followed the link > _https://www.vultr.com/docs/how-to-install-opensips-control-panel-on-ubuntu-16-04 > _ > > __ > > It gives the OpenSIPS Control Panel Login Screen. But at the time of > login it gives the error message > > *Error while connecting : MDB2 Error: not found* > > *How to solve this issue? Please give the solution ASAP.* > > *_Note_*: > > Latest MDB2 Driver is installed for MySQL. (*MDB2-Driver-MYSQL > 1.5.0b4*) and*MDB2 Version is 2.5.0b5* > > *Thanks,* > > */Karthick.R.M/**//* > > /+91 8124774480/ > > /+91 9698289216/ > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mlittlet5 at gmail.com Tue Jul 11 13:54:06 2017 From: mlittlet5 at gmail.com (mike Little) Date: Tue, 11 Jul 2017 13:54:06 -0400 Subject: [OpenSIPS-Users] Change Options method into invite Message-ID: We would like to know if it is possible to have anther sip proxy send a sip options message to opensips and have opensips change the sip options message into a sip invite to send to a sip server? The reason we are looking to use the invite versus sip options is to have the sip server run through some dialplan code to verify it is available to process calls. TIA From tito at xsvoce.com Tue Jul 11 14:58:28 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 11 Jul 2017 14:58:28 -0400 Subject: [OpenSIPS-Users] compile with openssl version Message-ID: Group, I've updated openssl in order to use opensips 2.3 but I am having issues after compiling and running openssl version -a OpenSSL 1.0.2k 26 Jan 2017 built on: reproducible build, date unspecified platform: linux-x86_64 options: bn(64,64) rc4(8x,int) des(idx,cisc,16,int) idea(int) blowfish(idx) compiler: gcc -I. -I.. -I../include -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -m64 -DL_ENDIAN -O3 -Wall -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM -DECP_NISTZ256_ASM OPENSSLDIR: "/usr/local/ssl" but when I run opensips I get ERROR:tls_mgm:mod_init: unable to set the memory allocation functions Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: ERROR:tls_mgm:mod_init: NOTE: check if you are using openssl 1.0.1e-fips, (or other FIPS version of openssl, as this is known to be broken; if so, you need to upgrade or downgrade to a different openssl version! Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: ERROR:tls_mgm:mod_init: current version: OpenSSL 1.0.1e-fips 11 Feb 2013 How so I force opensips to use the newer version?? Thanks, Tito -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Jul 11 15:46:04 2017 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 11 Jul 2017 22:46:04 +0300 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: Message-ID: It looks like your distro's libssl still has priority over the custom one. To avoid both uninstalling libssl and forcing all apps to use the newest library, I suggest you compile a hardcoded search path into tls_mgm.so. Just make a small modification in modules/tls_mgm/Makefile, like in this example: LIBS += -Wl,-rpath /home/liviu/lib $(shell $(SSL_BUILDER) --libs) Compile the tls_mgm, and if all goes well, the linker should spot the custom libssl first: [liviu ◄ Y510P opensips (master)]$ ldd modules/tls_mgm/tls_mgm.so linux-vdso.so.1 => (0x00007ffff040d000) libssl.so.1.0.0 => /home/liviu/lib/libssl.so.1.0.0 (0x00007fd9cde0a000) <---- the forced "runtime path" is working! libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fd9cda21000) libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x00007fd9cd5dc000) /lib64/ld-linux-x86-64.so.2 (0x000055a69a1b7000) libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007fd9cd3d8000) Another solution could be: echo "/usr/local/lib" > /etc/ld.so.conf.d/libssl.conf; ldconfig But note that this will "upgrade" the library for all apps in your system that require it. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 11.07.2017 21:58, Tito Cumpen wrote: > Group, > > > I've updated openssl in order to use opensips 2.3 but I am having > issues after compiling and running > > > openssl version -a > OpenSSL 1.0.2k 26 Jan 2017 > built on: reproducible build, date unspecified > platform: linux-x86_64 > options: bn(64,64) rc4(8x,int) des(idx,cisc,16,int) idea(int) > blowfish(idx) > compiler: gcc -I. -I.. -I../include -DOPENSSL_THREADS -D_REENTRANT > -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -m64 -DL_ENDIAN -O3 -Wall > -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 > -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM > -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM > -DGHASH_ASM -DECP_NISTZ256_ASM > OPENSSLDIR: "/usr/local/ssl" > > > but when I run opensips I get > > ERROR:tls_mgm:mod_init: unable to set the memory allocation functions > Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: > ERROR:tls_mgm:mod_init: NOTE: check if you are using openssl > 1.0.1e-fips, (or other FIPS version of openssl, as this is known to be > broken; if so, you need to upgrade or downgrade to a different openssl > version! > Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: > ERROR:tls_mgm:mod_init: current version: OpenSSL 1.0.1e-fips 11 Feb 2013 > > > How so I force opensips to use the newer version?? > > Thanks, > Tito > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Tue Jul 11 15:51:06 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Tue, 11 Jul 2017 19:51:06 +0000 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: Message-ID: Why hardcode it, just use LD_LIBRARY_PATH Robert From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu Sent: Tuesday, July 11, 2017 3:46 PM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] compile with openssl version It looks like your distro's libssl still has priority over the custom one. To avoid both uninstalling libssl and forcing all apps to use the newest library, I suggest you compile a hardcoded search path into tls_mgm.so. Just make a small modification in modules/tls_mgm/Makefile, like in this example: LIBS += -Wl,-rpath /home/liviu/lib $(shell $(SSL_BUILDER) --libs) Compile the tls_mgm, and if all goes well, the linker should spot the custom libssl first: [liviu ◄ Y510P opensips (master)]$ ldd modules/tls_mgm/tls_mgm.so linux-vdso.so.1 => (0x00007ffff040d000) libssl.so.1.0.0 => /home/liviu/lib/libssl.so.1.0.0 (0x00007fd9cde0a000) <---- the forced "runtime path" is working! libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fd9cda21000) libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 (0x00007fd9cd5dc000) /lib64/ld-linux-x86-64.so.2 (0x000055a69a1b7000) libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007fd9cd3d8000) Another solution could be: echo "/usr/local/lib" > /etc/ld.so.conf.d/libssl.conf; ldconfig But note that this will "upgrade" the library for all apps in your system that require it. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 11.07.2017 21:58, Tito Cumpen wrote: Group, I've updated openssl in order to use opensips 2.3 but I am having issues after compiling and running openssl version -a OpenSSL 1.0.2k 26 Jan 2017 built on: reproducible build, date unspecified platform: linux-x86_64 options: bn(64,64) rc4(8x,int) des(idx,cisc,16,int) idea(int) blowfish(idx) compiler: gcc -I. -I.. -I../include -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -m64 -DL_ENDIAN -O3 -Wall -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM -DECP_NISTZ256_ASM OPENSSLDIR: "/usr/local/ssl" but when I run opensips I get ERROR:tls_mgm:mod_init: unable to set the memory allocation functions Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: ERROR:tls_mgm:mod_init: NOTE: check if you are using openssl 1.0.1e-fips, (or other FIPS version of openssl, as this is known to be broken; if so, you need to upgrade or downgrade to a different openssl version! Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: ERROR:tls_mgm:mod_init: current version: OpenSSL 1.0.1e-fips 11 Feb 2013 How so I force opensips to use the newer version?? Thanks, Tito _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Tue Jul 11 16:54:15 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 11 Jul 2017 16:54:15 -0400 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: Message-ID: I tried both suggestions. Finally I settled for editing the make file. Now I am getting this error Jul 11 20:50:59 cloud-server-06 opensips: DBG:core:load_module: loading module /usr/lib64/opensips/modules/tls_mgm.so Jul 11 20:50:59 cloud-server-06 opensips: ERROR:core:sr_load_module: could not open module : /usr/lib64/opensips/modules/tls_mgm.so: undefined symbol: GENERAL_NAME_free Jul 11 20:50:59 cloud-server-06 opensips: ERROR:core:load_module: failed to load module Jul 11 20:50:59 cloud-server-06 opensips: CRITICAL:core:yyerror: parse error in config file /etc/opensips/opensips.cfg, line 68, column 13-14: failed to load module tls_mgm.so Here is the edited make file # 2 # WARNING: do not run this directly, it should be run by the master Makefile 3 4 include ../../Makefile.defs 5 auto_gen= 6 NAME=tls_mgm.so 7 8 ETC_DIR?=../../etc/ 9 10 tls_configs=$(patsubst $(ETC_DIR)/%, %, $(wildcard $(ETC_DIR)/tls/*) \ 11 $(wildcard $(ETC_DIR)/tls/rootCA/*) $(wildcard $(ETC_DIR)/tls/rootCA/certs/*) \ 12 $(wildcard $(ETC_DIR)/tls/rootCA/private/*) $(wildcard $(ETC_DIR)/tls/user/*)) 13 14 15 ifeq ($(CROSS_COMPILE),) 16 SSL_BUILDER=$(shell \ 17 if pkg-config --exists libssl; then \ 18 echo 'pkg-config libssl'; \ 19 fi) 20 endif 21 22 ifneq ($(SSL_BUILDER),) 23 DEFS += $(shell $(SSL_BUILDER) --cflags) 24 LIBS += -Wl,-rpath /usr/local/ssl/lib/ $(shell $(SSL_BUILDER) —libs) 25 else 26 DEFS += -I$(LOCALBASE)/ssl/include \ 27 -I$(LOCALBASE)/include 28 LIBS += -Wl,-rpath /usr/local/ssl/lib/ $(shell $(SSL_BUILDER) —libs) 29 endif 30 31 include ../../Makefile.modules 32 33 install_module_custom: 34 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls ; \ 35 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA ; \ 36 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA/certs ; \ 37 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA/private ; \ 38 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/user ; \ 39 for FILE in $(tls_configs) ; do \ 40 if [ -f $(ETC_DIR)/$$FILE ]; then \ 41 if [ "$(tls_overwrite_certs)" != "" -o \ 42 ! -f $(cfg_prefix)/$(cfg_dir)/$$FILE ] ; then \ 43 $(INSTALL_TOUCH) $(ETC_DIR)/$$FILE \ 44 $(cfg_prefix)/$(cfg_dir)/$$FILE ; \ 45 $(INSTALL_CFG) $(ETC_DIR)/$$FILE \ 46 $(cfg_prefix)/$(cfg_dir)/$$FILE ; \ 47 fi; \ 48 fi ;\ 49 done ; \ On Tue, Jul 11, 2017 at 3:51 PM, Mundkowsky, Robert wrote: > Why hardcode it, just use LD_LIBRARY_PATH > > > > > > Robert > > > > *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Liviu > Chircu > *Sent:* Tuesday, July 11, 2017 3:46 PM > *To:* users at lists.opensips.org > *Subject:* Re: [OpenSIPS-Users] compile with openssl version > > > > It looks like your distro's libssl still has priority over the custom one. > To avoid both uninstalling libssl and forcing all apps to use the newest > library, I suggest you compile a hardcoded search path into tls_mgm.so. > > Just make a small modification in modules/tls_mgm/Makefile, like in this > example: > > LIBS += -Wl,-rpath /home/liviu/lib $(shell $(SSL_BUILDER) --libs) > > Compile the tls_mgm, and if all goes well, the linker should spot the > custom libssl first: > > [liviu ◄ Y510P opensips (master)]$ ldd modules/tls_mgm/tls_mgm.so > linux-vdso.so.1 => (0x00007ffff040d000) > libssl.so.1.0.0 => /home/liviu/lib/libssl.so.1.0.0 > (0x00007fd9cde0a000) <---- the forced "runtime path" is working! > libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fd9cda21000) > libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > (0x00007fd9cd5dc000) > /lib64/ld-linux-x86-64.so.2 (0x000055a69a1b7000) > libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007fd9cd3d8000) > > Another solution could be: > > echo "/usr/local/lib" > /etc/ld.so.conf.d/libssl.conf; ldconfig > > But note that this will "upgrade" the library for all apps in your system > that require it. > > Liviu Chircu > > OpenSIPS Developer > > http://www.opensips-solutions.com > > On 11.07.2017 21:58, Tito Cumpen wrote: > > Group, > > > > > > I've updated openssl in order to use opensips 2.3 but I am having issues > after compiling and running > > > > > > openssl version -a > > OpenSSL 1.0.2k 26 Jan 2017 > > built on: reproducible build, date unspecified > > platform: linux-x86_64 > > options: bn(64,64) rc4(8x,int) des(idx,cisc,16,int) idea(int) > blowfish(idx) > > compiler: gcc -I. -I.. -I../include -DOPENSSL_THREADS -D_REENTRANT > -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -m64 -DL_ENDIAN -O3 -Wall > -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 > -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM > -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM > -DECP_NISTZ256_ASM > > OPENSSLDIR: "/usr/local/ssl" > > > > > > but when I run opensips I get > > > > ERROR:tls_mgm:mod_init: unable to set the memory allocation functions > > Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: > ERROR:tls_mgm:mod_init: NOTE: check if you are using openssl 1.0.1e-fips, > (or other FIPS version of openssl, as this is known to be broken; if so, > you need to upgrade or downgrade to a different openssl version! > > Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: > ERROR:tls_mgm:mod_init: current version: OpenSSL 1.0.1e-fips 11 Feb 2013 > > > > > > How so I force opensips to use the newer version?? > > > > Thanks, > > Tito > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Jul 11 17:34:44 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 12 Jul 2017 00:34:44 +0300 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: Message-ID: <08b956b4-b4bb-fc90-b45d-1f4731c68d5a@opensips.org> That's a libcrypto symbol - make sure that one is also compiled and installed under /usr/local/ssl/lib Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 11.07.2017 23:54, Tito Cumpen wrote: > I tried both suggestions. Finally I settled for editing the make file. > Now I am getting this error > > > Jul 11 20:50:59 cloud-server-06 opensips: DBG:core:load_module: > loading module /usr/lib64/opensips/modules/tls_mgm.so > > Jul 11 20:50:59 cloud-server-06 opensips: ERROR:core:sr_load_module: > could not open module : > /usr/lib64/opensips/modules/tls_mgm.so: undefined symbol: > GENERAL_NAME_free > > Jul 11 20:50:59 cloud-server-06 opensips: ERROR:core:load_module: > failed to load module > > Jul 11 20:50:59 cloud-server-06 opensips: CRITICAL:core:yyerror: parse > error in config file /etc/opensips/opensips.cfg, line 68, column > 13-14: failed to load module tls_mgm.so > > > > Here is the edited make file > > # > > 2 # WARNING: do not run this directly, it should be run by the > master Makefile > > 3 > > 4 include ../../Makefile.defs > > 5 auto_gen= > > 6 NAME=tls_mgm.so > > 7 > > 8 ETC_DIR?=../../etc/ > > 9 > > 10 tls_configs=$(patsubst $(ETC_DIR)/%, %, $(wildcard $(ETC_DIR)/tls/*) \ > > 11 $(wildcard $(ETC_DIR)/tls/rootCA/*) $(wildcard > $(ETC_DIR)/tls/rootCA/certs/*) \ > > 12 $(wildcard $(ETC_DIR)/tls/rootCA/private/*) > $(wildcard $(ETC_DIR)/tls/user/*)) > > 13 > > 14 > > 15 ifeq ($(CROSS_COMPILE),) > > 16 SSL_BUILDER=$(shell \ > > 17 if pkg-config --exists libssl; then \ > > 18 echo 'pkg-config libssl'; \ > > 19 fi) > > 20 endif > > 21 > > 22 ifneq ($(SSL_BUILDER),) > > 23 DEFS += $(shell $(SSL_BUILDER) --cflags) > > 24 LIBS += -Wl,-rpath /usr/local/ssl/lib/ $(shell > $(SSL_BUILDER) —libs) > > 25 else > > 26 DEFS += -I$(LOCALBASE)/ssl/include \ > > 27 -I$(LOCALBASE)/include > > 28 LIBS += -Wl,-rpath /usr/local/ssl/lib/ $(shell > $(SSL_BUILDER) —libs) > > 29 endif > > 30 > > 31 include ../../Makefile.modules > > 32 > > 33 install_module_custom: > > 34 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls ; \ > > 35 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA ; \ > > 36 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA/certs ; \ > > 37 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA/private ; \ > > 38 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/user ; \ > > 39 for FILE in $(tls_configs) ; do \ > > 40 if [ -f $(ETC_DIR)/$$FILE ]; then \ > > 41 if [ "$(tls_overwrite_certs)" != "" -o \ > > 42 ! -f $(cfg_prefix)/$(cfg_dir)/$$FILE ] ; then \ > > 43 $(INSTALL_TOUCH) $(ETC_DIR)/$$FILE \ > > 44 $(cfg_prefix)/$(cfg_dir)/$$FILE ; \ > > 45 $(INSTALL_CFG) $(ETC_DIR)/$$FILE \ > > 46 $(cfg_prefix)/$(cfg_dir)/$$FILE ; \ > > 47 fi; \ > > 48 fi ;\ > > 49 done ; \ > > > > On Tue, Jul 11, 2017 at 3:51 PM, Mundkowsky, Robert > > wrote: > > Why hardcode it, just use LD_LIBRARY_PATH > > Robert > > *From:*Users [mailto:users-bounces at lists.opensips.org > ] *On Behalf Of *Liviu Chircu > *Sent:* Tuesday, July 11, 2017 3:46 PM > *To:* users at lists.opensips.org > *Subject:* Re: [OpenSIPS-Users] compile with openssl version > > It looks like your distro's libssl still has priority over the > custom one. To avoid both uninstalling libssl and forcing all apps > to use the newest library, I suggest you compile a hardcoded > search path into tls_mgm.so. > > Just make a small modification in modules/tls_mgm/Makefile, like > in this example: > > LIBS += -Wl,-rpath /home/liviu/lib $(shell $(SSL_BUILDER) --libs) > > Compile the tls_mgm, and if all goes well, the linker should spot > the custom libssl first: > > [liviu ◄ Y510P opensips (master)]$ ldd modules/tls_mgm/tls_mgm.so > linux-vdso.so.1 => (0x00007ffff040d000) > libssl.so.1.0.0 => /home/liviu/lib/libssl.so.1.0.0 > (0x00007fd9cde0a000) <---- the forced "runtime path" is working! > libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fd9cda21000) > libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 > (0x00007fd9cd5dc000) > /lib64/ld-linux-x86-64.so.2 (0x000055a69a1b7000) > libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 > (0x00007fd9cd3d8000) > > Another solution could be: > > echo "/usr/local/lib" > /etc/ld.so.conf.d/libssl.conf; ldconfig > > But note that this will "upgrade" the library for all apps in your > system that require it. > > Liviu Chircu > > OpenSIPS Developer > > http://www.opensips-solutions.com > > > On 11.07.2017 21:58, Tito Cumpen wrote: > > Group, > > I've updated openssl in order to use opensips 2.3 but I am > having issues after compiling and running > > openssl version -a > > OpenSSL 1.0.2k 26 Jan 2017 > > built on: reproducible build, date unspecified > > platform: linux-x86_64 > > options: bn(64,64) rc4(8x,int) des(idx,cisc,16,int) idea(int) > blowfish(idx) > > compiler: gcc -I. -I.. -I../include -DOPENSSL_THREADS > -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -m64 > -DL_ENDIAN -O3 -Wall -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT > -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM > -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM > -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM > -DECP_NISTZ256_ASM > > OPENSSLDIR: "/usr/local/ssl" > > but when I run opensips I get > > ERROR:tls_mgm:mod_init: unable to set the memory allocation > functions > > Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: > ERROR:tls_mgm:mod_init: NOTE: check if you are using openssl > 1.0.1e-fips, (or other FIPS version of openssl, as this is > known to be broken; if so, you need to upgrade or downgrade to > a different openssl version! > > Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: > ERROR:tls_mgm:mod_init: current version: OpenSSL 1.0.1e-fips > 11 Feb 2013 > > How so I force opensips to use the newer version?? > > Thanks, > > Tito > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ------------------------------------------------------------------------ > > This e-mail and any files transmitted with it may contain > privileged or confidential information. It is solely for use by > the individual for whom it is intended, even if addressed > incorrectly. If you received this e-mail in error, please notify > the sender; do not disclose, copy, distribute, or take any action > in reliance on the contents of this information; and delete it > from your system. Any other use of this e-mail is prohibited. > > > Thank you for your compliance. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Tue Jul 11 17:38:44 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 11 Jul 2017 17:38:44 -0400 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: <08b956b4-b4bb-fc90-b45d-1f4731c68d5a@opensips.org> References: <08b956b4-b4bb-fc90-b45d-1f4731c68d5a@opensips.org> Message-ID: Liviu, it is check out the following ls -al /usr/local/ssl/lib/ total 5780 drwxr-xr-x 4 root root 4096 Jul 11 18:22 . drwxr-xr-x 9 root root 4096 Jul 11 18:22 .. drwxr-xr-x 2 root root 4096 Apr 24 21:35 engines -rw-r--r-- 1 root root 5122378 Jul 11 18:22 libcrypto.a -rw-r--r-- 1 root root 776104 Jul 11 18:22 libssl.a drwxr-xr-x 2 root root 4096 Apr 24 21:35 pkgconfig is there an extra module I need to enable when compiling openssl? On Tue, Jul 11, 2017 at 5:34 PM, Liviu Chircu wrote: > That's a libcrypto symbol - make sure that one is also compiled and > installed under /usr/local/ssl/lib > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 11.07.2017 23:54, Tito Cumpen wrote: > > I tried both suggestions. Finally I settled for editing the make file. Now > I am getting this error > > > Jul 11 20:50:59 cloud-server-06 opensips: DBG:core:load_module: loading > module /usr/lib64/opensips/modules/tls_mgm.so > > Jul 11 20:50:59 cloud-server-06 opensips: ERROR:core:sr_load_module: could > not open module : > /usr/lib64/opensips/modules/tls_mgm.so: undefined symbol: > GENERAL_NAME_free > > Jul 11 20:50:59 cloud-server-06 opensips: ERROR:core:load_module: failed > to load module > > Jul 11 20:50:59 cloud-server-06 opensips: CRITICAL:core:yyerror: parse > error in config file /etc/opensips/opensips.cfg, line 68, column 13-14: > failed to load module tls_mgm.so > > > > Here is the edited make file > > # > > 2 # WARNING: do not run this directly, it should be run by the master > Makefile > > 3 > > 4 include ../../Makefile.defs > > 5 auto_gen= > > 6 NAME=tls_mgm.so > > 7 > > 8 ETC_DIR?=../../etc/ > > 9 > > 10 tls_configs=$(patsubst $(ETC_DIR)/%, %, $(wildcard $(ETC_DIR)/tls/*) \ > > 11 $(wildcard $(ETC_DIR)/tls/rootCA/*) $(wildcard > $(ETC_DIR)/tls/rootCA/certs/*) \ > > 12 $(wildcard $(ETC_DIR)/tls/rootCA/private/*) > $(wildcard $(ETC_DIR)/tls/user/*)) > > 13 > > 14 > > 15 ifeq ($(CROSS_COMPILE),) > > 16 SSL_BUILDER=$(shell \ > > 17 if pkg-config --exists libssl; then \ > > 18 echo 'pkg-config libssl'; \ > > 19 fi) > > 20 endif > > 21 > > 22 ifneq ($(SSL_BUILDER),) > > 23 DEFS += $(shell $(SSL_BUILDER) --cflags) > > 24 LIBS += -Wl,-rpath /usr/local/ssl/lib/ $(shell > $(SSL_BUILDER) —libs) > > 25 else > > 26 DEFS += -I$(LOCALBASE)/ssl/include \ > > 27 -I$(LOCALBASE)/include > > 28 LIBS += -Wl,-rpath /usr/local/ssl/lib/ $(shell $(SSL_BUILDER) > —libs) > > 29 endif > > 30 > > 31 include ../../Makefile.modules > > 32 > > 33 install_module_custom: > > 34 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls ; \ > > 35 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA ; \ > > 36 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA/certs ; \ > > 37 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/rootCA/private ; \ > > 38 mkdir -p $(cfg_prefix)/$(cfg_dir)/tls/user ; \ > > 39 for FILE in $(tls_configs) ; do \ > > 40 if [ -f $(ETC_DIR)/$$FILE ]; then \ > > 41 if [ "$(tls_overwrite_certs)" != "" -o \ > > 42 ! -f > $(cfg_prefix)/$(cfg_dir)/$$FILE ] ; then \ > > 43 $(INSTALL_TOUCH) $(ETC_DIR)/$$FILE \ > > 44 $(cfg_prefix)/$(cfg_dir)/$$FILE > ; \ > > 45 $(INSTALL_CFG) $(ETC_DIR)/$$FILE \ > > 46 $(cfg_prefix)/$(cfg_dir)/$$FILE > ; \ > > 47 fi; \ > > 48 fi ;\ > > 49 done ; \ > > > On Tue, Jul 11, 2017 at 3:51 PM, Mundkowsky, Robert > wrote: > >> Why hardcode it, just use LD_LIBRARY_PATH >> >> >> >> >> >> Robert >> >> >> >> *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Liviu >> Chircu >> *Sent:* Tuesday, July 11, 2017 3:46 PM >> *To:* users at lists.opensips.org >> *Subject:* Re: [OpenSIPS-Users] compile with openssl version >> >> >> >> It looks like your distro's libssl still has priority over the custom >> one. To avoid both uninstalling libssl and forcing all apps to use the >> newest library, I suggest you compile a hardcoded search path into >> tls_mgm.so. >> >> Just make a small modification in modules/tls_mgm/Makefile, like in this >> example: >> >> LIBS += -Wl,-rpath /home/liviu/lib $(shell $(SSL_BUILDER) --libs) >> >> Compile the tls_mgm, and if all goes well, the linker should spot the >> custom libssl first: >> >> [liviu ◄ Y510P opensips (master)]$ ldd modules/tls_mgm/tls_mgm.so >> linux-vdso.so.1 => (0x00007ffff040d000) >> libssl.so.1.0.0 => /home/liviu/lib/libssl.so.1.0.0 >> (0x00007fd9cde0a000) <---- the forced "runtime path" is working! >> libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007fd9cda21000) >> libcrypto.so.1.0.0 => /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 >> (0x00007fd9cd5dc000) >> /lib64/ld-linux-x86-64.so.2 (0x000055a69a1b7000) >> libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007fd9cd3d8000) >> >> Another solution could be: >> >> echo "/usr/local/lib" > /etc/ld.so.conf.d/libssl.conf; ldconfig >> >> But note that this will "upgrade" the library for all apps in your system >> that require it. >> >> Liviu Chircu >> >> OpenSIPS Developer >> >> http://www.opensips-solutions.com >> >> On 11.07.2017 21:58, Tito Cumpen wrote: >> >> Group, >> >> >> >> >> >> I've updated openssl in order to use opensips 2.3 but I am having issues >> after compiling and running >> >> >> >> >> >> openssl version -a >> >> OpenSSL 1.0.2k 26 Jan 2017 >> >> built on: reproducible build, date unspecified >> >> platform: linux-x86_64 >> >> options: bn(64,64) rc4(8x,int) des(idx,cisc,16,int) idea(int) >> blowfish(idx) >> >> compiler: gcc -I. -I.. -I../include -DOPENSSL_THREADS -D_REENTRANT >> -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -m64 -DL_ENDIAN -O3 -Wall >> -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 >> -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM >> -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM >> -DECP_NISTZ256_ASM >> >> OPENSSLDIR: "/usr/local/ssl" >> >> >> >> >> >> but when I run opensips I get >> >> >> >> ERROR:tls_mgm:mod_init: unable to set the memory allocation functions >> >> Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: >> ERROR:tls_mgm:mod_init: NOTE: check if you are using openssl 1.0.1e-fips, >> (or other FIPS version of openssl, as this is known to be broken; if so, >> you need to upgrade or downgrade to a different openssl version! >> >> Jul 11 18:52:56 cloud-server-06 /sbin/opensips[32421]: >> ERROR:tls_mgm:mod_init: current version: OpenSSL 1.0.1e-fips 11 Feb 2013 >> >> >> >> >> >> How so I force opensips to use the newer version?? >> >> >> >> Thanks, >> >> Tito >> >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> ------------------------------ >> >> This e-mail and any files transmitted with it may contain privileged or >> confidential information. It is solely for use by the individual for whom >> it is intended, even if addressed incorrectly. If you received this e-mail >> in error, please notify the sender; do not disclose, copy, distribute, or >> take any action in reliance on the contents of this information; and delete >> it from your system. Any other use of this e-mail is prohibited. >> >> Thank you for your compliance. >> ------------------------------ >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 12 05:51:08 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 12 Jul 2017 12:51:08 +0300 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: <08b956b4-b4bb-fc90-b45d-1f4731c68d5a@opensips.org> Message-ID: Can you post the output of the following: LD_LIBRARY_PATH=/usr/local/ssl/lib/ ldd modules/tls_mgm/tls_mgm.so Remember, we want to get it to find the new shared libraries, not some statically compiled libraries (aka ".a" files). Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 12.07.2017 00:38, Tito Cumpen wrote: > Liviu, > > > it is check out the following > > ls -al /usr/local/ssl/lib/ > > total 5780 > > drwxr-xr-x 4 root root4096 Jul 11 18:22 . > > drwxr-xr-x 9 root root4096 Jul 11 18:22 .. > > drwxr-xr-x 2 root root4096 Apr 24 21:35 engines > > -rw-r--r-- 1 root root 5122378 Jul 11 18:22 libcrypto.a > > -rw-r--r-- 1 root root776104 Jul 11 18:22 libssl.a > > drwxr-xr-x 2 root root4096 Apr 24 21:35 pkgconfig > > > > is there an extra module I need to enable when compiling openssl? > > > > > > On Tue, Jul 11, 2017 at 5:34 PM, Liviu Chircu > wrote: > > That's a libcrypto symbol - make sure that one is also compiled > and installed under /usr/local/ssl/lib > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From josh at nowlin.tk Wed Jul 12 08:30:35 2017 From: josh at nowlin.tk (Josh S.) Date: Wed, 12 Jul 2017 14:30:35 +0200 Subject: [OpenSIPS-Users] Multiple sites - Registering and NAT'ing issue -- Best options for implementing Message-ID: <4bc009ce-89ce-1c93-f34b-f14e89779070@nowlin.tk> Hi all, We've got two twin sites sitting behind nat in different private networks, by twins I mean they both have the exact same config. UACs can register only once at any site, also they can move between sites and place calls. (just locally though) Each opensips instance registers the local UACs. Opensips version is 2.3 RTTProxy is integrated, and running just fine. UACs are being registered into the twin remote instance using t_replicate("sip:IP:PORT") (although I'm not sure whether this is the best method to achieve this) So, the opensipsctl ul show command shows: AOR:: 100 Contact:: sip:100 at 192.168.1.2:12345;ob AOR:: 101 Contact:: sip:101 at 11.22.33.44:5060;ob and in the other site, same command shows: AOR:: 100 Contact:: sip:100 at 22.33.44.55:5060;ob AOR:: 101 Contact:: sip:101 at 192.168.1.3:54321;ob being 11.22.33.44 and 22.33.44.55 the public IP address of each opensips registar. Each opensips instance is working ok with the stock config, and the UACs on its private internal network can make calls between each other. But of course not to the UACS on the remote site. And that is the issue. So I started to play around with nat_tranversal, nathelper, topology_hiding, and reading all the many info in the docs/modules and what not, and in this list of course. I also tried the b2bua module, to no avail. I reached to an end route where progress is not made-.... I began to think this cannot be that complicated. I refuse to play 'inventing the wheel' again game. I need some little help in finding the right direction to get the two sites routing calls between each other. Any hint would be greatly appreciated, guys. Just for clarifying purposes, a little drawing Private Net Site A Public IP side UAC1 | -----------------------------------------------> OpenSIPS 11.22.33.44 UAC2 | | | | | <-------> RTTProxy Private Net Site B Public IP side UAC3 | -----------------------------------------------> OpenSIPS 22.33.44.55 UAC4 | | | | | <-------> RTTProxy it's required that the UACs can move around to any site, same username (uri), and still be reachable from any other point. from stock config, I made the slight changes to allow for RTTProxy be integrated, and it's working. No NAT was involved, as yet. The two sites share the same domain, and also the same internal IP network in 192.168.1.0/24 space, although that can change if needed. Each site works fine individually. What changes need to be done in the config for the calls be routed between the two sites? Any example around? Thanks a lot. Josh Smit From xaled at web.de Wed Jul 12 08:45:49 2017 From: xaled at web.de (xaled) Date: Wed, 12 Jul 2017 14:45:49 +0200 Subject: [OpenSIPS-Users] Set userpart parameter using $rU Message-ID: <040101d2fb0c$ca830850$5f8918f0$@web.de> Hello, I cannot set the userpart parameter in the RURI using $rU. The needed parameter always becomes the URI and not userpart parameter. rewriteuser("+1234556;rn=+1234567"); or $rU = "+1234556;rn=+1234567"; Produces the same result with the userpart parameter becoming URI parameter: "INVITE sip:+1234567 at test.com:5080;rn=+1234567 SIP/2.0\r Only modifying the whole URI using $ru works as needed: $ru="sip:" + "+1234567;rn=+1234567" + "@" + "test.com"; "INVITE sip:+1234567;rn=+1234567 at test.com SIP/2.0\r Is there a reason for this behavior? Thanks, xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Jul 12 08:54:18 2017 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 12 Jul 2017 12:54:18 +0000 Subject: [OpenSIPS-Users] Set userpart parameter using $rU In-Reply-To: <040101d2fb0c$ca830850$5f8918f0$@web.de> References: <040101d2fb0c$ca830850$5f8918f0$@web.de> Message-ID: Xaled, I ran into this issue very recently myself. Please reference this thread from the mailing list: http://lists.opensips.org/pipermail/users/2017-July/037666.html Thanks, Ben From: Users on behalf of xaled Reply-To: OpenSIPS users mailling list Date: Wednesday, July 12, 2017 at 8:45 AM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Set userpart parameter using $rU Hello, I cannot set the userpart parameter in the RURI using $rU. The needed parameter always becomes the URI and not userpart parameter. rewriteuser("+1234556;rn=+1234567"); or $rU = "+1234556;rn=+1234567"; Produces the same result with the userpart parameter becoming URI parameter: "INVITE sip:+1234567 at test.com:5080;rn=+1234567 SIP/2.0\r Only modifying the whole URI using $ru works as needed: $ru="sip:" + "+1234567;rn=+1234567" + "@" + "test.com"; "INVITE sip:+1234567;rn=+1234567 at test.com SIP/2.0\r Is there a reason for this behavior? Thanks, xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Jul 12 09:10:11 2017 From: xaled at web.de (xaled) Date: Wed, 12 Jul 2017 15:10:11 +0200 Subject: [OpenSIPS-Users] Set userpart parameter using $rU In-Reply-To: References: <040101d2fb0c$ca830850$5f8918f0$@web.de> Message-ID: <042501d2fb10$3183b790$948b26b0$@web.de> Hi Ben, thanks for pointing this out. I’ll try the fix on my 2.3 opensips. Greetings, Xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ben Newlin Sent: Mittwoch, 12. Juli 2017 14:54 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Set userpart parameter using $rU Xaled, I ran into this issue very recently myself. Please reference this thread from the mailing list: http://lists.opensips.org/pipermail/users/2017-July/037666.html Thanks, Ben From: Users > on behalf of xaled > Reply-To: OpenSIPS users mailling list > Date: Wednesday, July 12, 2017 at 8:45 AM To: 'OpenSIPS users mailling list' > Subject: [OpenSIPS-Users] Set userpart parameter using $rU Hello, I cannot set the userpart parameter in the RURI using $rU. The needed parameter always becomes the URI and not userpart parameter. rewriteuser("+1234556;rn=+1234567"); or $rU = "+1234556;rn=+1234567"; Produces the same result with the userpart parameter becoming URI parameter: "INVITE sip:+1234567 at test.com:5080;rn=+1234567 SIP/2.0\r Only modifying the whole URI using $ru works as needed: $ru="sip:" + "+1234567;rn=+1234567" + "@" + "test.com"; "INVITE sip:+1234567;rn=+1234567 at test.com SIP/2.0\r Is there a reason for this behavior? Thanks, xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From ali.raza at timegroup.ae Wed Jul 12 09:26:11 2017 From: ali.raza at timegroup.ae (Ali Raza) Date: Wed, 12 Jul 2017 09:26:11 -0400 (EDT) Subject: [OpenSIPS-Users] Mid-Registrar Absorb 2nd Register(w/ AuthHeader) Request - Only on Reg Renewal. Message-ID: <1963922431.2029514.1499865970353.JavaMail.zimbra@timegroup.ae> Hello Guys, I am new to OpenSIPS. I am currently test mid-registrar module with FreeSwitch and I am facing a issue not sure if its a bug or its me. Let me explain whats happening: I am running OpenSIPS mid-registrar in contact-throttling mode(mode:1) with usrloc mode:0 - because mid-registrar was crashing again and again then I saw the post https://github.com/OpenSIPS/opensips/issues/1094 - so that issue is now gone with usrloc mode:0. When I run opensips my devices(soft phone: zopier and sip phone: fanvil) register perfectly but as soon the registration time for fanvilphone is reaching expiry(outging expiry) opensips passes my registrartion request to freeswitch - Freeswitch sends back 401unauthorised message which is delivered to the phone by opensips. This time phone sends register request with AuthHeader but this request gets absorbed by mid-registrar and reply from UAC with AuthHeader never reach UAS and freeswitch then remove the registrartion from its database assuming the UAC is dead. But when the softphone-Zopier outgoing register expiring somehow Softphone sends the register request with AuthHeader and as OpenSIPs forwards 1 register request to freeswtich - Freeswitch renew its registrartion. So softphone remain registered and works fine. 2nd/Renew Registration from FanvilPhone: 1. UAC==Reg==>OpenSIPS==>FreeSwitch 2. FreeSwitch==401==>OpenSIPs==401==>UAC 3. UAC==Reg w/Auth-Header==>OpenSIPs(Absorbe by mid-registrar: Returncode: 2) 4. OpenSIPs==Reply 200==>UAC (so actually phone thinks its registered) 2nd/Renew Registration from SoftPhone Zopier: 1. SoftPhone==Reg w/Auth-Header==>OpenSIPs== Reg w/Auth-Header==>FreeSwitch 2. FreeSwitch==Reply 200==>OpenSIPs==Reply 200==>SoftPhone-Zopier. MY OPENSIPS CONFIG -- USED FOR MID-REGISTRAR: #### USeR LOCation module loadmodule "usrloc.so" modparam("usrloc", "nat_bflag", "NAT") modparam("usrloc", "db_mode", 0) modparam("usrloc", "use_domain", 1) #### MID-REGISTRAR module loadmodule "mid_registrar.so" modparam("mid_registrar", "mode", 1) /* 0 = mirror / 1 = ct / 2 = AoR */ modparam("mid_registrar", "outgoing_expires", 180) ##Set to low for testing purpose. modparam("mid_registrar", "insertion_mode", 0) /* 0 = contact; 1 = path */ if ( !(is_method("REGISTER")) ) { if (check_source_address("10")) { xlog("looking up $ru!\n"); if (!mid_registrar_lookup("location")) { t_reply("404", "Not Found"); exit; } t_relay(); exit; } } if (is_method("REGISTER")) { #mid_registrar_save("location"); xlog("BEFORE IT PASS TO MID-REG SAVE!"); mid_registrar_save("","m"); switch ($retcode) { case 1: xlog("forwarding REGISTER to main registrar ($$ci=$ci) - $fd\n"); #Call script to set $ru - Testing! #perl_exec("dest_host","$fd"); $ru = "sip:dispatcher\@10.10.7.206:5070"; xlog("NEW HOST VALUE: $ru"); t_relay(); break; case 2: xlog("absorbing REGISTER! ($$ci=$ci)\n"); break; default: xlog("failed to save registration! ($$ci=$ci)\n"); } exit; } LET ME KNOW IF YOU NEED ANYTHING ELSE. THANKS! Regards Ali Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 12 10:26:10 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 12 Jul 2017 17:26:10 +0300 Subject: [OpenSIPS-Users] Mid-Registrar Absorb 2nd Register(w/ AuthHeader) Request - Only on Reg Renewal. In-Reply-To: <1963922431.2029514.1499865970353.JavaMail.zimbra@timegroup.ae> References: <1963922431.2029514.1499865970353.JavaMail.zimbra@timegroup.ae> Message-ID: Thanks for the nice report, Ali - I'm already testing a fix for this, and will keep you posted! Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 12.07.2017 16:26, Ali Raza wrote: > Hello Guys, > > I am new to OpenSIPS. I am currently test mid-registrar module with > FreeSwitch and I am facing a issue not sure if its a bug or its me. > > Let me explain whats happening: > > I am running OpenSIPS mid-registrar in contact-throttling > mode(mode:1) with usrloc mode:0 - because mid-registrar was crashing > again and again then I saw the > post https://github.com/OpenSIPS/opensips/issues/1094 - so that issue > is now gone with usrloc mode:0. > > When I run opensips my devices(soft phone: zopier and sip phone: > fanvil) register perfectly but as soon the registration time for > fanvilphone is reaching expiry(outging expiry) opensips passes my > registrartion request to freeswitch - Freeswitch sends back > 401unauthorised message which is delivered to the phone by opensips. > This time phone sends register request with AuthHeader but this > request gets absorbed by mid-registrar and reply from UAC with > AuthHeader never reach UAS and freeswitch then remove the > registrartion from its database assuming the UAC is dead. But when the > softphone-Zopier outgoing register expiring somehow Softphone sends > the register request with AuthHeader and as OpenSIPs forwards 1 > register request to freeswtich - Freeswitch renew its registrartion. > So softphone remain registered and works fine. > > 2nd/Renew Registration from FanvilPhone: > > 1. UAC==Reg==>OpenSIPS==>FreeSwitch > > 2. FreeSwitch==401==>OpenSIPs==401==>UAC > > 3. UAC==Reg w/Auth-Header==>OpenSIPs(Absorbe by mid-registrar: > Returncode: 2) > > 4. OpenSIPs==Reply 200==>UAC (so actually phone thinks its registered) > > 2nd/Renew Registration from SoftPhone Zopier: > > 1. SoftPhone==Reg w/Auth-Header==>OpenSIPs==Reg w/Auth-Header==>FreeSwitch > > 2. FreeSwitch==Reply 200==>OpenSIPs==Reply 200==>SoftPhone-Zopier. > > > MY OPENSIPS CONFIG -- USED FOR MID-REGISTRAR: > > #### USeR LOCation module > loadmodule "usrloc.so" > modparam("usrloc", "nat_bflag", "NAT") > modparam("usrloc", "db_mode", 0) > modparam("usrloc", "use_domain", 1) > > #### MID-REGISTRAR module > loadmodule "mid_registrar.so" > modparam("mid_registrar", "mode", 1) /* 0 = mirror / 1 = ct / 2 = AoR */ > modparam("mid_registrar", "outgoing_expires", 180) ##Set to low for > testing purpose. > modparam("mid_registrar", "insertion_mode", 0) /* 0 = contact; 1 = path */ > > if ( !(is_method("REGISTER")) ) { > if (check_source_address("10")) { > xlog("looking up $ru!\n"); > if (!mid_registrar_lookup("location")) { > t_reply("404", "Not Found"); > exit; > } > > t_relay(); > exit; > } > } > > if (is_method("REGISTER")) > { > #mid_registrar_save("location"); > xlog("BEFORE IT PASS TO MID-REG SAVE!"); > mid_registrar_save("","m"); > switch ($retcode) { > case 1: > xlog("forwarding REGISTER to main registrar ($$ci=$ci) - > $fd\n"); > #Call script to set $ru - Testing! > #perl_exec("dest_host","$fd"); > $ru = "sip:dispatcher\@10.10.7.206:5070"; > xlog("NEW HOST VALUE: $ru"); > t_relay(); > break; > case 2: > xlog("absorbing REGISTER! ($$ci=$ci)\n"); > break; > default: > xlog("failed to save registration! ($$ci=$ci)\n"); > } > exit; > } > > > LET ME KNOW IF YOU NEED ANYTHING ELSE. > > THANKS! > > Regards > Ali Raza > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Wed Jul 12 19:48:11 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Wed, 12 Jul 2017 19:48:11 -0400 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: <08b956b4-b4bb-fc90-b45d-1f4731c68d5a@opensips.org> Message-ID: Liviu, Here is the output: linux-vdso.so.1 => (0x00007ffee9d89000) libdl.so.2 => /lib64/libdl.so.2 (0x00007f096f341000) libresolv.so.2 => /lib64/libresolv.so.2 (0x00007f096f121000) libc.so.6 => /lib64/libc.so.6 (0x00007f096ed59000) /lib64/ld-linux-x86-64.so.2 (0x00007f096f791000) On Wed, Jul 12, 2017 at 5:51 AM, Liviu Chircu wrote: > Can you post the output of the following: > > LD_LIBRARY_PATH=/usr/local/ssl/lib/ ldd modules/tls_mgm/tls_mgm.so > Remember, we want to get it to find the new shared libraries, not some > statically compiled libraries (aka ".a" files). > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 12.07.2017 00:38, Tito Cumpen wrote: > > Liviu, > > > it is check out the following > > ls -al /usr/local/ssl/lib/ > > total 5780 > > drwxr-xr-x 4 root root 4096 Jul 11 18:22 . > > drwxr-xr-x 9 root root 4096 Jul 11 18:22 .. > > drwxr-xr-x 2 root root 4096 Apr 24 21:35 engines > > -rw-r--r-- 1 root root 5122378 Jul 11 18:22 libcrypto.a > > -rw-r--r-- 1 root root 776104 Jul 11 18:22 libssl.a > > drwxr-xr-x 2 root root 4096 Apr 24 21:35 pkgconfig > > > > is there an extra module I need to enable when compiling openssl? > > > > > > On Tue, Jul 11, 2017 at 5:34 PM, Liviu Chircu wrote: > >> That's a libcrypto symbol - make sure that one is also compiled and >> installed under /usr/local/ssl/lib >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Jul 13 05:27:43 2017 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 13 Jul 2017 12:27:43 +0300 Subject: [OpenSIPS-Users] compile with openssl version In-Reply-To: References: <08b956b4-b4bb-fc90-b45d-1f4731c68d5a@opensips.org> Message-ID: <16c8aca8-0a9b-639d-8d77-bfbcb4ae26ba@opensips.org> That's not good at all. Both libssl and libcrypto should be in there, this explains the startup errors - it's not linked against those libraries at all now! If you still want to proceed with the rpath solution, please compile tls_mgm like so: "NICER=0 make modules module=tls_mgm", and post the output, so we know how to fix the make environment. OTOH, we can follow Robert's suggestion, revert all Makefile changes, recompile back to the default tls_mgm and just do: export LD_LIBRARY_PATH=/usr/local/ssl/lib ldd modules/tls_mgm/tls_mgm.so If the above works, you can add a similar logic to your OpenSIPS startup script. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 13.07.2017 02:48, Tito Cumpen wrote: > Liviu, > > Here is the output: > > linux-vdso.so.1 =>(0x00007ffee9d89000) > > libdl.so.2 => /lib64/libdl.so.2 (0x00007f096f341000) > > libresolv.so.2 => /lib64/libresolv.so.2 (0x00007f096f121000) > > libc.so.6 => /lib64/libc.so.6 (0x00007f096ed59000) > > /lib64/ld-linux-x86-64.so.2 (0x00007f096f791000) > > > > > On Wed, Jul 12, 2017 at 5:51 AM, Liviu Chircu > wrote: > > Can you post the output of the following: > > LD_LIBRARY_PATH=/usr/local/ssl/lib/ ldd modules/tls_mgm/tls_mgm.so > > Remember, we want to get it to find the new shared libraries, not > some statically compiled libraries (aka ".a" files). > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 12.07.2017 00:38, Tito Cumpen wrote: >> Liviu, >> >> >> it is check out the following >> >> ls -al /usr/local/ssl/lib/ >> >> total 5780 >> >> drwxr-xr-x 4 root root4096 Jul 11 18:22 . >> >> drwxr-xr-x 9 root root4096 Jul 11 18:22 .. >> >> drwxr-xr-x 2 root root4096 Apr 24 21:35 engines >> >> -rw-r--r-- 1 root root 5122378 Jul 11 18:22 libcrypto.a >> >> -rw-r--r-- 1 root root776104 Jul 11 18:22 libssl.a >> >> drwxr-xr-x 2 root root4096 Apr 24 21:35 pkgconfig >> >> >> >> is there an extra module I need to enable when compiling openssl? >> >> >> >> >> >> On Tue, Jul 11, 2017 at 5:34 PM, Liviu Chircu > > wrote: >> >> That's a libcrypto symbol - make sure that one is also >> compiled and installed under /usr/local/ssl/lib >> >> Liviu Chircu >> OpenSIPS 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 > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From samusenko at msm.ru Thu Jul 13 07:24:35 2017 From: samusenko at msm.ru (=?utf-8?B?0KHQsNC80YPRgdC10L3QutC+INCQ0L3QtNGA0LXQuQ==?=) Date: Thu, 13 Jul 2017 14:24:35 +0300 Subject: [OpenSIPS-Users] Cluesterer and ul_add permanent In-Reply-To: <9ea53d0a-8112-ae55-b6dc-0eafdf87270f@opensips.org> References: <4647681.e5x6FOWdfS@samusenko> <2068901.zWUMlDRjOt@samusenko> <9ea53d0a-8112-ae55-b6dc-0eafdf87270f@opensips.org> Message-ID: <2390515.VIuJF9ngEW@samusenko> Yes, all right! :) MI process crash. On понедельник, 10 июля 2017 г. 19:21:52 MSK Bogdan-Andrei Iancu wrote: > Hi, > > Ok, so you configured a two nodes cluster (using clusterer) and > configured the USRLOC replication via the clusterer support. > > And when you add a static contact via "ul_add", you get an error with > 2.2.4 and possible a crash on trunk ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/07/2017 06:18 PM, Самусенко Андрей wrote: > > I try to setup HA OpenSIPS 2.2.4 > > Using modules clusterer and usrloc. > > > > Master management IP 192.168.1.2 > > Slave management IP 192.168.1.3 > > > > SIP REGISTER replication OK. > > REGISTER -> master > > master -> slave > > > > Error with MI: > > on master run opensipsctl ul add ............ introduce a > > permanent usrloc entry > > or XMLRPC ul_add('location', ...) > > > > Brrr, today no warning:usrloc... but on master there is > > DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed > > by the remote peer! > > CRITICAL:core:receive_fd: EOF on 11 > > DBG:core:handle_worker: dead child 2, pid 7562 (shutting down?) > > > > On четверг, 6 июля 2017 г. 16:28:53 MSK Bogdan-Andrei Iancu wrote: > >> Hi, > >> > >> Could you be more specific on what you are trying to do ? > >> > >> Regards, > >> > >> Bogdan-Andrei Iancu > >> > >> OpenSIPS Founder and Developer > >> http://www.opensips-solutions.com > >> > >> OpenSIPS Bootcamp 2017, Houston, US > >> > >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > >> > >> On 07/05/2017 07:03 PM, Andrey wrote: > >>> OpenSIPS 2.2.4 > >>> Subject does not work. > >>> Slave: WARNING:usrloc:receive_binary_packet: received bin packet from > >>> unknown source: 192.168.1.2:4414 > >>> and MI process shutdown. > >>> > >>>> select url from clusterer: > >>> bin:192.168.1.2:8026 > >>> bin:192.168.1.3:8026 > >>> > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users at lists.opensips.org > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From liviu at opensips.org Thu Jul 13 10:19:56 2017 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 13 Jul 2017 17:19:56 +0300 Subject: [OpenSIPS-Users] [BLOG] Hunting Down Complex OpenSIPS Bugs in Production Environments Message-ID: <836f957d-6f5f-31e0-eb76-10e4ca38c132@opensips.org> Hi all, For those of you who are interested in how we recently not only troubleshooted a bunch of nasty bugs in the OpenSIPS TCP engine, but also included a helper API to speed up the resolution of similar problems in the future, here is a blog post nicely summing everything up: https://blog.opensips.org/2017/07/13/hunting-down-complex-opensips-bugs-in-production-environments/ Enjoy! -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com From bogdan at opensips.org Mon Jul 17 07:48:35 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 17 Jul 2017 14:48:35 +0300 Subject: [OpenSIPS-Users] Cluesterer and ul_add permanent In-Reply-To: <2390515.VIuJF9ngEW@samusenko> References: <4647681.e5x6FOWdfS@samusenko> <2068901.zWUMlDRjOt@samusenko> <9ea53d0a-8112-ae55-b6dc-0eafdf87270f@opensips.org> <2390515.VIuJF9ngEW@samusenko> Message-ID: <08c005a9-17e9-4905-322f-fa99e410649a@opensips.org> Hi Andrey, Thanks for the report - found the problem and fixed it trunk, 2.3 and 2.2 - please update from GIT and give it a new try. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/13/2017 02:24 PM, Самусенко Андрей wrote: > Yes, all right! :) > MI process crash. > > On понедельник, 10 июля 2017 г. 19:21:52 MSK Bogdan-Andrei Iancu wrote: >> Hi, >> >> Ok, so you configured a two nodes cluster (using clusterer) and >> configured the USRLOC replication via the clusterer support. >> >> And when you add a static contact via "ul_add", you get an error with >> 2.2.4 and possible a crash on trunk ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/07/2017 06:18 PM, Самусенко Андрей wrote: >>> I try to setup HA OpenSIPS 2.2.4 >>> Using modules clusterer and usrloc. >>> >>> Master management IP 192.168.1.2 >>> Slave management IP 192.168.1.3 >>> >>> SIP REGISTER replication OK. >>> REGISTER -> master >>> master -> slave >>> >>> Error with MI: >>> on master run opensipsctl ul add ............ introduce a >>> permanent usrloc entry >>> or XMLRPC ul_add('location', ...) >>> >>> Brrr, today no warning:usrloc... but on master there is >>> DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed >>> by the remote peer! >>> CRITICAL:core:receive_fd: EOF on 11 >>> DBG:core:handle_worker: dead child 2, pid 7562 (shutting down?) >>> >>> On четверг, 6 июля 2017 г. 16:28:53 MSK Bogdan-Andrei Iancu wrote: >>>> Hi, >>>> >>>> Could you be more specific on what you are trying to do ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> On 07/05/2017 07:03 PM, Andrey wrote: >>>>> OpenSIPS 2.2.4 >>>>> Subject does not work. >>>>> Slave: WARNING:usrloc:receive_binary_packet: received bin packet from >>>>> unknown source: 192.168.1.2:4414 >>>>> and MI process shutdown. >>>>> >>>>>> select url from clusterer: >>>>> bin:192.168.1.2:8026 >>>>> bin:192.168.1.3:8026 >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From me at nevian.org Mon Jul 17 11:06:26 2017 From: me at nevian.org (Serge S. Yuriev) Date: Mon, 17 Jul 2017 18:06:26 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms Message-ID: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Hi, OpenSIPS git revision: 02da97c96 Just after start without any load. Log full of Jul 17 17:45:43 rossiten /usr/local/sbin/opensips[11099]: WARNING:core:utimer_ticker: utimer task already scheduled for 100 ms (now 290 ms), it may overlap.. Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: WARNING:core:utimer_ticker: utimer task already scheduled for 100 ms (now 390 ms), it may overlap.. Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: WARNING:core:utimer_ticker: utimer task already scheduled for 100 ms (now 490 ms), it may overlap.. Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: WARNING:core:utimer_ticker: utimer task already scheduled for 100 ms (now 590 ms), it may overlap.. Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11107]: WARNING:core:handle_timer_job: utimer job has a 520000 us delay in execution Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: WARNING:core:handle_timer_job: timer job has a 20000 us delay in execution Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: WARNING:core:handle_timer_job: timer job has a 20000 us delay in execution Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 100000 us delay in execution Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 100000 us delay in execution Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 100000 us delay in execution Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 100000 us delay in execution Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 100000 us delay in execution Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 90000 us delay in execution Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: WARNING:core:handle_timer_job: timer job has a 90000 us delay in execution -- Serge S. Yuriev Lead VoIP engineer From lamiriahmedamin at gmail.com Mon Jul 17 09:24:24 2017 From: lamiriahmedamin at gmail.com (ahmeddd) Date: Mon, 17 Jul 2017 06:24:24 -0700 (MST) Subject: [OpenSIPS-Users] SIP Trunking In-Reply-To: <828940906.1508111250790836724.JavaMail.root@zimbra1.crocker.com> References: <828940906.1508111250790836724.JavaMail.root@zimbra1.crocker.com> Message-ID: <1500297864682-7608019.post@n2.nabble.com> Hallo! Can any one help me in configuring SIP trunk in a local notwork ! I have two IP phones and I have already installed opensips in my VM, and I want to route calls between those two IP phones. thnx -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-Trunking-tp3480673p7608019.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From bogdan at opensips.org Mon Jul 17 11:38:07 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 17 Jul 2017 18:38:07 +0300 Subject: [OpenSIPS-Users] Change Options method into invite In-Reply-To: References: Message-ID: Hi Mike, Such conversion is not possible with OpenSIPS. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/11/2017 08:54 PM, mike Little wrote: > We would like to know if it is possible to have anther sip proxy send > a sip options message to opensips and have opensips change the sip > options message into a sip invite to send to a sip server? > > The reason we are looking to use the invite versus sip options is to > have the sip server run through some dialplan code to verify it is > available to process calls. > > TIA > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Jul 17 11:47:34 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 17 Jul 2017 18:47:34 +0300 Subject: [OpenSIPS-Users] Multiple sites - Registering and NAT'ing issue -- Best options for implementing In-Reply-To: <4bc009ce-89ce-1c93-f34b-f14e89779070@nowlin.tk> References: <4bc009ce-89ce-1c93-f34b-f14e89779070@nowlin.tk> Message-ID: <89918d9c-6e91-bd21-0f62-9a507b0bccca@opensips.org> Hi Josh, Things are not simple here - basically you have to configure your platforms to communicate with the public internet (as the two sites can talk only via the public internet). SO, you have to configure OpenSIPS and RTPproxy to do SIP and RTP bridging between local private network and the public network. Again, I say this considering that there is no way for the 2 sites to exchange traffic directly. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/12/2017 03:30 PM, Josh S. via Users wrote: > Hi all, > > > We've got two twin sites sitting behind nat in different private > networks, by twins I mean they both have the exact same config. > UACs can register only once at any site, also they can move between > sites and place calls. (just locally though) > Each opensips instance registers the local UACs. Opensips version is 2.3 > RTTProxy is integrated, and running just fine. > > UACs are being registered into the twin remote instance using > t_replicate("sip:IP:PORT") (although I'm not sure whether this is > the best method to achieve this) > > So, the opensipsctl ul show command shows: > AOR:: 100 > Contact:: sip:100 at 192.168.1.2:12345;ob > AOR:: 101 > Contact:: sip:101 at 11.22.33.44:5060;ob > > and in the other site, same command shows: > > AOR:: 100 > Contact:: sip:100 at 22.33.44.55:5060;ob > AOR:: 101 > Contact:: sip:101 at 192.168.1.3:54321;ob > > being 11.22.33.44 and 22.33.44.55 the public IP address of each > opensips registar. > > Each opensips instance is working ok with the stock config, and the > UACs on its private internal network can make calls between each other. > But of course not to the UACS on the remote site. And that is the issue. > > So I started to play around with nat_tranversal, nathelper, > topology_hiding, and reading all the many info in the docs/modules and > what not, and in this list of course. > I also tried the b2bua module, to no avail. > > I reached to an end route where progress is not made-.... I began to > think this cannot be that complicated. I refuse to play 'inventing the > wheel' again game. > > I need some little help in finding the right direction to get the two > sites routing calls between each other. > Any hint would be greatly appreciated, guys. > > > Just for clarifying purposes, a little drawing > > > > Private Net Site A > Public IP side > > UAC1 | > -----------------------------------------------> OpenSIPS > 11.22.33.44 > UAC2 | | > | > | > | <-------> RTTProxy > > > > > > Private Net Site B > Public IP side > > UAC3 | > -----------------------------------------------> > OpenSIPS 22.33.44.55 > UAC4 | | > | > | > | <-------> RTTProxy > > > it's required that the UACs can move around to any site, same username > (uri), and still be reachable from any other point. > > from stock config, I made the slight changes to allow for RTTProxy be > integrated, and it's working. No NAT was involved, as yet. > The two sites share the same domain, and also the same internal IP > network in 192.168.1.0/24 space, > although that can change if needed. > Each site works fine individually. > > What changes need to be done in the config for the calls be routed > between the two sites? > Any example around? > > Thanks a lot. > > Josh Smit > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Jul 17 11:49:40 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 17 Jul 2017 18:49:40 +0300 Subject: [OpenSIPS-Users] SIP Trunking In-Reply-To: <1500297864682-7608019.post@n2.nabble.com> References: <828940906.1508111250790836724.JavaMail.root@zimbra1.crocker.com> <1500297864682-7608019.post@n2.nabble.com> Message-ID: <45e11ab7-9100-3380-508e-92e062d0e4cf@opensips.org> Hi, The OpenSIPS default script (or any generated via "make menuconfig" or "osipsconfig" for residential scenario) will do the trick for you - allowing multiple phones to register and call one each other. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/17/2017 04:24 PM, ahmeddd wrote: > Hallo! > > Can any one help me in configuring SIP trunk in a local notwork ! I have two > IP phones and I have already installed opensips in my VM, and I want to > route calls between those two IP phones. > > thnx > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-Trunking-tp3480673p7608019.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Jul 17 11:51:23 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 17 Jul 2017 18:51:23 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms In-Reply-To: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> References: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Message-ID: Hi Serge, Do you get all these during OpenSIP starting sequence (just after starting it) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/17/2017 06:06 PM, Serge S. Yuriev wrote: > Hi, > > OpenSIPS git revision: 02da97c96 > > Just after start without any load. > Log full of > Jul 17 17:45:43 rossiten /usr/local/sbin/opensips[11099]: > WARNING:core:utimer_ticker: utimer task already scheduled > for 100 ms (now 290 ms), it may overlap.. > Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: > WARNING:core:utimer_ticker: utimer task already scheduled > for 100 ms (now 390 ms), it may overlap.. > Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: > WARNING:core:utimer_ticker: utimer task already scheduled > for 100 ms (now 490 ms), it may overlap.. > Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: > WARNING:core:utimer_ticker: utimer task already scheduled > for 100 ms (now 590 ms), it may overlap.. > Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11107]: > WARNING:core:handle_timer_job: utimer job has a 520000 us > delay in execution > Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: > WARNING:core:handle_timer_job: timer job has a 20000 us > delay in execution > Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: > WARNING:core:handle_timer_job: timer job has a 20000 > us delay in execution > Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a 100000 us > delay in execution > Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a 100000 > us delay in execution > Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a > 100000 us delay in execution > Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a 100000 us > delay in execution > Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a 100000 > us delay in execution > Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a 90000 us > delay in execution > Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: > WARNING:core:handle_timer_job: timer job has a 90000 > us delay in execution > From me at nevian.org Tue Jul 18 10:24:27 2017 From: me at nevian.org (Serge S. Yuriev) Date: Tue, 18 Jul 2017 17:24:27 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms In-Reply-To: References: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Message-ID: Hi, Yes just after starting. And such lines repeated very often (each 3-5 mins). This machine only captures very small amount of HEP traffic. I first installed this somewhere in May and there is no such problem IIRC. I'll try to revert and report back. On 17/07/17 18:51, Bogdan-Andrei Iancu wrote: > Hi Serge, > > Do you get all these during OpenSIP starting sequence (just after > starting it) ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/17/2017 06:06 PM, Serge S. Yuriev wrote: >> Hi, >> >> OpenSIPS git revision: 02da97c96 >> >> Just after start without any load. >> Log full of >> Jul 17 17:45:43 rossiten /usr/local/sbin/opensips[11099]: >> WARNING:core:utimer_ticker: utimer task already scheduled >> for 100 ms (now 290 ms), it may overlap.. >> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >> WARNING:core:utimer_ticker: utimer task already scheduled >> for 100 ms (now 390 ms), it may overlap.. >> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >> WARNING:core:utimer_ticker: utimer task already scheduled >> for 100 ms (now 490 ms), it may overlap.. >> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >> WARNING:core:utimer_ticker: utimer task already scheduled >> for 100 ms (now 590 ms), it may overlap.. >> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11107]: >> WARNING:core:handle_timer_job: utimer job has a 520000 us >> delay in execution >> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >> WARNING:core:handle_timer_job: timer job has a 20000 us >> delay in execution >> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >> WARNING:core:handle_timer_job: timer job has a 20000 >> us delay in execution >> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a 100000 us >> delay in execution >> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a 100000 >> us delay in execution >> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a >> 100000 us delay in execution >> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a 100000 us >> delay in execution >> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a 100000 >> us delay in execution >> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a 90000 us >> delay in execution >> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >> WARNING:core:handle_timer_job: timer job has a 90000 >> us delay in execution >> > -- Serge S. Yuriev Lead VoIP engineer From bogdan at opensips.org Tue Jul 18 12:04:44 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 18 Jul 2017 19:04:44 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms In-Reply-To: References: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Message-ID: Serge, what OpenSIPS version you have (opensips -v) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/18/2017 05:24 PM, Serge S. Yuriev wrote: > Hi, > > Yes just after starting. > And such lines repeated very often (each 3-5 mins). > This machine only captures very small amount of HEP traffic. > I first installed this somewhere in May and there is no such problem > IIRC. > I'll try to revert and report back. > > On 17/07/17 18:51, Bogdan-Andrei Iancu wrote: >> Hi Serge, >> >> Do you get all these during OpenSIP starting sequence (just after >> starting it) ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/17/2017 06:06 PM, Serge S. Yuriev wrote: >>> Hi, >>> >>> OpenSIPS git revision: 02da97c96 >>> >>> Just after start without any load. >>> Log full of >>> Jul 17 17:45:43 rossiten /usr/local/sbin/opensips[11099]: >>> WARNING:core:utimer_ticker: utimer task already >>> scheduled for 100 ms (now 290 ms), it may overlap.. >>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>> WARNING:core:utimer_ticker: utimer task already >>> scheduled for 100 ms (now 390 ms), it may overlap.. >>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>> WARNING:core:utimer_ticker: utimer task already >>> scheduled for 100 ms (now 490 ms), it may overlap.. >>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>> WARNING:core:utimer_ticker: utimer task already >>> scheduled for 100 ms (now 590 ms), it may overlap.. >>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11107]: >>> WARNING:core:handle_timer_job: utimer job has a 520000 >>> us delay in execution >>> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >>> WARNING:core:handle_timer_job: timer job has a 20000 us >>> delay in execution >>> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >>> WARNING:core:handle_timer_job: timer job has a 20000 >>> us delay in execution >>> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a 100000 us >>> delay in execution >>> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a >>> 100000 us delay in execution >>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a >>> 100000 us delay in execution >>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a 100000 us >>> delay in execution >>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a >>> 100000 us delay in execution >>> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a 90000 us >>> delay in execution >>> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >>> WARNING:core:handle_timer_job: timer job has a 90000 >>> us delay in execution >>> >> > From me at nevian.org Tue Jul 18 12:09:59 2017 From: me at nevian.org (Serge S. Yuriev) Date: Tue, 18 Jul 2017 19:09:59 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms In-Reply-To: References: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Message-ID: <49df5075-45cb-9545-e9b2-b919c9e53ced@nevian.org> version: opensips 2.4.0-dev (x86_64/linux) flags: STATS: On, SHM_EXTRA_STATS, DISABLE_NAGLE, SHM_MMAP, PKG_MALLOC, QM_MALLOC, FAST_LOCK-FUTEX-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 02da97c96 main.c compiled on 13:48:39 Jul 6 2017 with gcc 6.3.0 On 18/07/17 19:04, Bogdan-Andrei Iancu wrote: > opensips -v -- Serge S. Yuriev Lead VoIP engineer From tito at xsvoce.com Tue Jul 18 17:42:46 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 18 Jul 2017 17:42:46 -0400 Subject: [OpenSIPS-Users] exec environmental var Message-ID: Group, I am trying to migrate to the latest opensips 2.3 but I am having issues when setting this in my script. $avp(env) = ""; xlog(" $var(input) being executed\n"); exec("php /etc/opensips/authenticate.php $var(input2)", "", "$var(outinvite)", "$var(err)", "$avp(env)"); I get this error in my log. Jul 18 21:34:30 cloud-server-09 /sbin/opensips[31402]: DBG:core:__search_avp_map: looking for [env] avp - found 4 Jul 18 21:34:30 cloud-server-09 /sbin/opensips[31402]: ERROR:exec:exec_fixup: env var must be a single variable Any idea why this may be ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Tue Jul 18 17:51:30 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 18 Jul 2017 17:51:30 -0400 Subject: [OpenSIPS-Users] support for cloudamqp In-Reply-To: <5bf131f8-337a-5ef3-b9f7-d8c4a417da33@opensips.org> References: <5bf131f8-337a-5ef3-b9f7-d8c4a417da33@opensips.org> Message-ID: Hey Razvan, Can you share the syntax for 2.3 again ? Although it looks like the documentation claims: 2.3. What is the vhost used by the AMQP server? Currently, the only vhost supported is '/'. Thanks, Tito On Mon, Apr 24, 2017 at 10:09 AM, Răzvan Crainea wrote: > Hi, Tito! > > Here's[1] the snippet I used for my tests. > > [1] https://pastebin.com/rRZhDXTX > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 04/24/2017 10:35 AM, Tito Cumpen wrote: > > Razvan, > > How do specify the virtualhost in 2.3? can you send me a sample of the > syntax ? > > On Apr 24, 2017 3:30 AM, "Răzvan Crainea" wrote: > >> Hi, Tito! >> >> You are right, this is not possible with the event_rabbitmq module, >> because there is no way to specify the virtual host in the url. >> But you can use the latest rabbitmq module in OpenSIPS 2.3. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 04/21/2017 09:53 PM, Tito Cumpen wrote: >> >> Hey Razvan, >> >> I tried the following on Opensips: >> >> subscribe_event("E_ACC_EVENT","rabbitmq:vhostuser:pw at host.cl >> oudamqp.com/vhostuser/queuename"); >> >> their URL string is in this format : >> >> amqp://vhostusert:pw at host.cloudamqp.com/vhostuser >> >> >> When Opensips tries to connect to this queue it sends >> >> / as the argument for openvhost >> >> Cloudamqp replies with >> >> >> NOT_ALLOWED - access to vhost '/' refused for user 'vhostuser' >> >> >> Is my opensips syntax incorrect ? or is this a bug ? >> >> Also I am not sure what you mean by immediate. >> >> Thanks, >> >> Tito >> >> >> >> On Fri, Apr 21, 2017 at 5:56 AM, Răzvan Crainea >> wrote: >> >>> Hi, Tito! >>> >>> I've just made a free cloudamqp account for testing and used the new >>> rabbitmq module to send a message in the queue. The message was not >>> published initially due to the fact that I was using the "immediate" flag, >>> (I was receiving NOT_IMPLEMENTED, probably because the free account lacks >>> some features), but after I removed the setting the message was >>> successfully delivered in the queue. >>> I didn't test with the event_rabbitmq module though, but I imagine it >>> will work too, since both modules (event_rabbitmq and rabbitmq) use the >>> same AMQP library. >>> Can you detail a bit how you tested so we can find where the problem is >>> on your setup? >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 04/20/2017 09:18 PM, Tito Cumpen wrote: >>> >>> Hello, >>> >>> I was wondering if there was any intention to update the amqp support to >>> allow opensips to post events to cloudamqp(https://www.cloudamqp.com). >>> When I try to connect cloudamqp it fails because it is outdated. >>> >>> Thanks, >>> Tito >>> >>> >>> _______________________________________________ >>> 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 > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 19 04:15:32 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 19 Jul 2017 11:15:32 +0300 Subject: [OpenSIPS-Users] exec environmental var In-Reply-To: References: Message-ID: Hi Tito, This seems to be caused by some misused startup optimization logic. Below is a scripting trick that should solve your problem until we take care of providing the official solution. Change this: ..., "$avp(env)"); into this: ..., "$avp(env) "); Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 19.07.2017 00:42, Tito Cumpen wrote: > Group, > > > I am trying to migrate to the latest opensips 2.3 but I am having > issues when setting this in my script. From johan at democon.be Wed Jul 19 05:35:46 2017 From: johan at democon.be (Johan De Clercq) Date: Wed, 19 Jul 2017 11:35:46 +0200 Subject: [OpenSIPS-Users] exec environmental var In-Reply-To: References: Message-ID: to me that seems like the same. 2017-07-19 10:15 GMT+02:00 Liviu Chircu : > Hi Tito, > > This seems to be caused by some misused startup optimization logic. Below > is a scripting trick that should solve your problem until we take care of > providing the official solution. > > Change this: > > ..., "$avp(env)"); > > into this: > > ..., "$avp(env) "); > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 19.07.2017 00:42, Tito Cumpen wrote: > >> Group, >> >> >> I am trying to migrate to the latest opensips 2.3 but I am having issues >> when setting this in my script. >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 19 05:43:42 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 19 Jul 2017 12:43:42 +0300 Subject: [OpenSIPS-Users] exec environmental var In-Reply-To: References: Message-ID: <9be1b58f-391f-879d-d638-797058f6d8f3@opensips.org> I didn't call it a "trick" without good reason :) Notice the extra whitespace (" "), which changes the internal type of that parameter from "single variable" to "variable format string". Nevertheless, the proper fix is already available [1] [1]: https://github.com/OpenSIPS/opensips/commit/8a1b3ef3e456 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 19.07.2017 12:35, Johan De Clercq wrote: > to me that seems like the same. > > 2017-07-19 10:15 GMT+02:00 Liviu Chircu >: > > Hi Tito, > > This seems to be caused by some misused startup optimization > logic. Below is a scripting trick that should solve your problem > until we take care of providing the official solution. > > Change this: > > ..., "$avp(env)"); > > into this: > > ..., "$avp(env) "); > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 19.07.2017 00:42, Tito Cumpen wrote: > > Group, > > > I am trying to migrate to the latest opensips 2.3 but I am > having issues when setting this in my script. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From lamiriahmedamin at gmail.com Mon Jul 17 12:06:42 2017 From: lamiriahmedamin at gmail.com (ahmeddd) Date: Mon, 17 Jul 2017 09:06:42 -0700 (MST) Subject: [OpenSIPS-Users] SIP Trunking In-Reply-To: <45e11ab7-9100-3380-508e-92e062d0e4cf@opensips.org> References: <828940906.1508111250790836724.JavaMail.root@zimbra1.crocker.com> <1500297864682-7608019.post@n2.nabble.com> <45e11ab7-9100-3380-508e-92e062d0e4cf@opensips.org> Message-ID: <1500307602024-7608027.post@n2.nabble.com> Hi! yes it's clear, but I mean how to enable the trunk between two different IP phones which should be registered to my opensips proxy !! should I do somethig special to register them ! or should I specify the domain :( VM1 : x.x.13.87 and VM2 x.x.13.82 ) and for my opensips server (x.x.15.18) ! or it's the same domain as x.x.15.18 !! thank you . -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-Trunking-tp3480673p7608027.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From lamiriahmedamin at gmail.com Tue Jul 18 09:00:26 2017 From: lamiriahmedamin at gmail.com (ahmeddd) Date: Tue, 18 Jul 2017 06:00:26 -0700 (MST) Subject: [OpenSIPS-Users] SIP Trunking In-Reply-To: <45e11ab7-9100-3380-508e-92e062d0e4cf@opensips.org> References: <828940906.1508111250790836724.JavaMail.root@zimbra1.crocker.com> <1500297864682-7608019.post@n2.nabble.com> <45e11ab7-9100-3380-508e-92e062d0e4cf@opensips.org> Message-ID: <1500382826693-7608029.post@n2.nabble.com> Hi! yes it's clear, but I mean how to enable the trunk between two different IP phones which should be registered to my opensips proxy !! should I do somethig special to register them ! or should I specify the domain :( VM1 : x.x.13.87 and VM2 x.x.13.82 ) and for my opensips server (x.x.15.18) ! or it's the same domain as x.x.15.18 !! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-Trunking-tp3480673p7608029.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From bogdan at opensips.org Wed Jul 19 13:52:59 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 19 Jul 2017 20:52:59 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms In-Reply-To: References: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Message-ID: Hi Serge, Have you tried to monitor the load: and net: statistic groups in OpenSIPS to see if there is any internal load in opensips: opensipsctl fifo get_statistics load: Also, are you using the standard cfg for HEP capturing node (form sipcapture.org) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/18/2017 07:04 PM, Bogdan-Andrei Iancu wrote: > Serge, > > what OpenSIPS version you have (opensips -v) ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/18/2017 05:24 PM, Serge S. Yuriev wrote: >> Hi, >> >> Yes just after starting. >> And such lines repeated very often (each 3-5 mins). >> This machine only captures very small amount of HEP traffic. >> I first installed this somewhere in May and there is no such problem >> IIRC. >> I'll try to revert and report back. >> >> On 17/07/17 18:51, Bogdan-Andrei Iancu wrote: >>> Hi Serge, >>> >>> Do you get all these during OpenSIP starting sequence (just after >>> starting it) ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> OpenSIPS Bootcamp 2017, Houston, US >>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>> >>> On 07/17/2017 06:06 PM, Serge S. Yuriev wrote: >>>> Hi, >>>> >>>> OpenSIPS git revision: 02da97c96 >>>> >>>> Just after start without any load. >>>> Log full of >>>> Jul 17 17:45:43 rossiten /usr/local/sbin/opensips[11099]: >>>> WARNING:core:utimer_ticker: utimer task already >>>> scheduled for 100 ms (now 290 ms), it may overlap.. >>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>>> WARNING:core:utimer_ticker: utimer task already >>>> scheduled for 100 ms (now 390 ms), it may overlap.. >>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>>> WARNING:core:utimer_ticker: utimer task already >>>> scheduled for 100 ms (now 490 ms), it may overlap.. >>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>>> WARNING:core:utimer_ticker: utimer task already >>>> scheduled for 100 ms (now 590 ms), it may overlap.. >>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11107]: >>>> WARNING:core:handle_timer_job: utimer job has a 520000 >>>> us delay in execution >>>> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >>>> WARNING:core:handle_timer_job: timer job has a 20000 us >>>> delay in execution >>>> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >>>> WARNING:core:handle_timer_job: timer job has a >>>> 20000 us delay in execution >>>> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a 100000 us >>>> delay in execution >>>> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a >>>> 100000 us delay in execution >>>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a >>>> 100000 us delay in execution >>>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a 100000 us >>>> delay in execution >>>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a >>>> 100000 us delay in execution >>>> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a 90000 us >>>> delay in execution >>>> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >>>> WARNING:core:handle_timer_job: timer job has a >>>> 90000 us delay in execution >>>> >>> >> > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From me at nevian.org Thu Jul 20 04:15:56 2017 From: me at nevian.org (Serge S. Yuriev) Date: Thu, 20 Jul 2017 11:15:56 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already, scheduled for 100 ms In-Reply-To: References: <6b2b0b4c-5e82-a87f-bb17-1e3337b6594d@nevian.org> Message-ID: <8fb4ec23-6d91-f27d-4cae-7b34917229f8@nevian.org> Hello, nevian at rossiten:/var/log$ opensipsctl fifo get_statistics load: load:udp:10.186.45.70:5062-load:: 0 load:hep_udp:10.186.45.70:9063-load:: 0 load:tcp-load:: 0 nevian at rossiten:/var/log$ opensipsctl fifo get_statistics net: net:waiting_udp:: 0 net:waiting_tcp:: 0 net:waiting_tls:: 0 Stats got in the very same time as WARN printed. Config pretty standart, only RTCP stats capture added. On 19/07/17 20:52, Bogdan-Andrei Iancu wrote: > Hi Serge, > > Have you tried to monitor the load: and net: statistic groups in > OpenSIPS to see if there is any internal load in opensips: > opensipsctl fifo get_statistics load: > > Also, are you using the standard cfg for HEP capturing node (form > sipcapture.org) ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/18/2017 07:04 PM, Bogdan-Andrei Iancu wrote: >> Serge, >> >> what OpenSIPS version you have (opensips -v) ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/18/2017 05:24 PM, Serge S. Yuriev wrote: >>> Hi, >>> >>> Yes just after starting. >>> And such lines repeated very often (each 3-5 mins). >>> This machine only captures very small amount of HEP traffic. >>> I first installed this somewhere in May and there is no such problem >>> IIRC. >>> I'll try to revert and report back. >>> >>> On 17/07/17 18:51, Bogdan-Andrei Iancu wrote: >>>> Hi Serge, >>>> >>>> Do you get all these during OpenSIP starting sequence (just after >>>> starting it) ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> OpenSIPS Bootcamp 2017, Houston, US >>>> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >>>> >>>> On 07/17/2017 06:06 PM, Serge S. Yuriev wrote: >>>>> Hi, >>>>> >>>>> OpenSIPS git revision: 02da97c96 >>>>> >>>>> Just after start without any load. >>>>> Log full of >>>>> Jul 17 17:45:43 rossiten /usr/local/sbin/opensips[11099]: >>>>> WARNING:core:utimer_ticker: utimer task already >>>>> scheduled for 100 ms (now 290 ms), it may overlap.. >>>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>>>> WARNING:core:utimer_ticker: utimer task already >>>>> scheduled for 100 ms (now 390 ms), it may overlap.. >>>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>>>> WARNING:core:utimer_ticker: utimer task already >>>>> scheduled for 100 ms (now 490 ms), it may overlap.. >>>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11099]: >>>>> WARNING:core:utimer_ticker: utimer task already >>>>> scheduled for 100 ms (now 590 ms), it may overlap.. >>>>> Jul 17 17:45:44 rossiten /usr/local/sbin/opensips[11107]: >>>>> WARNING:core:handle_timer_job: utimer job has a 520000 >>>>> us delay in execution >>>>> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >>>>> WARNING:core:handle_timer_job: timer job has a 20000 us >>>>> delay in execution >>>>> Jul 17 17:46:44 rossiten /usr/local/sbin/opensips[11100]: >>>>> WARNING:core:handle_timer_job: timer job has a >>>>> 20000 us delay in execution >>>>> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a 100000 us >>>>> delay in execution >>>>> Jul 17 17:50:43 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a >>>>> 100000 us delay in execution >>>>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a >>>>> 100000 us delay in execution >>>>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a 100000 us >>>>> delay in execution >>>>> Jul 17 17:55:44 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a >>>>> 100000 us delay in execution >>>>> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a 90000 us >>>>> delay in execution >>>>> Jul 17 18:00:44 rossiten /usr/local/sbin/opensips[11101]: >>>>> WARNING:core:handle_timer_job: timer job has a >>>>> 90000 us delay in execution >>>>> >>>> >>> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Serge S. Yuriev Lead VoIP engineer From spanda at 3clogic.com Thu Jul 20 05:14:15 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 20 Jul 2017 14:44:15 +0530 Subject: [OpenSIPS-Users] need some information regarding DNS SRV query in opensips proxy . Message-ID: Hi All , My UAC end point uses a proxy before reaching the destination .Againest the destination domain I have added 2 SRV record . So I am expection when the call reaches the proxy that will do SRV query in the destination domain and then send to any IP which ever will get selected from the list . This is happenign in opensips-2.2 by default . I havnt added any extra parameter for DNS SRV query . But in opensips-1.11 it not working by default . When Invite reaches the prosy that wont resove the destinataion host domain and sends "476 unresolve host" to my client . Can anybody let me know what will be the parameter I will use to make this work ?I know I am missing very small things here . Thanks in advance . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 20 08:49:56 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 20 Jul 2017 15:49:56 +0300 Subject: [OpenSIPS-Users] need some information regarding DNS SRV query in opensips proxy . In-Reply-To: References: Message-ID: <4e6aebc2-b917-7e6e-57ba-716f023ceb9c@opensips.org> Hi Sasmita, Post the exact RURI you have before doing the t_relay(). The presence of protocol or port indications in RURI may bypass NAPTR or SRV lookups. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/20/2017 12:14 PM, Sasmita Panda wrote: > Hi All , > > > My UAC end point uses a proxy before reaching the destination > .Againest the destination domain I have added 2 SRV record . So I am > expection when the call reaches the proxy that will do SRV query in > the destination domain and then send to any IP which ever will get > selected from the list . > > This is happenign in opensips-2.2 by default . I havnt added any > extra parameter for DNS SRV query . > > > But in opensips-1.11 it not working by default . When Invite > reaches the prosy that wont resove the destinataion host domain and > sends "476 unresolve host" to my client . > > Can anybody let me know what will be the parameter I will use > to make this work ?I know I am missing very small things here . Thanks > in advance . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu Jul 20 08:52:45 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 20 Jul 2017 18:22:45 +0530 Subject: [OpenSIPS-Users] need some information regarding DNS SRV query in opensips proxy . In-Reply-To: <4e6aebc2-b917-7e6e-57ba-716f023ceb9c@opensips.org> References: <4e6aebc2-b917-7e6e-57ba-716f023ceb9c@opensips.org> Message-ID: I got it . There is a parameter "dns_try_naptr" , default value for this is "yes" . After making it "no" the SRV lookup takes place . I got this from the doc finally . Now its seems like its working for me . Thanks *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Jul 20, 2017 at 6:19 PM, Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > Post the exact RURI you have before doing the t_relay(). The presence of > protocol or port indications in RURI may bypass NAPTR or SRV lookups. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/20/2017 12:14 PM, Sasmita Panda wrote: > > Hi All , > > > My UAC end point uses a proxy before reaching the destination > .Againest the destination domain I have added 2 SRV record . So I am > expection when the call reaches the proxy that will do SRV query in the > destination domain and then send to any IP which ever will get selected > from the list . > > This is happenign in opensips-2.2 by default . I havnt added any > extra parameter for DNS SRV query . > > > But in opensips-1.11 it not working by default . When Invite > reaches the prosy that wont resove the destinataion host domain and sends > "476 unresolve host" to my client . > > Can anybody let me know what will be the parameter I will use to > make this work ?I know I am missing very small things here . Thanks in > advance . > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Thu Jul 20 14:40:29 2017 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Thu, 20 Jul 2017 15:40:29 -0300 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE Message-ID: In what exactly moment the 200OK and BYE messages are accounted and written to the database? At the moment Opensips receive the 200 OK or after receive ACK of 200 OK? Also on BYE, when receive BYE or on 200 OK of BYE? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Thu Jul 20 14:45:47 2017 From: abalashov at evaristesys.com (Alex Balashov) Date: Thu, 20 Jul 2017 14:45:47 -0400 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: References: Message-ID: <20170720184547.GA11475@tlaquepaque.localdomain> On Thu, Jul 20, 2017 at 03:40:29PM -0300, Daniel Zanutti wrote: > In what exactly moment the 200OK and BYE messages are accounted and written > to the database? > > At the moment Opensips receive the 200 OK or after receive ACK of 200 OK? > > Also on BYE, when receive BYE or on 200 OK of BYE? Are you referring to the ACC module, or some other method of accounting? :-) -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From daniel.zanutti at gmail.com Thu Jul 20 14:56:36 2017 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Thu, 20 Jul 2017 15:56:36 -0300 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: <20170720184547.GA11475@tlaquepaque.localdomain> References: <20170720184547.GA11475@tlaquepaque.localdomain> Message-ID: Yes, ACC module. On Thu, Jul 20, 2017 at 3:45 PM, Alex Balashov wrote: > On Thu, Jul 20, 2017 at 03:40:29PM -0300, Daniel Zanutti wrote: > > > In what exactly moment the 200OK and BYE messages are accounted and > written > > to the database? > > > > At the moment Opensips receive the 200 OK or after receive ACK of 200 OK? > > > > Also on BYE, when receive BYE or on 200 OK of BYE? > > Are you referring to the ACC module, or some other method of accounting? > :-) > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Thu Jul 20 15:26:05 2017 From: abalashov at evaristesys.com (Alex Balashov) Date: Thu, 20 Jul 2017 15:26:05 -0400 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: References: <20170720184547.GA11475@tlaquepaque.localdomain> Message-ID: <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> My understanding is that this is a rather simple module without sophisticated state componentry, and that it logs things immediately as received, in the same iteration of message processing. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. From daniel.zanutti at gmail.com Thu Jul 20 15:49:16 2017 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Thu, 20 Jul 2017 16:49:16 -0300 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> References: <20170720184547.GA11475@tlaquepaque.localdomain> <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> Message-ID: Hi Alex I'm having a billing problem from receiving BYE to 200 OK is taking more than 500ms. If BYE is accounted when it's received, great! Are you absolutely sure it works this way? Thanks On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov wrote: > My understanding is that this is a rather simple module without > sophisticated state componentry, and that it logs things immediately as > received, in the same iteration of message processing. > > -- Alex > > -- > Principal, Evariste Systems LLC (www.evaristesys.com) > > Sent from my Google Nexus. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Thu Jul 20 15:57:40 2017 From: abalashov at evaristesys.com (Alex Balashov) Date: Thu, 20 Jul 2017 15:57:40 -0400 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: References: <20170720184547.GA11475@tlaquepaque.localdomain> <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> Message-ID: <20170720195740.GC11475@tlaquepaque.localdomain> On Thu, Jul 20, 2017 at 04:49:16PM -0300, Daniel Zanutti wrote: > I'm having a billing problem from receiving BYE to 200 OK is taking more > than 500ms. If BYE is accounted when it's received, great! > Are you absolutely sure it works this way? As far as I have always seen it behave, yeah. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From Abdirahman at vocantas.com Thu Jul 20 16:07:28 2017 From: Abdirahman at vocantas.com (Abdirahman A. Osman) Date: Thu, 20 Jul 2017 20:07:28 +0000 Subject: [OpenSIPS-Users] Call continuity Message-ID: Hi Everyone, This is a non-related OpenSIPS question, it is a SIP question. I am not expert in SIP , so let me know what you guys think about this. Is it possible to keep a live call continue , if the internet connection fails and route it through another internet connection ? Does SIP protocol support this kind of Call continuity? Thanks Abdirahman Osman -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Thu Jul 20 16:11:38 2017 From: abalashov at evaristesys.com (Alex Balashov) Date: Thu, 20 Jul 2017 16:11:38 -0400 Subject: [OpenSIPS-Users] Call continuity In-Reply-To: References: Message-ID: <20170720201138.GD11475@tlaquepaque.localdomain> On Thu, Jul 20, 2017 at 08:07:28PM +0000, Abdirahman A. Osman wrote: > Is it possible to keep a live call continue , if the internet > connection fails and route it through another internet connection ? > Does SIP protocol support this kind of Call continuity? Generally speaking, no, though the answer will vary with the modalities of the failover mechanism. But in general, failing anything over to another Internet connection means changing the address of one of the endpoints involved. All session-based Internet applications, whether using a connection-orientated transport or not, presume that the IP and port endpoints on both ends stay the same. So, if you suddenly start sending media from another place and expecting to receive it there likewise, that will not be considered to be part of the same phone call. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From voransoy at gmail.com Fri Jul 21 05:34:22 2017 From: voransoy at gmail.com (Volkan Oransoy) Date: Fri, 21 Jul 2017 12:34:22 +0300 Subject: [OpenSIPS-Users] Uac registrant check Message-ID: <8D149D5C-B73E-42C5-BC2C-7BE157F94869@gmail.com> Hi all, I use uac_registrant to register to remote SIP systems and registration phase seems ok. What I want to do is to receive calls from these systems and before accepting calls to my box, I want to check destination, if it is a valid record on my system. I found a couple of replies on list archives and one of them suggests to lookup agains AOR. But that doesn’t work right now. What is the most suitable way to do this? if ( check_source_address("1","$avp(trunk_attrs)") ) { # request comes from trunks setflag(IS_TRUNK); } else if ( is_from_gw() ) { # request comes from GWs } else if ( lookup("location","","$ru") ){ xlog("Location check for $ru passed.\n"); } else { xlog("Location check for $ru failed.\n"); send_reply("403","Forbidden"); exit; } Regards, /Volkan -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Fri Jul 21 09:43:43 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Fri, 21 Jul 2017 10:43:43 -0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm Message-ID: Hello, Im getting this error (below) when trying to compile the tls_mgm on version 2.2.5 (github 2.2) In file included from proto_tls.c:67:0: ../tls_mgm/tls_conn_ops.h: In function ‘tls_conn_init’: ../tls_mgm/tls_conn_ops.h:120:29: error: dereferencing pointer to incomplete type ‘SSL {aka struct ssl_st}’ if ( ((SSL *)c->extra_data)->kssl_ctx ) { ^~ ../tls_mgm/tls_conn_ops.h:121:3: warning: implicit declaration of function ‘kssl_ctx_free’ [-Wimplicit-function-declaration] kssl_ctx_free( ((SSL *)c->extra_data)->kssl_ctx ); ^~~~~~~~~~~~~ The system is a debian 9 with the libs below for tls: libcurl3-gnutls:amd64 - 7.52.1-5 libcurl4-gnutls-dev:amd64 - 7.52.1-5 libgnutls-dane0:amd64 - 3.5.8-5+deb9u1 libgnutls-openssl27:amd64 - 3.5.8-5+deb9u1 libgnutls28-dev:amd64 - 3.5.8-5+deb9u1 libgnutls30:amd64 - 3.5.8-5+deb9u1 libgnutlsxx28:amd64 - 3.5.8-5+deb9u1 Can you confirm how can i solve this ? Thank you. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From razvan at opensips.org Fri Jul 21 10:09:40 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 21 Jul 2017 17:09:40 +0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm In-Reply-To: References: Message-ID: Hello, Mike! OpenSIPS tls module uses libssl. can you tell us what version you are using for libssl-dev? Thanks, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/21/2017 04:43 PM, Mike Tesliuk wrote: > Hello, > > > Im getting this error (below) when trying to compile the tls_mgm on > version 2.2.5 (github 2.2) > > > In file included from proto_tls.c:67:0: > ../tls_mgm/tls_conn_ops.h: In function ‘tls_conn_init’: > ../tls_mgm/tls_conn_ops.h:120:29: error: dereferencing pointer to > incomplete type ‘SSL {aka struct ssl_st}’ > if ( ((SSL *)c->extra_data)->kssl_ctx ) { > ^~ > ../tls_mgm/tls_conn_ops.h:121:3: warning: implicit declaration of > function ‘kssl_ctx_free’ [-Wimplicit-function-declaration] > kssl_ctx_free( ((SSL *)c->extra_data)->kssl_ctx ); > ^~~~~~~~~~~~~ > > > The system is a debian 9 with the libs below for tls: > > libcurl3-gnutls:amd64 - 7.52.1-5 > libcurl4-gnutls-dev:amd64 - 7.52.1-5 > libgnutls-dane0:amd64 - 3.5.8-5+deb9u1 > libgnutls-openssl27:amd64 - 3.5.8-5+deb9u1 > libgnutls28-dev:amd64 - 3.5.8-5+deb9u1 > libgnutls30:amd64 - 3.5.8-5+deb9u1 > libgnutlsxx28:amd64 - 3.5.8-5+deb9u1 > > > Can you confirm how can i solve this ? > > Thank you. > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Fri Jul 21 10:51:21 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Fri, 21 Jul 2017 11:51:21 -0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm In-Reply-To: References: Message-ID: <8c1a96ec-ce3a-87e8-9851-0dcde7362d25@wsu.com.br> Hello Razvan, ii libssl-dev:amd64 1.1.0f-3 amd64 Secure Sockets Layer toolkit - development files is the version from the repository, is a clean installation PS: im sending again cause i send directly to you Razvan not to the list . Em 21/07/17 11:09, Răzvan Crainea escreveu: > Hello, Mike! > > OpenSIPS tls module uses libssl. can you tell us what version you are > using for libssl-dev? > > Thanks, > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > On 07/21/2017 04:43 PM, Mike Tesliuk wrote: >> Hello, >> >> >> Im getting this error (below) when trying to compile the tls_mgm on >> version 2.2.5 (github 2.2) >> >> >> In file included from proto_tls.c:67:0: >> ../tls_mgm/tls_conn_ops.h: In function ‘tls_conn_init’: >> ../tls_mgm/tls_conn_ops.h:120:29: error: dereferencing pointer to >> incomplete type ‘SSL {aka struct ssl_st}’ >> if ( ((SSL *)c->extra_data)->kssl_ctx ) { >> ^~ >> ../tls_mgm/tls_conn_ops.h:121:3: warning: implicit declaration of >> function ‘kssl_ctx_free’ [-Wimplicit-function-declaration] >> kssl_ctx_free( ((SSL *)c->extra_data)->kssl_ctx ); >> ^~~~~~~~~~~~~ >> >> >> The system is a debian 9 with the libs below for tls: >> >> libcurl3-gnutls:amd64 - 7.52.1-5 >> libcurl4-gnutls-dev:amd64 - 7.52.1-5 >> libgnutls-dane0:amd64 - 3.5.8-5+deb9u1 >> libgnutls-openssl27:amd64 - 3.5.8-5+deb9u1 >> libgnutls28-dev:amd64 - 3.5.8-5+deb9u1 >> libgnutls30:amd64 - 3.5.8-5+deb9u1 >> libgnutlsxx28:amd64 - 3.5.8-5+deb9u1 >> >> >> Can you confirm how can i solve this ? >> >> Thank you. >> >> >> >> >> _______________________________________________ >> 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 -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From razvan at opensips.org Fri Jul 21 10:54:18 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 21 Jul 2017 17:54:18 +0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm In-Reply-To: <8c1a96ec-ce3a-87e8-9851-0dcde7362d25@wsu.com.br> References: <8c1a96ec-ce3a-87e8-9851-0dcde7362d25@wsu.com.br> Message-ID: Hi, Mike! OpenSIPS 2.2 is not compatible with the latest 1.1.0 openssl library. Only OpenSIPS 2.3 is compatible with it. You can try to compile tls_mgm with the following command, and post somewhere it's output. CC_EXTRA_OPTS=-DOPENSSL_NO_KRB5 make modules modules=modules/tls_mgm Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/21/2017 05:51 PM, Mike Tesliuk wrote: > > Hello Razvan, > > > ii libssl-dev:amd64 1.1.0f-3 amd64 > Secure Sockets Layer toolkit - development files > > > is the version from the repository, is a clean installation > > > PS: im sending again cause i send directly to you Razvan not to the list . > > > Em 21/07/17 11:09, Răzvan Crainea escreveu: >> Hello, Mike! >> >> OpenSIPS tls module uses libssl. can you tell us what version you are >> using for libssl-dev? >> >> Thanks, >> Răzvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> On 07/21/2017 04:43 PM, Mike Tesliuk wrote: >>> Hello, >>> >>> >>> Im getting this error (below) when trying to compile the tls_mgm on >>> version 2.2.5 (github 2.2) >>> >>> >>> In file included from proto_tls.c:67:0: >>> ../tls_mgm/tls_conn_ops.h: In function ‘tls_conn_init’: >>> ../tls_mgm/tls_conn_ops.h:120:29: error: dereferencing pointer to >>> incomplete type ‘SSL {aka struct ssl_st}’ >>> if ( ((SSL *)c->extra_data)->kssl_ctx ) { >>> ^~ >>> ../tls_mgm/tls_conn_ops.h:121:3: warning: implicit declaration of >>> function ‘kssl_ctx_free’ [-Wimplicit-function-declaration] >>> kssl_ctx_free( ((SSL *)c->extra_data)->kssl_ctx ); >>> ^~~~~~~~~~~~~ >>> >>> >>> The system is a debian 9 with the libs below for tls: >>> >>> libcurl3-gnutls:amd64 - 7.52.1-5 >>> libcurl4-gnutls-dev:amd64 - 7.52.1-5 >>> libgnutls-dane0:amd64 - 3.5.8-5+deb9u1 >>> libgnutls-openssl27:amd64 - 3.5.8-5+deb9u1 >>> libgnutls28-dev:amd64 - 3.5.8-5+deb9u1 >>> libgnutls30:amd64 - 3.5.8-5+deb9u1 >>> libgnutlsxx28:amd64 - 3.5.8-5+deb9u1 >>> >>> >>> Can you confirm how can i solve this ? >>> >>> Thank you. >>> >>> >>> >>> >>> _______________________________________________ >>> 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 > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Fri Jul 21 12:16:23 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Fri, 21 Jul 2017 13:16:23 -0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm In-Reply-To: References: <8c1a96ec-ce3a-87e8-9851-0dcde7362d25@wsu.com.br> Message-ID: <6f4a5002-a924-8e12-5ffc-bc67e1178067@wsu.com.br> I got more errors on this case. i will try the 2.3 version, thank you. Em 21/07/17 11:54, Răzvan Crainea escreveu: > CC_EXTRA_OPTS=-DOPENSSL_NO_KRB5 make modules modules=modules/tls_mgm -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From razvan at opensips.org Fri Jul 21 12:49:47 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 21 Jul 2017 19:49:47 +0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm In-Reply-To: <6f4a5002-a924-8e12-5ffc-bc67e1178067@wsu.com.br> References: <8c1a96ec-ce3a-87e8-9851-0dcde7362d25@wsu.com.br> <6f4a5002-a924-8e12-5ffc-bc67e1178067@wsu.com.br> Message-ID: <90768343-8f08-3b87-8a3b-3487f313d593@opensips.org> Hi, Mike! You can apply this[1] patch to make opensips 2.2 compatible with ssl 1.1.0. However, this has not yet been fully tested, that's why it's not present in the current version. Please give it a try and let me know how it goes. [1] https://sources.debian.net/src/opensips/2.2.2-3/debian/patches/port-tls-1.1.0.patch/ Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/21/2017 07:16 PM, Mike Tesliuk wrote: > > I got more errors on this case. > > > i will try the 2.3 version, thank you. > > > Em 21/07/17 11:54, Răzvan Crainea escreveu: >> CC_EXTRA_OPTS=-DOPENSSL_NO_KRB5 make modules modules=modules/tls_mgm > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goley_ev_sp at mail.ru Fri Jul 21 14:31:41 2017 From: goley_ev_sp at mail.ru (dgoni_sp) Date: Fri, 21 Jul 2017 11:31:41 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory Message-ID: <1500661901795-7608065.post@n2.nabble.com> Hello! Please, tell me. Opensips for some reason does not release shared memory (shmem), it was tested as in the documentation "https://www.opensips.org/Documentation/TroubleShooting-OutOfMem". Waited more than 20 minutes. Now I want to get a memory dump. I tried the same results on versions 2.2.5 and 2.3.1. Set for comparison kamailio, with a similar configuration, the memory is freed immediately, once the call is completed. What could be the problem ? Opensips v.2.3.1 OC CentOS 7 (virtual machine) At startup [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl start INFO: Starting OpenSIPS : INFO: started (pid: 4274) [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl fifo get_statistics shmem: shmem:total_size:: 33554432 shmem:used_size:: 3271272 shmem:real_used_size:: 3557488 shmem:max_used_size:: 3557488 shmem:free_size:: 29996944 shmem:fragments:: 490 I made one call. [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl fifo get_statistics shmem: shmem:total_size:: 33554432 shmem:used_size:: 3336136 shmem:real_used_size:: 3632024 shmem:max_used_size:: 3635208 shmem:free_size:: 29922408 shmem:fragments:: 577 [root at new-centos7 ~]# In 20 minutes. [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl fifo get_statistics shmem: shmem:total_size:: 33554432 shmem:used_size:: 3336136 shmem:real_used_size:: 3632024 shmem:max_used_size:: 3635208 shmem:free_size:: 29922408 shmem:fragments:: 577 Stopped [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl stop INFO: Stopping OpenSIPS : INFO: stopped [root at new-centos7 ~]# Dump memory 2017-07-21T18:44:57.5212 B2BUA Memory status (pkg): 2017-07-21T18:44:57.5212 B2BUA qm_status (0x7f1d017fe010): 2017-07-21T18:44:57.5213 B2BUA heap size= 2097152 2017-07-21T18:44:57.5214 B2BUA used= 63904, used+overhead=350224, free=1746928 2017-07-21T18:44:57.5215 B2BUA max used (+overhead)= 355224 2017-07-21T18:44:57.5216 B2BUA dumping summary of all alloc'ed. fragments: 2017-07-21T18:44:57.5217 B2BUA 32 : 2 x [script_var.c: add_var, line 59] 2017-07-21T18:44:57.5218 B2BUA 14400 : 72 x [route_struct.c: mk_action, line 106] 2017-07-21T18:44:57.5218 B2BUA 80 : 2 x [cfg.y: mk_listen_id, line 2785] 2017-07-21T18:44:57.5219 B2BUA 48 : 1 x [ipc.c: ipc_register_handler, line 50] 2017-07-21T18:44:57.5220 B2BUA 320 : 2 x [db/db.c: db_do_init, line 321] 2017-07-21T18:44:57.5220 B2BUA 80 : 2 x [pvar.c: new_pv_context, line 4745] 2017-07-21T18:44:57.5221 B2BUA 96 : 2 x [script_var.c: add_var, line 52] 2017-07-21T18:44:57.5222 B2BUA 64 : 1 x [ds_fixups.c: set_list_from_string, line 226] 2017-07-21T18:44:57.5223 B2BUA 16 : 1 x [ds_fixups.c: ds_select_fixup, line 730] 2017-07-21T18:44:57.5224 B2BUA 96 : 2 x [sr_module_deps.c: alloc_module_dep, line 54] 2017-07-21T18:44:57.5225 B2BUA 128 : 1 x [net/net_tcp.c: tcp_init, line 1633] 2017-07-21T18:44:57.5226 B2BUA 80 : 1 x [mi/mi_trace.c: try_load_trace_api, line 55] 2017-07-21T18:44:57.5226 B2BUA 24 : 1 x [mod_fix.c: fixup_spve, line 938] 2017-07-21T18:44:57.5227 B2BUA 32 : 2 x [socket_info.c: new_sock_info, line 116] 2017-07-21T18:44:57.5228 B2BUA 192 : 15 x [map.c: map_get, line 150] 2017-07-21T18:44:57.5229 B2BUA 32 : 1 x [map.c: map_create, line 79] 2017-07-21T18:44:57.5229 B2BUA 48 : 2 x [socket_info.c: fix_socket_list, line 670] 2017-07-21T18:44:57.5230 B2BUA 184 : 5 x [mod_fix.c: fixup_sgp, line 771] 2017-07-21T18:44:57.5231 B2BUA 24 : 1 x [ds_fixups.c: set_list_from_string, line 197] 2017-07-21T18:44:57.5232 B2BUA 4928 : 37 x [pvar.c: pv_parse_format, line 4099] 2017-07-21T18:44:57.5232 B2BUA 1768 : 17 x [cfg.y: yyparse, line 2299] 2017-07-21T18:44:57.5233 B2BUA 32 : 2 x [socket_info.c: fix_socket_list, line 591] 2017-07-21T18:44:57.5234 B2BUA 2496 : 24 x [cfg.y: yyparse, line 1464] 2017-07-21T18:44:57.5235 B2BUA 24 : 1 x [ds_fixups.c: set_list_from_string, line 146] 2017-07-21T18:44:57.5236 B2BUA 26240 : 205 x [cfg.lex: addstr, line 919] 2017-07-21T18:44:57.5237 B2BUA 480 : 2 x [socket_info.c: new_sock_info, line 111] 2017-07-21T18:44:57.5237 B2BUA 4216 : 1 x [mi/mi.c: register_mi_cmd, line 146] 2017-07-21T18:44:57.5238 B2BUA 104 : 1 x [ds_fixups.c: set_list_from_string, line 232] 2017-07-21T18:44:57.5239 B2BUA 1792 : 32 x [route_struct.c: mk_elem, line 70] 2017-07-21T18:44:57.5239 B2BUA 72 : 2 x [socket_info.c: fix_socket_list, line 540] 2017-07-21T18:44:57.5240 B2BUA 32 : 2 x [cfg.y: yyparse, line 519] 2017-07-21T18:44:57.5241 B2BUA 16 : 1 x [b2b_logic.h: prepare_b2b_scen_fl_struct, line 179] 2017-07-21T18:44:57.5241 B2BUA 104 : 1 x [context.c: register_context_destroy, line 74] 2017-07-21T18:44:57.5242 B2BUA 4104 : 1 x [xlog.c: buf_init, line 69] 2017-07-21T18:44:57.5243 B2BUA 48 : 2 x [cfg.y: yyparse, line 1162] 2017-07-21T18:44:57.5244 B2BUA 128 : 3 x [sipmsgops.c: fixup_method, line 892] 2017-07-21T18:44:57.5245 B2BUA 504 : 9 x [route_struct.c: mk_exp, line 54] 2017-07-21T18:44:57.5245 B2BUA 840 : 15 x [map.c: map_get, line 139] 2017-07-21T18:44:57.5246 B2BUA dumping free list stats : 2017-07-21T18:44:57.5247 B2BUA hash= 5. fragments no.: 1, unused: 0#012#011#011 bucket size: 40 - 40 (first 40) 2017-07-21T18:44:57.5248 B2BUA hash= 6. fragments no.: 11, unused: 0#012#011#011 bucket size: 48 - 48 (first 48) 2017-07-21T18:44:57.5249 B2BUA hash= 19. fragments no.: 1, unused: 0#012#011#011 bucket size: 152 - 152 (first 152) 2017-07-21T18:44:57.5250 B2BUA hash= 23. fragments no.: 1, unused: 0#012#011#011 bucket size: 184 - 184 (first 184) 2017-07-21T18:44:57.5250 B2BUA hash= 29. fragments no.: 1, unused: 0#012#011#011 bucket size: 232 - 232 (first 232) 2017-07-21T18:44:57.5251 B2BUA hash= 181. fragments no.: 1, unused: 0#012#011#011 bucket size: 1448 - 1448 (first 1448) 2017-07-21T18:44:57.5252 B2BUA hash= 302. fragments no.: 1, unused: 0#012#011#011 bucket size: 2416 - 2416 (first 2416) 2017-07-21T18:44:57.5253 B2BUA hash= 2055. fragments no.: 1, unused: 0#012#011#011 bucket size: 1048576 - 2097152 (first 1741928) 2017-07-21T18:44:57.5254 B2BUA ----------------------------- 2017-07-21T18:44:57.5255 B2BUA shm_free(0x7f1cffb592a8), called from main.c: cleanup(337) 2017-07-21T18:44:57.5256 B2BUA freeing frag. 0x7f1cffb59270 alloc'ed from pt.c: init_multi_proc_support(69) 2017-07-21T18:44:57.5256 B2BUA Memory status (shm): 2017-07-21T18:44:57.5257 B2BUA qm_status (0x7f1cff7fe000): 2017-07-21T18:44:57.5258 B2BUA heap size= 33554432 2017-07-21T18:44:57.5258 B2BUA used= 2856, used+overhead=245912, free=33308520 2017-07-21T18:44:57.5259 B2BUA max used (+overhead)= 3635208 2017-07-21T18:44:57.5260 B2BUA dumping summary of all alloc'ed. fragments: 2017-07-21T18:44:57.5261 B2BUA 16 : 2 x [statistics.c: register_udp_load_stat, line 160] 2017-07-21T18:44:57.5262 B2BUA 280 : 2 x [statistics.c: register_udp_load_stat, line 152] 2017-07-21T18:44:57.5262 B2BUA 56 : 2 x [statistics.c: build_stat_name, line 122] 2017-07-21T18:44:57.5263 B2BUA 264 : 33 x [mi/mi.c: register_mi_cmd, line 174] 2017-07-21T18:44:57.5264 B2BUA 616 : 5 x [timer.c: new_os_timer, line 145] 2017-07-21T18:44:57.5265 B2BUA 64 : 1 x [statistics.c: register_tcp_load_stat, line 179] 2017-07-21T18:44:57.5266 B2BUA 32 : 1 x [map.c: map_create, line 79] 2017-07-21T18:44:57.5266 B2BUA 864 : 1 x [core_stats.c: init_pkg_stats, line 173] 2017-07-21T18:44:57.5267 B2BUA 8 : 1 x [timer.c: init_timer, line 82] 2017-07-21T18:44:57.5268 B2BUA 8 : 1 x [usr_avp.c: init_extra_avps, line 83] 2017-07-21T18:44:57.5269 B2BUA 24 : 1 x [ds_fixups.c: fixup_partition_sets_null, line 386] 2017-07-21T18:44:57.5270 B2BUA 40 : 5 x [evi/event_interface.c: evi_publish_event, line 75] 2017-07-21T18:44:57.5271 B2BUA 8 : 1 x [statistics.c: register_tcp_load_stat, line 186] 2017-07-21T18:44:57.5271 B2BUA 8 : 1 x [usr_avp.c: init_extra_avps, line 74] 2017-07-21T18:44:57.5272 B2BUA 144 : 1 x [core_stats.c: init_pkg_stats, line 174] 2017-07-21T18:44:57.5274 B2BUA 400 : 1 x [evi/event_interface.c: evi_publish_event, line 61] 2017-07-21T18:44:57.5275 B2BUA 8 : 1 x [mem/shm_mem.c: shm_mem_init_mallocs, line 387] 2017-07-21T18:44:57.5275 B2BUA 8 : 1 x [dispatch.c: init_ds_data, line 98] 2017-07-21T18:44:57.5276 B2BUA 8 : 1 x [daemonize.c: create_status_pipe, line 90] 2017-07-21T18:44:57.5277 B2BUA dumping free list stats : 2017-07-21T18:44:57.5278 B2BUA hash= 12. fragments no.: 1, unused: 0#012#011#011 bucket size: 96 - 96 (first 96) 2017-07-21T18:44:57.5278 B2BUA hash= 15. fragments no.: 1, unused: 0#012#011#011 bucket size: 120 - 120 (first 120) 2017-07-21T18:44:57.5279 B2BUA hash= 37. fragments no.: 1, unused: 0#012#011#011 bucket size: 296 - 296 (first 296) 2017-07-21T18:44:57.5280 B2BUA hash= 50. fragments no.: 1, unused: 0#012#011#011 bucket size: 400 - 400 (first 400) 2017-07-21T18:44:57.5281 B2BUA hash= 69. fragments no.: 1, unused: 0#012#011#011 bucket size: 552 - 552 (first 552) 2017-07-21T18:44:57.5281 B2BUA hash= 378. fragments no.: 1, unused: 0#012#011#011 bucket size: 3024 - 3024 (first 3024) 2017-07-21T18:44:57.5282 B2BUA hash= 434. fragments no.: 1, unused: 0#012#011#011 bucket size: 3472 - 3472 (first 3472) 2017-07-21T18:44:57.5283 B2BUA hash= 498. fragments no.: 1, unused: 0#012#011#011 bucket size: 3984 - 3984 (first 3984) 2017-07-21T18:44:57.5284 B2BUA hash= 715. fragments no.: 1, unused: 0#012#011#011 bucket size: 5720 - 5720 (first 5720) 2017-07-21T18:44:57.5284 B2BUA hash= 2049. fragments no.: 1, unused: 0#012#011#011 bucket size: 16384 - 32768 (first 30672) 2017-07-21T18:44:57.5285 B2BUA hash= 2051. fragments no.: 1, unused: 0#012#011#011 bucket size: 65536 - 131072 (first 101152) 2017-07-21T18:44:57.5286 B2BUA hash= 2054. fragments no.: 1, unused: 0#012#011#011 bucket size: 524288 - 1048576 (first 539096) 2017-07-21T18:44:57.5287 B2BUA hash= 2056. fragments no.: 1, unused: 0#012#011#011 bucket size: 2097152 - 4194304 (first 2622816) 2017-07-21T18:44:57.5287 B2BUA hash= 2059. fragments no.: 1, unused: 0#012#011#011 bucket size: 16777216 - 33554432 (first 29997120) 2017-07-21T18:44:57.5288 B2BUA ----------------------------- 2017-07-21T18:44:57.5289 B2BUA DBG:core:shm_mem_destroy: destroying the shared memory lock ^C [root at new-centos7 sbin]# Sorry for bad english =) -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From liviu at opensips.org Fri Jul 21 16:52:20 2017 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 21 Jul 2017 23:52:20 +0300 Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1500661901795-7608065.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> Message-ID: <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> Hi, dgoni! That's not a really fair test, IMO. There are plenty of one-time buffer allocations happening in all sorts of modules. Can you please confirm if your diagnostic is still valid even after you run some more calls through OpenSIPS? The leak should be obvious in the memory map, too - this isn't the case now. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.coaf On 21.07.2017 21:31, dgoni_sp via Users wrote: > Hello! > > Please, tell me. > Opensips for some reason does not release shared memory (shmem), it was > tested as in the documentation > "https://www.opensips.org/Documentation/TroubleShooting-OutOfMem". Waited > more than 20 minutes. Now I want to get a memory dump. I tried the same > results on versions 2.2.5 and 2.3.1. Set for comparison kamailio, with a > similar configuration, the memory is freed immediately, once the call is > completed. What could be the problem ? > Opensips v.2.3.1 OC CentOS 7 (virtual machine) > At startup > > [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl start > > INFO: Starting OpenSIPS : > INFO: started (pid: 4274) > > [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl fifo > get_statistics shmem: > shmem:total_size:: 33554432 > shmem:used_size:: 3271272 > shmem:real_used_size:: 3557488 > shmem:max_used_size:: 3557488 > shmem:free_size:: 29996944 > shmem:fragments:: 490 > > I made one call. > > [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl fifo > get_statistics shmem: > shmem:total_size:: 33554432 > shmem:used_size:: 3336136 > shmem:real_used_size:: 3632024 > shmem:max_used_size:: 3635208 > shmem:free_size:: 29922408 > shmem:fragments:: 577 > [root at new-centos7 ~]# > > In 20 minutes. > > [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl fifo > get_statistics shmem: > shmem:total_size:: 33554432 > shmem:used_size:: 3336136 > shmem:real_used_size:: 3632024 > shmem:max_used_size:: 3635208 > shmem:free_size:: 29922408 > shmem:fragments:: 577 > > Stopped > [root at new-centos7 ~]# /usr/local/opensips225/sbin/opensipsctl stop > > INFO: Stopping OpenSIPS : > INFO: stopped > [root at new-centos7 ~]# > > Dump memory > 2017-07-21T18:44:57.5212 B2BUA Memory status (pkg): > 2017-07-21T18:44:57.5212 B2BUA qm_status (0x7f1d017fe010): > 2017-07-21T18:44:57.5213 B2BUA heap size= 2097152 > 2017-07-21T18:44:57.5214 B2BUA used= 63904, used+overhead=350224, > free=1746928 > 2017-07-21T18:44:57.5215 B2BUA max used (+overhead)= 355224 > 2017-07-21T18:44:57.5216 B2BUA dumping summary of all alloc'ed. fragments: > 2017-07-21T18:44:57.5217 B2BUA 32 : 2 x [script_var.c: add_var, > line 59] > 2017-07-21T18:44:57.5218 B2BUA 14400 : 72 x [route_struct.c: > mk_action, line 106] > 2017-07-21T18:44:57.5218 B2BUA 80 : 2 x [cfg.y: mk_listen_id, line > 2785] > 2017-07-21T18:44:57.5219 B2BUA 48 : 1 x [ipc.c: > ipc_register_handler, line 50] > 2017-07-21T18:44:57.5220 B2BUA 320 : 2 x [db/db.c: db_do_init, line > 321] > 2017-07-21T18:44:57.5220 B2BUA 80 : 2 x [pvar.c: new_pv_context, > line 4745] > 2017-07-21T18:44:57.5221 B2BUA 96 : 2 x [script_var.c: add_var, > line 52] > 2017-07-21T18:44:57.5222 B2BUA 64 : 1 x [ds_fixups.c: > set_list_from_string, line 226] > 2017-07-21T18:44:57.5223 B2BUA 16 : 1 x [ds_fixups.c: > ds_select_fixup, line 730] > 2017-07-21T18:44:57.5224 B2BUA 96 : 2 x [sr_module_deps.c: > alloc_module_dep, line 54] > 2017-07-21T18:44:57.5225 B2BUA 128 : 1 x [net/net_tcp.c: tcp_init, > line 1633] > 2017-07-21T18:44:57.5226 B2BUA 80 : 1 x [mi/mi_trace.c: > try_load_trace_api, line 55] > 2017-07-21T18:44:57.5226 B2BUA 24 : 1 x [mod_fix.c: fixup_spve, > line 938] > 2017-07-21T18:44:57.5227 B2BUA 32 : 2 x [socket_info.c: > new_sock_info, line 116] > 2017-07-21T18:44:57.5228 B2BUA 192 : 15 x [map.c: map_get, line 150] > 2017-07-21T18:44:57.5229 B2BUA 32 : 1 x [map.c: map_create, line > 79] > 2017-07-21T18:44:57.5229 B2BUA 48 : 2 x [socket_info.c: > fix_socket_list, line 670] > 2017-07-21T18:44:57.5230 B2BUA 184 : 5 x [mod_fix.c: fixup_sgp, line > 771] > 2017-07-21T18:44:57.5231 B2BUA 24 : 1 x [ds_fixups.c: > set_list_from_string, line 197] > 2017-07-21T18:44:57.5232 B2BUA 4928 : 37 x [pvar.c: pv_parse_format, > line 4099] > 2017-07-21T18:44:57.5232 B2BUA 1768 : 17 x [cfg.y: yyparse, line > 2299] > 2017-07-21T18:44:57.5233 B2BUA 32 : 2 x [socket_info.c: > fix_socket_list, line 591] > 2017-07-21T18:44:57.5234 B2BUA 2496 : 24 x [cfg.y: yyparse, line > 1464] > 2017-07-21T18:44:57.5235 B2BUA 24 : 1 x [ds_fixups.c: > set_list_from_string, line 146] > 2017-07-21T18:44:57.5236 B2BUA 26240 : 205 x [cfg.lex: addstr, line > 919] > 2017-07-21T18:44:57.5237 B2BUA 480 : 2 x [socket_info.c: > new_sock_info, line 111] > 2017-07-21T18:44:57.5237 B2BUA 4216 : 1 x [mi/mi.c: register_mi_cmd, > line 146] > 2017-07-21T18:44:57.5238 B2BUA 104 : 1 x [ds_fixups.c: > set_list_from_string, line 232] > 2017-07-21T18:44:57.5239 B2BUA 1792 : 32 x [route_struct.c: mk_elem, > line 70] > 2017-07-21T18:44:57.5239 B2BUA 72 : 2 x [socket_info.c: > fix_socket_list, line 540] > 2017-07-21T18:44:57.5240 B2BUA 32 : 2 x [cfg.y: yyparse, line 519] > 2017-07-21T18:44:57.5241 B2BUA 16 : 1 x [b2b_logic.h: > prepare_b2b_scen_fl_struct, line 179] > 2017-07-21T18:44:57.5241 B2BUA 104 : 1 x [context.c: > register_context_destroy, line 74] > 2017-07-21T18:44:57.5242 B2BUA 4104 : 1 x [xlog.c: buf_init, line 69] > 2017-07-21T18:44:57.5243 B2BUA 48 : 2 x [cfg.y: yyparse, line 1162] > 2017-07-21T18:44:57.5244 B2BUA 128 : 3 x [sipmsgops.c: fixup_method, > line 892] > 2017-07-21T18:44:57.5245 B2BUA 504 : 9 x [route_struct.c: mk_exp, > line 54] > 2017-07-21T18:44:57.5245 B2BUA 840 : 15 x [map.c: map_get, line 139] > 2017-07-21T18:44:57.5246 B2BUA dumping free list stats : > 2017-07-21T18:44:57.5247 B2BUA hash= 5. fragments no.: 1, unused: > 0#012#011#011 bucket size: 40 - 40 (first 40) > 2017-07-21T18:44:57.5248 B2BUA hash= 6. fragments no.: 11, unused: > 0#012#011#011 bucket size: 48 - 48 (first 48) > 2017-07-21T18:44:57.5249 B2BUA hash= 19. fragments no.: 1, unused: > 0#012#011#011 bucket size: 152 - 152 (first 152) > 2017-07-21T18:44:57.5250 B2BUA hash= 23. fragments no.: 1, unused: > 0#012#011#011 bucket size: 184 - 184 (first 184) > 2017-07-21T18:44:57.5250 B2BUA hash= 29. fragments no.: 1, unused: > 0#012#011#011 bucket size: 232 - 232 (first 232) > 2017-07-21T18:44:57.5251 B2BUA hash= 181. fragments no.: 1, unused: > 0#012#011#011 bucket size: 1448 - 1448 (first 1448) > 2017-07-21T18:44:57.5252 B2BUA hash= 302. fragments no.: 1, unused: > 0#012#011#011 bucket size: 2416 - 2416 (first 2416) > 2017-07-21T18:44:57.5253 B2BUA hash= 2055. fragments no.: 1, unused: > 0#012#011#011 bucket size: 1048576 - 2097152 (first 1741928) > 2017-07-21T18:44:57.5254 B2BUA ----------------------------- > 2017-07-21T18:44:57.5255 B2BUA shm_free(0x7f1cffb592a8), called from main.c: > cleanup(337) > 2017-07-21T18:44:57.5256 B2BUA freeing frag. 0x7f1cffb59270 alloc'ed from > pt.c: init_multi_proc_support(69) > 2017-07-21T18:44:57.5256 B2BUA Memory status (shm): > 2017-07-21T18:44:57.5257 B2BUA qm_status (0x7f1cff7fe000): > 2017-07-21T18:44:57.5258 B2BUA heap size= 33554432 > 2017-07-21T18:44:57.5258 B2BUA used= 2856, used+overhead=245912, > free=33308520 > 2017-07-21T18:44:57.5259 B2BUA max used (+overhead)= 3635208 > 2017-07-21T18:44:57.5260 B2BUA dumping summary of all alloc'ed. fragments: > 2017-07-21T18:44:57.5261 B2BUA 16 : 2 x [statistics.c: > register_udp_load_stat, line 160] > 2017-07-21T18:44:57.5262 B2BUA 280 : 2 x [statistics.c: > register_udp_load_stat, line 152] > 2017-07-21T18:44:57.5262 B2BUA 56 : 2 x [statistics.c: > build_stat_name, line 122] > 2017-07-21T18:44:57.5263 B2BUA 264 : 33 x [mi/mi.c: register_mi_cmd, > line 174] > 2017-07-21T18:44:57.5264 B2BUA 616 : 5 x [timer.c: new_os_timer, > line 145] > 2017-07-21T18:44:57.5265 B2BUA 64 : 1 x [statistics.c: > register_tcp_load_stat, line 179] > 2017-07-21T18:44:57.5266 B2BUA 32 : 1 x [map.c: map_create, line > 79] > 2017-07-21T18:44:57.5266 B2BUA 864 : 1 x [core_stats.c: > init_pkg_stats, line 173] > 2017-07-21T18:44:57.5267 B2BUA 8 : 1 x [timer.c: init_timer, line > 82] > 2017-07-21T18:44:57.5268 B2BUA 8 : 1 x [usr_avp.c: > init_extra_avps, line 83] > 2017-07-21T18:44:57.5269 B2BUA 24 : 1 x [ds_fixups.c: > fixup_partition_sets_null, line 386] > 2017-07-21T18:44:57.5270 B2BUA 40 : 5 x [evi/event_interface.c: > evi_publish_event, line 75] > 2017-07-21T18:44:57.5271 B2BUA 8 : 1 x [statistics.c: > register_tcp_load_stat, line 186] > 2017-07-21T18:44:57.5271 B2BUA 8 : 1 x [usr_avp.c: > init_extra_avps, line 74] > 2017-07-21T18:44:57.5272 B2BUA 144 : 1 x [core_stats.c: > init_pkg_stats, line 174] > 2017-07-21T18:44:57.5274 B2BUA 400 : 1 x [evi/event_interface.c: > evi_publish_event, line 61] > 2017-07-21T18:44:57.5275 B2BUA 8 : 1 x [mem/shm_mem.c: > shm_mem_init_mallocs, line 387] > 2017-07-21T18:44:57.5275 B2BUA 8 : 1 x [dispatch.c: init_ds_data, > line 98] > 2017-07-21T18:44:57.5276 B2BUA 8 : 1 x [daemonize.c: > create_status_pipe, line 90] > 2017-07-21T18:44:57.5277 B2BUA dumping free list stats : > 2017-07-21T18:44:57.5278 B2BUA hash= 12. fragments no.: 1, unused: > 0#012#011#011 bucket size: 96 - 96 (first 96) > 2017-07-21T18:44:57.5278 B2BUA hash= 15. fragments no.: 1, unused: > 0#012#011#011 bucket size: 120 - 120 (first 120) > 2017-07-21T18:44:57.5279 B2BUA hash= 37. fragments no.: 1, unused: > 0#012#011#011 bucket size: 296 - 296 (first 296) > 2017-07-21T18:44:57.5280 B2BUA hash= 50. fragments no.: 1, unused: > 0#012#011#011 bucket size: 400 - 400 (first 400) > 2017-07-21T18:44:57.5281 B2BUA hash= 69. fragments no.: 1, unused: > 0#012#011#011 bucket size: 552 - 552 (first 552) > 2017-07-21T18:44:57.5281 B2BUA hash= 378. fragments no.: 1, unused: > 0#012#011#011 bucket size: 3024 - 3024 (first 3024) > 2017-07-21T18:44:57.5282 B2BUA hash= 434. fragments no.: 1, unused: > 0#012#011#011 bucket size: 3472 - 3472 (first 3472) > 2017-07-21T18:44:57.5283 B2BUA hash= 498. fragments no.: 1, unused: > 0#012#011#011 bucket size: 3984 - 3984 (first 3984) > 2017-07-21T18:44:57.5284 B2BUA hash= 715. fragments no.: 1, unused: > 0#012#011#011 bucket size: 5720 - 5720 (first 5720) > 2017-07-21T18:44:57.5284 B2BUA hash= 2049. fragments no.: 1, unused: > 0#012#011#011 bucket size: 16384 - 32768 (first 30672) > 2017-07-21T18:44:57.5285 B2BUA hash= 2051. fragments no.: 1, unused: > 0#012#011#011 bucket size: 65536 - 131072 (first 101152) > 2017-07-21T18:44:57.5286 B2BUA hash= 2054. fragments no.: 1, unused: > 0#012#011#011 bucket size: 524288 - 1048576 (first 539096) > 2017-07-21T18:44:57.5287 B2BUA hash= 2056. fragments no.: 1, unused: > 0#012#011#011 bucket size: 2097152 - 4194304 (first 2622816) > 2017-07-21T18:44:57.5287 B2BUA hash= 2059. fragments no.: 1, unused: > 0#012#011#011 bucket size: 16777216 - 33554432 (first 29997120) > 2017-07-21T18:44:57.5288 B2BUA ----------------------------- > 2017-07-21T18:44:57.5289 B2BUA DBG:core:shm_mem_destroy: destroying the > shared memory lock > ^C > [root at new-centos7 sbin]# > > > Sorry for bad english =) > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From mike at wsu.com.br Sat Jul 22 08:09:02 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sat, 22 Jul 2017 09:09:02 -0300 Subject: [OpenSIPS-Users] Error in compile of tls_mgm In-Reply-To: <90768343-8f08-3b87-8a3b-3487f313d593@opensips.org> References: <8c1a96ec-ce3a-87e8-9851-0dcde7362d25@wsu.com.br> <6f4a5002-a924-8e12-5ffc-bc67e1178067@wsu.com.br> <90768343-8f08-3b87-8a3b-3487f313d593@opensips.org> Message-ID: i got a Hunk [2017-07-22 08:07:33] root at opensipsHomolog /usr/src/opensips-2.2 # patch -p1 < /root/port-tls-1.1.0.patch patching file modules/tls_mgm/tls.h patching file modules/tls_mgm/tls_conn_ops.h patching file modules/tls_mgm/tls_conn_server.h patching file modules/tls_mgm/tls_mgm.c Hunk #3 succeeded at 1154 (offset 81 lines). Hunk #4 succeeded at 1178 (offset 81 lines). Hunk #5 succeeded at 1229 (offset 81 lines). Hunk #6 succeeded at 1239 (offset 81 lines). Hunk #7 succeeded at 1388 (offset 80 lines). Hunk #8 succeeded at 1406 (offset 80 lines). Hunk #9 succeeded at 1436 with fuzz 1 (offset 80 lines). patching file modules/identity/identity.c i will try to apply mannually and let you know Em 21/07/17 13:49, Răzvan Crainea escreveu: > Hi, Mike! > > You can apply this[1] patch to make opensips 2.2 compatible with ssl > 1.1.0. However, this has not yet been fully tested, that's why it's > not present in the current version. > Please give it a try and let me know how it goes. > > [1] > https://sources.debian.net/src/opensips/2.2.2-3/debian/patches/port-tls-1.1.0.patch/ > > Best regards, > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > On 07/21/2017 07:16 PM, Mike Tesliuk wrote: >> >> I got more errors on this case. >> >> >> i will try the 2.3 version, thank you. >> >> >> Em 21/07/17 11:54, Răzvan Crainea escreveu: >>> CC_EXTRA_OPTS=-DOPENSSL_NO_KRB5 make modules modules=modules/tls_mgm >> >> -- >> >> >> ​Atenciosamente, >> WSU TECNOLOGIA >> Mike Tesliuk >> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >> 12387 SW 125th ter, Miami, Florida 33186 - USA >> tel +55 (41) 3941.0650 +1 (786) 719.6253 >> *website | mapa >> | email >> * >> . >> >> >> >> _______________________________________________ >> 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 -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Sat Jul 22 10:18:37 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sat, 22 Jul 2017 11:18:37 -0300 Subject: [OpenSIPS-Users] error on script generation Message-ID: <1a3db7da-2506-2f4e-1ecc-bc2530da05c6@wsu.com.br> Hello there, im compiling the opensips 2.3 and the generated script with tls generate the lines below: modparam("proto_tls","verify_cert", "1") modparam("proto_tls","require_cert", "0") modparam("proto_tls","tls_method", "TLSv1") modparam("proto_tls","certificate", "/usr/local/etc/opensips/tls/user/user-cert.pem") modparam("proto_tls","private_key", "/usr/local/etc/opensips/tls/user/user-privkey.pem") modparam("proto_tls","ca_list", "/usr/local/etc/opensips/tls/user/user-calist.pem") Those parameters are from tls_mgm not from proto_tls right ? on module documentation are on tls_mgm section -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Sat Jul 22 15:41:11 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sat, 22 Jul 2017 16:41:11 -0300 Subject: [OpenSIPS-Users] Call continuity In-Reply-To: <20170720201138.GD11475@tlaquepaque.localdomain> References: <20170720201138.GD11475@tlaquepaque.localdomain> Message-ID: <7a4a059b-68cf-bada-1020-01872531ec69@wsu.com.br> This is the kind of structure where you need a BGP session with your carriers, when you have an ASN and your own IP Block you can have your communication flowing between any internet links you want, as the ip address will flow between all of them, this is the right way. if you have two simples internet connections there is no way to do that Em 20/07/17 17:11, Alex Balashov escreveu: > On Thu, Jul 20, 2017 at 08:07:28PM +0000, Abdirahman A. Osman wrote: > >> Is it possible to keep a live call continue , if the internet >> connection fails and route it through another internet connection ? >> Does SIP protocol support this kind of Call continuity? > Generally speaking, no, though the answer will vary with the modalities > of the failover mechanism. > > But in general, failing anything over to another Internet connection > means changing the address of one of the endpoints involved. All > session-based Internet applications, whether using a > connection-orientated transport or not, presume that the IP and port > endpoints on both ends stay the same. > > So, if you suddenly start sending media from another place and expecting > to receive it there likewise, that will not be considered to be part of > the same phone call. > > -- Alex > -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Sat Jul 22 17:36:40 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sat, 22 Jul 2017 18:36:40 -0300 Subject: [OpenSIPS-Users] LoadBalancer and Clusterer Message-ID: <5350f828-51b6-aaea-0652-adf6c0b2d754@wsu.com.br> Hello, On the past, i had implemented the dialog with cachedb and load_balancer using a nosql, using the resource with the /s , as the load_balancer have the parameter receive the replication, how i use that ? without the /s when i create a call i do not se the resource being used on node 2 , is that supposed to happen ? Thank you. PS: testing the 2.3 version -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Sun Jul 23 10:13:29 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sun, 23 Jul 2017 11:13:29 -0300 Subject: [OpenSIPS-Users] Permission denied bo bind port 443 or 80 Message-ID: <05da99c0-9a96-89cd-14da-aeebc8ae9416@wsu.com.br> Hello, Im creating an enviroment with TLS and WSS and i got permission denied when trying to start the wss and ws using port 80 or 443 Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]: ERROR:core:tcp_init_listener: bind(c, 0x7efca6dc1e5c, 16) on 168.194.68.29:443 : Permission denied Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]: ERROR:core:trans_init_all_listeners: failed to init listener [168.194.68.29], proto wss This occur why im running opensips as a user (opensips) and not as root, there is a setcap option that can allow this to happen, but, i think that this is some kind of mistake on my configuration, im right ? as we got daemons like nginx or apache that run as user but can have use of those ports , how can i do the same on opensips ? Thank you -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From david.villasmil.work at gmail.com Sun Jul 23 11:24:19 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 23 Jul 2017 15:24:19 +0000 Subject: [OpenSIPS-Users] Permission denied bo bind port 443 or 80 In-Reply-To: <05da99c0-9a96-89cd-14da-aeebc8ae9416@wsu.com.br> References: <05da99c0-9a96-89cd-14da-aeebc8ae9416@wsu.com.br> Message-ID: There's some permission restrictions that won't allow a non-root user to bind to those ports. Have a look at the OS documentation to figure out how to allow that... don't know what OS you're on On Sun, Jul 23, 2017 at 4:18 PM Mike Tesliuk wrote: > Hello, > > Im creating an enviroment with TLS and WSS and i got permission denied > when trying to start the wss and ws using port 80 or 443 > > > Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]: > ERROR:core:tcp_init_listener: bind(c, 0x7efca6dc1e5c, 16) on > 168.194.68.29:443 : Permission denied > Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]: > ERROR:core:trans_init_all_listeners: failed to init listener > [168.194.68.29], proto wss > > > This occur why im running opensips as a user (opensips) and not as root, > there is a setcap option that can allow this to happen, but, i think that > this is some kind of mistake on my configuration, im right ? as we got > daemons like nginx or apache that run as user but can have use of those > ports , how can i do the same on opensips ? > > > Thank you > > > -- > > > ​Atenciosamente, > [image: WSU TECNOLOGIA] > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | email > * > . > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Sun Jul 23 11:47:50 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sun, 23 Jul 2017 12:47:50 -0300 Subject: [OpenSIPS-Users] Permission denied bo bind port 443 or 80 In-Reply-To: References: <05da99c0-9a96-89cd-14da-aeebc8ae9416@wsu.com.br> Message-ID: <40156edf-d03f-6278-6032-ce6e38b30669@wsu.com.br> As i told i already have done that, the question is because one of the documentation have used the port 443 http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 but that is not a problems i was just trying to understood Em 23/07/17 12:24, David Villasmil escreveu: > There's some permission restrictions that won't allow a non-root user > to bind to those ports. Have a look at the OS documentation to figure > out how to allow that... don't know what OS you're on > On Sun, Jul 23, 2017 at 4:18 PM Mike Tesliuk > wrote: > > Hello, > > Im creating an enviroment with TLS and WSS and i got permission > denied when trying to start the wss and ws using port 80 or 443 > > > Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]: > ERROR:core:tcp_init_listener: bind(c, 0x7efca6dc1e5c, 16) on > 168.194.68.29:443 : Permission denied > Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]: > ERROR:core:trans_init_all_listeners: failed to init listener > [168.194.68.29], proto wss > > > This occur why im running opensips as a user (opensips) and not as > root, there is a setcap option that can allow this to happen, but, > i think that this is some kind of mistake on my configuration, im > right ? as we got daemons like nginx or apache that run as user > but can have use of those ports , how can i do the same on opensips ? > > > Thank you > > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | email > * > . > > _______________________________________________ > 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 -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Sun Jul 23 11:49:58 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sun, 23 Jul 2017 12:49:58 -0300 Subject: [OpenSIPS-Users] Error on WSS call with RTPEngine Message-ID: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> Hello Guys, On my tests with WSS call, im trying to go trough an asterisk and cameback to another extension, when this happen the opensips crash and show on log the message below Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]: CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch : calculated 1271, written 1237#012#012It seems you have hit a programming bug.#012Please help us make OpenSIPS better by reporting it at https://github.com/OpenSIPS/opensips/issues This happen on the answer of the call To you guys have any tip about this question ? Thank you. -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Sun Jul 23 16:38:03 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Sun, 23 Jul 2017 17:38:03 -0300 Subject: [OpenSIPS-Users] Error on WSS call with RTPEngine In-Reply-To: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> References: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> Message-ID: <850728b0-d688-49c2-f80c-bec8e8fc278d@wsu.com.br> The same error when i try a ipv6 -> ipv4 using rtpproxy Em 23/07/17 12:49, Mike Tesliuk escreveu: > > Hello Guys, > > > On my tests with WSS call, im trying to go trough an asterisk and > cameback to another extension, when this happen the opensips crash and > show on log the message below > > > Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]: > CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch : > calculated 1271, written 1237#012#012It seems you have hit a > programming bug.#012Please help us make OpenSIPS better by reporting > it at https://github.com/OpenSIPS/opensips/issues > > > This happen on the answer of the call > > > To you guys have any tip about this question ? > > > Thank you. > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | email > * > . > -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From razvan at opensips.org Mon Jul 24 03:47:36 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 24 Jul 2017 10:47:36 +0300 Subject: [OpenSIPS-Users] Error on WSS call with RTPEngine In-Reply-To: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> References: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> Message-ID: <61558305-8788-76f1-114b-68502bc6383e@opensips.org> Hi, Mike! Can you send us the debugging log for this error? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/23/2017 06:49 PM, Mike Tesliuk wrote: > > Hello Guys, > > > On my tests with WSS call, im trying to go trough an asterisk and > cameback to another extension, when this happen the opensips crash and > show on log the message below > > > Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]: > CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch : > calculated 1271, written 1237#012#012It seems you have hit a > programming bug.#012Please help us make OpenSIPS better by reporting > it at https://github.com/OpenSIPS/opensips/issues > > > This happen on the answer of the call > > > To you guys have any tip about this question ? > > > Thank you. > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Jul 24 03:52:27 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 24 Jul 2017 10:52:27 +0300 Subject: [OpenSIPS-Users] support for cloudamqp In-Reply-To: References: <5bf131f8-337a-5ef3-b9f7-d8c4a417da33@opensips.org> Message-ID: Hi, Tito! I have attached again the snippet for my tests[1]. Note that event_rabbitmq and rabbitmq are two different modules: event_rabbitmq does not support vhosts, but the newly rabbitmq module does. Check the snippet[1]. [1] https://pastebin.com/tsq2wmf1 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/19/2017 12:51 AM, Tito Cumpen wrote: > Hey Razvan, > > > Can you share the syntax for 2.3 again ? Although it looks like the > documentation claims: > > > 2.3. > > What is the vhost used by the AMQP server? > > Currently, the only vhost supported is '/'. > > > Thanks, > Tito > > On Mon, Apr 24, 2017 at 10:09 AM, Răzvan Crainea > wrote: > > Hi, Tito! > > Here's[1] the snippet I used for my tests. > > [1] https://pastebin.com/rRZhDXTX > > Best regards, > > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 04/24/2017 10:35 AM, Tito Cumpen wrote: >> Razvan, >> >> How do specify the virtualhost in 2.3? can you send me a sample >> of the syntax ? >> >> On Apr 24, 2017 3:30 AM, "Răzvan Crainea" > > wrote: >> >> Hi, Tito! >> >> You are right, this is not possible with the event_rabbitmq >> module, because there is no way to specify the virtual host >> in the url. >> But you can use the latest rabbitmq module in OpenSIPS 2.3. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> >> On 04/21/2017 09:53 PM, Tito Cumpen wrote: >>> Hey Razvan, >>> >>> I tried the following on Opensips: >>> >>> subscribe_event("E_ACC_EVENT","rabbitmq:vhostuser:pw at host.cloudamqp.com/vhostuser/queuename >>> "); >>> >>> >>> their URL string is in this format : >>> >>> amqp://vhostusert:pw at host.cloudamqp.com/vhostuser >>> >>> >>> >>> When Opensips tries to connect to this queue it sends >>> >>> / as the argument for openvhost >>> >>> Cloudamqp replies with >>> >>> >>> NOT_ALLOWED - access to vhost '/' refused for user 'vhostuser' >>> >>> >>> Is my opensips syntax incorrect ? or is this a bug ? >>> >>> Also I am not sure what you mean by immediate. >>> >>> Thanks, >>> >>> Tito >>> >>> >>> >>> On Fri, Apr 21, 2017 at 5:56 AM, Răzvan Crainea >>> > wrote: >>> >>> Hi, Tito! >>> >>> I've just made a free cloudamqp account for testing and >>> used the new rabbitmq module to send a message in the >>> queue. The message was not published initially due to >>> the fact that I was using the "immediate" flag, (I was >>> receiving NOT_IMPLEMENTED, probably because the free >>> account lacks some features), but after I removed the >>> setting the message was successfully delivered in the queue. >>> I didn't test with the event_rabbitmq module though, but >>> I imagine it will work too, since both modules >>> (event_rabbitmq and rabbitmq) use the same AMQP library. >>> Can you detail a bit how you tested so we can find where >>> the problem is on your setup? >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutions >>> www.opensips-solutions.com >>> >>> >>> On 04/20/2017 09:18 PM, Tito Cumpen wrote: >>>> Hello, >>>> >>>> I was wondering if there was any intention to update >>>> the amqp support to allow opensips to post events to >>>> cloudamqp(https://www.cloudamqp.com >>>> ). When I try to connect >>>> cloudamqp it fails because it is outdated. >>>> >>>> Thanks, >>>> Tito >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> _______________________________________________ Users >>> mailing list Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ Users mailing >> list Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goley_ev_sp at mail.ru Mon Jul 24 04:28:53 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Mon, 24 Jul 2017 01:28:53 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> Message-ID: <1500884933063-7608080.post@n2.nabble.com> Hi, Liviu Chircu I found out the memory stick when Opensips fell. I measured the average memory usage value by one call of 0.064MB. Then, at startup, Opensips indicated using 32M of shared memory. Using the values from the statistics (shmem: free_size 29996944 B) translated it into MB (28,6073). The result was divided by 0,064 resulting in about 447 calls. And then the magic begins =). After 448 calls, OpenSIPs crashes. Similarly, it happens when you specify a memory of more than 32MB. I use the Opencips on CentOS 7. I did not observe such a problem on Ubuntu 16.04 TLS. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608080.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From liviu at opensips.org Mon Jul 24 06:56:00 2017 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 24 Jul 2017 13:56:00 +0300 Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1500884933063-7608080.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> Message-ID: Please post some crash logs, a backtrace, or anything that might be useful in making progress with this. For example, you could do 200 calls, print out the memory status with SIGUSR1 [1] and post it here. Or you could post some pre-crash logs. Or post-crash. [1]: http://www.opensips.org/Documentation/TroubleShooting-OutOfMem Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 24.07.2017 11:28, Evgeniy G. via Users wrote: > Hi, Liviu Chircu > I found out the memory stick when Opensips fell. I measured the average > memory usage value by one call of 0.064MB. Then, at startup, Opensips > indicated using 32M of shared memory. Using the values from the statistics > (shmem: free_size 29996944 B) translated it into MB (28,6073). The result > was divided by 0,064 resulting in about 447 calls. And then the magic begins > =). > After 448 calls, OpenSIPs crashes. Similarly, it happens when you specify a > memory of more than 32MB. > I use the Opencips on CentOS 7. I did not observe such a problem on Ubuntu > 16.04 TLS. > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608080.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From goley_ev_sp at mail.ru Mon Jul 24 07:37:08 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Mon, 24 Jul 2017 04:37:08 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1500884933063-7608080.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> Message-ID: <1500896228657-7608082.post@n2.nabble.com> Today, I checked the results at CENOS 6. The first graph shows the use of memory on the CentOS 6, the second on CentOS 7. The second graph shows that the memory at the end of calls is not released. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608082.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From goley_ev_sp at mail.ru Mon Jul 24 07:38:01 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Mon, 24 Jul 2017 04:38:01 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1500896228657-7608082.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> Message-ID: <1500896281600-7608083.post@n2.nabble.com> -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608083.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From spanda at 3clogic.com Mon Jul 24 08:24:29 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 24 Jul 2017 17:54:29 +0530 Subject: [OpenSIPS-Users] I have a query regarding loadbalance of opensips . Message-ID: Hi All , I wanted to use a single DB against 2 or more registrar server . Below is my requirement : Opensips1 (x.x.x.x) -> DB Opensips2 (x.x.x.y) -> DB DB is in different machine and both opensips servers are accessing the same DB . Opensips is running on Db only mode . db_mode: 3 . I had mapped a domain in route53 ex. " loadbalance.i3clogic.com " and against this domain I have added both the IPs of opensips1 and 2 as SRV . I have added this domain in my opensips config file as aslias in both the config files . what my client do is , it does SRV query in the domain and resolve 1 Ip at a time and send requests to that . example : A-> SRV query( resolve Ip opensips1) -> sent Register Opensips1 -> store in DB This can change in ttl expire or in re-login of A . A is making TCP connection with opensips . when an Invite comes for A to opensips2 , that send 477 send fail . B -> Inv (TCP connection for A ) Opensips2 Opensips2 replys with 477 send fail to B . The reason behind this is A is registered through Opensips1 . This is what I want and the problem I am facing . My questing is how can I solve this . I dont want to add any proxy in between client and opensips1/2 . Is there any way this will work for me ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Mon Jul 24 08:44:24 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Mon, 24 Jul 2017 09:44:24 -0300 Subject: [OpenSIPS-Users] Error on WSS call with RTPEngine In-Reply-To: <61558305-8788-76f1-114b-68502bc6383e@opensips.org> References: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> <61558305-8788-76f1-114b-68502bc6383e@opensips.org> Message-ID: <7b5186ba-75d7-e566-92b1-1d6df7209e37@wsu.com.br> Hello Razvan , i have a debug log here: http://sip.wsu.com.br/pub/test1.txt This was a try of call fro 101500000393 to 1017 , the call need to go to an asterisk, and reach the extension on another opensips, this work well with two simple softphone, but got the error when i try to use the rtpengine the same message as i send on another email happen when i try from ipv4 to ipv6 using rtpproxy Em 24/07/17 04:47, Răzvan Crainea escreveu: > Hi, Mike! > > Can you send us the debugging log for this error? > > Best regards, > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > On 07/23/2017 06:49 PM, Mike Tesliuk wrote: >> >> Hello Guys, >> >> >> On my tests with WSS call, im trying to go trough an asterisk and >> cameback to another extension, when this happen the opensips crash >> and show on log the message below >> >> >> Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]: >> CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch : >> calculated 1271, written 1237#012#012It seems you have hit a >> programming bug.#012Please help us make OpenSIPS better by reporting >> it at https://github.com/OpenSIPS/opensips/issues >> >> >> This happen on the answer of the call >> >> >> To you guys have any tip about this question ? >> >> >> Thank you. >> >> -- >> >> >> ​Atenciosamente, >> WSU TECNOLOGIA >> Mike Tesliuk >> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >> 12387 SW 125th ter, Miami, Florida 33186 - USA >> tel +55 (41) 3941.0650 +1 (786) 719.6253 >> *website | mapa >> | email >> * >> . >> >> >> >> _______________________________________________ >> 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 -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From liviu at opensips.org Mon Jul 24 09:27:23 2017 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 24 Jul 2017 16:27:23 +0300 Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1500896281600-7608083.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> Message-ID: <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> Just to make sure I got it right: Although the graph says "memory usage", it actually graphs "free memory", correct? If yes, could you please redo the test, properly wait for all calls to finish, and then generate a memory map with "kill -SIGUSR1 "? You can find out the attendant's PID with "opensipsctl fifo ps". These logs should give us a strong hint as to what happens with the shared memory on that CentOS 7 box. Also, you may send any relevant out-of-memory error logs to liviu at opensips.org, if privacy is a concern. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 24.07.2017 14:38, Evgeniy G. via Users wrote: > > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608083.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Mon Jul 24 10:32:41 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 24 Jul 2017 17:32:41 +0300 Subject: [OpenSIPS-Users] Error on WSS call with RTPEngine In-Reply-To: <7b5186ba-75d7-e566-92b1-1d6df7209e37@wsu.com.br> References: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> <61558305-8788-76f1-114b-68502bc6383e@opensips.org> <7b5186ba-75d7-e566-92b1-1d6df7209e37@wsu.com.br> Message-ID: <988828e0-bfd2-fd13-4e26-3c3e4dc1af3a@opensips.org> Hi, Mike! The debug log you have posted does not contain any CRITICAL or ERROR message in it. Was it done with a different scenario? If so, can you post the debug logs from the scenario that generates the CRITICAL message? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/24/2017 03:44 PM, Mike Tesliuk wrote: > > Hello Razvan , > > > i have a debug log here: > > > http://sip.wsu.com.br/pub/test1.txt > > > This was a try of call fro 101500000393 to 1017 , the call need to go > to an asterisk, and reach the extension on another opensips, this work > well with two simple softphone, but got the error when i try to use > the rtpengine > > > the same message as i send on another email happen when i try from > ipv4 to ipv6 using rtpproxy > > > > > Em 24/07/17 04:47, Răzvan Crainea escreveu: >> Hi, Mike! >> >> Can you send us the debugging log for this error? >> >> Best regards, >> Răzvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> On 07/23/2017 06:49 PM, Mike Tesliuk wrote: >>> >>> Hello Guys, >>> >>> >>> On my tests with WSS call, im trying to go trough an asterisk and >>> cameback to another extension, when this happen the opensips crash >>> and show on log the message below >>> >>> >>> Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]: >>> CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch : >>> calculated 1271, written 1237#012#012It seems you have hit a >>> programming bug.#012Please help us make OpenSIPS better by reporting >>> it at https://github.com/OpenSIPS/opensips/issues >>> >>> >>> This happen on the answer of the call >>> >>> >>> To you guys have any tip about this question ? >>> >>> >>> Thank you. >>> >>> -- >>> >>> >>> ​Atenciosamente, >>> WSU TECNOLOGIA >>> Mike Tesliuk >>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >>> 12387 SW 125th ter, Miami, Florida 33186 - USA >>> tel +55 (41) 3941.0650 +1 (786) 719.6253 >>> *website | mapa >>> | >>> email * >>> . >>> >>> >>> >>> _______________________________________________ >>> 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 > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goley_ev_sp at mail.ru Mon Jul 24 10:46:23 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Mon, 24 Jul 2017 07:46:23 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> Message-ID: <1500907583776-7608091.post@n2.nabble.com> Liviu Chircu wrote > Just to make sure I got it right: Although the graph says "memory > usage", it actually graphs "free memory", correct? Yes > If yes, could you please redo the test, properly wait for all calls to > finish, and then generate a memory map with "kill -SIGUSR1 > > "? Ok, I'll do it and let you know the results. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608091.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From mike at wsu.com.br Mon Jul 24 12:35:56 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Mon, 24 Jul 2017 13:35:56 -0300 Subject: [OpenSIPS-Users] Error on WSS call with RTPEngine In-Reply-To: <988828e0-bfd2-fd13-4e26-3c3e4dc1af3a@opensips.org> References: <1beb30c5-0a7c-c8e1-168b-e6757420dc29@wsu.com.br> <61558305-8788-76f1-114b-68502bc6383e@opensips.org> <7b5186ba-75d7-e566-92b1-1d6df7209e37@wsu.com.br> <988828e0-bfd2-fd13-4e26-3c3e4dc1af3a@opensips.org> Message-ID: My bad Razvan, The sceneario was different, but is the same now http://sip.wsu.com.br/pub/test2.txt Em 24/07/17 11:32, Răzvan Crainea escreveu: > Hi, Mike! > > The debug log you have posted does not contain any CRITICAL or ERROR > message in it. Was it done with a different scenario? If so, can you > post the debug logs from the scenario that generates the CRITICAL message? > > Best regards, > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > On 07/24/2017 03:44 PM, Mike Tesliuk wrote: >> >> Hello Razvan , >> >> >> i have a debug log here: >> >> >> http://sip.wsu.com.br/pub/test1.txt >> >> >> This was a try of call fro 101500000393 to 1017 , the call need to go >> to an asterisk, and reach the extension on another opensips, this >> work well with two simple softphone, but got the error when i try to >> use the rtpengine >> >> >> the same message as i send on another email happen when i try from >> ipv4 to ipv6 using rtpproxy >> >> >> >> >> Em 24/07/17 04:47, Răzvan Crainea escreveu: >>> Hi, Mike! >>> >>> Can you send us the debugging log for this error? >>> >>> Best regards, >>> Răzvan Crainea >>> OpenSIPS Solutions >>> www.opensips-solutions.com >>> On 07/23/2017 06:49 PM, Mike Tesliuk wrote: >>>> >>>> Hello Guys, >>>> >>>> >>>> On my tests with WSS call, im trying to go trough an asterisk and >>>> cameback to another extension, when this happen the opensips crash >>>> and show on log the message below >>>> >>>> >>>> Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]: >>>> CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch : >>>> calculated 1271, written 1237#012#012It seems you have hit a >>>> programming bug.#012Please help us make OpenSIPS better by >>>> reporting it at https://github.com/OpenSIPS/opensips/issues >>>> >>>> >>>> This happen on the answer of the call >>>> >>>> >>>> To you guys have any tip about this question ? >>>> >>>> >>>> Thank you. >>>> >>>> -- >>>> >>>> >>>> ​Atenciosamente, >>>> WSU TECNOLOGIA >>>> Mike Tesliuk >>>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >>>> 12387 SW 125th ter, Miami, Florida 33186 - USA >>>> tel +55 (41) 3941.0650 +1 (786) 719.6253 >>>> *website | mapa >>>> | email >>>> * >>>> . >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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 >> >> -- >> >> >> ​Atenciosamente, >> WSU TECNOLOGIA >> Mike Tesliuk >> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >> 12387 SW 125th ter, Miami, Florida 33186 - USA >> tel +55 (41) 3941.0650 +1 (786) 719.6253 >> *website | mapa >> | email >> * >> . >> >> >> >> _______________________________________________ >> 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 -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From mike at wsu.com.br Mon Jul 24 16:21:38 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Mon, 24 Jul 2017 17:21:38 -0300 Subject: [OpenSIPS-Users] pv_proxy_authorize cache Message-ID: <12e7b238-9fb3-ba84-c97e-bbb7780b02f0@wsu.com.br> Hello there, Im trying to implement a proxy_authorize using cache without success. is that possible to perform the cache using proxy_authorize ? on the example [1] i see the www_challenge() no proxy_challenge, is that correct ? on my test im doing this (below): modparam("auth","username_spec", "$avp(usuario)") modparam("auth","password_spec", "$avp(senha)") modparam("auth_db", "load_credentials", "$avp(senha)=password") $avp(usuario) = $fU; if(cache_fetch("redis","passwd_$fU",$avp(senha))) { if(!pv_proxy_authorize("")){ proxy_challenge("","0"); exit; } }else{ if(!proxy_authorize("")){ proxy_challenge("","0"); exit; } cache_store("redis","passwd_$fU","$avp(senha)",3600); } But with this rule i do not get the user authenticated. what im doing wrong ? :) Thanks in advice [1] - https://www.opensips.org/Documentation/Tutorials-MemoryCaching#toc3 -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From pat at voxtelesys.com Mon Jul 24 20:32:20 2017 From: pat at voxtelesys.com (Pat Burke) Date: Mon, 24 Jul 2017 19:32:20 -0500 Subject: [OpenSIPS-Users] Registered trunks Message-ID: <8d4725b82a890a5fc9af8438f2498bdb@voxtelesys.com> Hello, As a SIP Provider, we implementing the ability to provide SIP trunks to customers with a PBX or Dialer that require Registration.  With this in mind, the customer wants to be able to set the CallerID on at least on the basis of the devices connected tho them, but potentially on a per call basis. For the challenge-response to the non-Register methods, we have implemented the script as follows (seems to be a very standard way).  My question is for the case of the CallerID not being the same as the username/authorization name, how do we do this?  Because the "FROM" user is different from the authorized user, the db_check_from fails.  I don't believe all phone systems support P-Asserted-ID, so we can't really go that route.  So can we just remove the "db_check_from"?   What risk does that expose us to? if ( !(is_method("REGISTER")) ) { if (is_from_local("$var(reg_domain_attr)")) { # from Registered device $avp(callee_number_type) := "Registered"; # authenticate if from local subscriber # authenticate all initial non-REGISTER request that pretend to be # generated by local subscriber (domain from FROM URI is local) if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!db_check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } } Regards,Pat Burke -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Mon Jul 24 22:10:01 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Mon, 24 Jul 2017 23:10:01 -0300 Subject: [OpenSIPS-Users] Registered trunks In-Reply-To: <8d4725b82a890a5fc9af8438f2498bdb@voxtelesys.com> References: <8d4725b82a890a5fc9af8438f2498bdb@voxtelesys.com> Message-ID: <1617ac53-eb8d-6db5-dae5-7f19a03ab5f7@wsu.com.br> Hello Pat, I think that you can ask them to set the From Name as the callerid so you can use transformation to take de information [1] Example: xlog("FROM NAME: $(hdr(From){nameaddr.name})"); xlog("FROM USER: $fU"); Result (on log): l 24 22:06:51 opensipsHomolog2 /usr/local/sbin/opensips[7960]: FROM NAME: "1016" Jul 24 22:06:51 opensipsHomolog2 /usr/local/sbin/opensips[7960]: FROM USER: 101600000393 After the authentication you can use the uac_replace_from[2] and change the callerid that you send you carriers. [1] - https://www.opensips.org/Documentation/Script-Tran-2-2 [2] - http://www.opensips.org/html/docs/modules/devel/uac.html#idp5265536 Em 24/07/17 21:32, Pat Burke escreveu: > Hello, > > As a SIP Provider, we implementing the ability to provide SIP trunks > to customers with a PBX or Dialer that require Registration. With > this in mind, > the customer wants to be able to set the CallerID on at least on the > basis of the devices connected tho them, but potentially on a per call > basis. > > For the challenge-response to the non-Register methods, we have > implemented the script as follows (seems to be a very standard way). > My question is > for the case of the CallerID not being the same as the > username/authorization name, how do we do this? Because the "FROM" > user is different from the > authorized user, the db_check_from fails. I don't believe all phone > systems support P-Asserted-ID, so we can't really go that route. So > can we just remove > the "db_check_from"? What risk does that expose us to? > > if ( !(is_method("REGISTER")) ) { > if (is_from_local("$var(reg_domain_attr)")) { # from Registered device > $avp(callee_number_type) := "Registered"; > > # authenticate if from local subscriber > # authenticate all initial non-REGISTER request that pretend to be > # generated by local subscriber (domain from FROM URI is local) > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > exit; > } > > if (!db_check_from()) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > > consume_credentials(); > # caller authenticated > } > } > Regards, > *Pat Burke* > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From bogdan at opensips.org Tue Jul 25 05:36:20 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 12:36:20 +0300 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: References: <20170720184547.GA11475@tlaquepaque.localdomain> <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> Message-ID: Hi Daniel, There are 3 types of accountings in OpenSIPS - per message, per transaction, per dialog. For the per message, it is clear :) . When doing per-transaction accounting, the ACC record is written when the transaction is completed with a final response (>=200) on the UAS side (towards caller). For the dialog based accounting, the time reference (for ending the call) is the reception of BYE request; still the CDR is written on the BYE final reply (as OpenSIPS allows you to collect CDR info from the BYE replies too). Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/20/2017 10:49 PM, Daniel Zanutti wrote: > Hi Alex > > I'm having a billing problem from receiving BYE to 200 OK is taking > more than 500ms. If BYE is accounted when it's received, great! > > Are you absolutely sure it works this way? > > Thanks > > On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov > > wrote: > > My understanding is that this is a rather simple module without > sophisticated state componentry, and that it logs things > immediately as received, in the same iteration of message processing. > > -- Alex > > -- > Principal, Evariste Systems LLC (www.evaristesys.com > ) > > Sent from my Google Nexus. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 05:40:10 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 12:40:10 +0300 Subject: [OpenSIPS-Users] Uac registrant check In-Reply-To: <8D149D5C-B73E-42C5-BC2C-7BE157F94869@gmail.com> References: <8D149D5C-B73E-42C5-BC2C-7BE157F94869@gmail.com> Message-ID: <97c2f56b-32d6-78ff-d6b7-4875d4a53756@opensips.org> Hi Volkan, I think you are looking for the is_registered() function: http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#idp5648928 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/21/2017 12:34 PM, Volkan Oransoy wrote: > Hi all, > > I use uac_registrant to register to remote SIP systems and > registration phase seems ok. What I want to do is to receive calls > from these systems and before accepting calls to my box, I want to > check destination, if it is a valid record on my system. I found a > couple of replies on list archives and one of them suggests to lookup > agains AOR. But that doesn’t work right now. What is the most suitable > way to do this? > > > if ( check_source_address("1","$avp(trunk_attrs)") ) { > # request comes from trunks > setflag(IS_TRUNK); > } else if ( is_from_gw() ) { > # request comes from GWs > * } else if ( lookup("location","","$ru") ){* > * xlog("Location check for $ru passed.\n");* > } else { > xlog("Location check for $ru failed.\n"); > send_reply("403","Forbidden"); > exit; > } > > Regards, > > /Volkan > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pat at voxtelesys.com Tue Jul 25 06:33:07 2017 From: pat at voxtelesys.com (Pat Burke) Date: Tue, 25 Jul 2017 05:33:07 -0500 Subject: [OpenSIPS-Users] Registered trunks Message-ID: Mike, Thanks for the response.  The PBX that is interfacing with us does not have that option.  And because of the varied nature of PBX's, I would like to be as generic as possible.  So to me it really come back to what is the exposure to not performing the "db_check_from"? Regards, Pat Burke Hello Pat, I think that you can ask them to set the From Name as the callerid so you can use transformation to take de information [1] Example:                xlog("FROM NAME: $(hdr(From){nameaddr.name})");                xlog("FROM USER: $fU");       Result (on log): l 24 22:06:51 opensipsHomolog2 /usr/local/sbin/opensips[7960]: FROM NAME: "1016" Jul 24 22:06:51 opensipsHomolog2 /usr/local/sbin/opensips[7960]: FROM USER: 101600000393 After the authentication you can use the uac_replace_from[2] and change the callerid that you send you carriers. [1] - https://www.opensips.org/Documentation/Script-Tran-2-2 [2] - http://www.opensips.org/html/docs/modules/devel/uac.html#idp5265536 Em 24/07/17 21:32, Pat Burke escreveu: > Hello, > > As a SIP Provider, we implementing the ability to provide SIP trunks > to customers with a PBX or Dialer that require Registration.  With > this in mind, > the customer wants to be able to set the CallerID on at least on the > basis of the devices connected tho them, but potentially on a per call > basis. > > For the challenge-response to the non-Register methods, we have > implemented the script as follows (seems to be a very standard way). > My question is > for the case of the CallerID not being the same as the > username/authorization name, how do we do this?  Because the "FROM" > user is different from the > authorized user, the db_check_from fails.  I don't believe all phone > systems support P-Asserted-ID, so we can't really go that route.  So > can we just remove > the "db_check_from"?   What risk does that expose us to? > > if ( !(is_method("REGISTER")) ) { >   if (is_from_local("$var(reg_domain_attr)")) { # from Registered device >       $avp(callee_number_type) := "Registered"; > >       # authenticate if from local subscriber >       # authenticate all initial non-REGISTER request that pretend to be >       # generated by local subscriber (domain from FROM URI is local) >       if (!proxy_authorize("", "subscriber")) { >          proxy_challenge("", "0"); >          exit; >       } > >       if (!db_check_from()) { >          sl_send_reply("403","Forbidden auth ID"); >          exit; >       } > >       consume_credentials(); >       # caller authenticated >    } > } > Regards, > *Pat Burke* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 07:27:02 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 14:27:02 +0300 Subject: [OpenSIPS-Users] error on script generation In-Reply-To: <1a3db7da-2506-2f4e-1ecc-bc2530da05c6@wsu.com.br> References: <1a3db7da-2506-2f4e-1ecc-bc2530da05c6@wsu.com.br> Message-ID: <59a3daf0-aec7-4a7d-d497-842b2b00dc40@opensips.org> Hi Mike, Yes, you are right, I just fixed this is on 2.2, 2.3 and devel branches. Please doublecheck for me. Many thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/22/2017 05:18 PM, Mike Tesliuk wrote: > > Hello there, > > > im compiling the opensips 2.3 and the generated script with tls > generate the lines below: > > modparam("proto_tls","verify_cert", "1") > modparam("proto_tls","require_cert", "0") > modparam("proto_tls","tls_method", "TLSv1") > > modparam("proto_tls","certificate", > "/usr/local/etc/opensips/tls/user/user-cert.pem") > modparam("proto_tls","private_key", > "/usr/local/etc/opensips/tls/user/user-privkey.pem") > modparam("proto_tls","ca_list", > "/usr/local/etc/opensips/tls/user/user-calist.pem") > > > Those parameters are from tls_mgm not from proto_tls right ? on module > documentation are on tls_mgm section > > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 07:34:13 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 14:34:13 +0300 Subject: [OpenSIPS-Users] LoadBalancer and Clusterer In-Reply-To: <5350f828-51b6-aaea-0652-adf6c0b2d754@wsu.com.br> References: <5350f828-51b6-aaea-0652-adf6c0b2d754@wsu.com.br> Message-ID: <83a91750-5dbb-7cf5-b425-6e7d83cc94e7@opensips.org> Hello Mike, The profile replication (the thing you were already doing) and the destination status replication (newly added in 2.3) are different kind of replications, different data in different modules. The /s is about dialog profile replication and it is at the dialog module level. At the LB module level we have the replication of the status of the destination. The two kinds of replications are independent. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/23/2017 12:36 AM, Mike Tesliuk wrote: > > Hello, > > > On the past, i had implemented the dialog with cachedb and > load_balancer using a nosql, using the resource with the /s , as the > load_balancer have the parameter receive the replication, how i use that ? > > > without the /s when i create a call i do not se the resource being > used on node 2 , is that supposed to happen ? > > > Thank you. > > > PS: testing the 2.3 version > > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 07:48:40 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 14:48:40 +0300 Subject: [OpenSIPS-Users] I have a query regarding loadbalance of opensips . In-Reply-To: References: Message-ID: Hi Sasmita, There is an incompatibility between your opensips cluster design and your network topology. If the end device is TCP connected to one Node, and it is not able (due network constraints) to receive TCP connections from any other Node, it makes no sense to share the registration data between the OpenSIPS Nodes as only the Node that received the registration will be able to reach back the device (again, due network constraints) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/24/2017 03:24 PM, Sasmita Panda wrote: > Hi All , > I wanted to use a single DB against 2 or more registrar server . > > Below is my requirement : > > Opensips1 (x.x.x.x) -> DB > Opensips2 (x.x.x.y) -> DB > > DB is in different machine and both opensips servers are > accessing the same DB . Opensips is running on Db only mode . > db_mode: 3 . > > > I had mapped a domain in route53 ex. " loadbalance.i3clogic.com > " and against this domain I have > added both the IPs of opensips1 and 2 as SRV . > > I have added this domain in my opensips config file as aslias > in both the config files . > > what my client do is , it does SRV query in the domain and > resolve 1 Ip at a time and send requests to that . > > example : > A-> SRV query( resolve Ip opensips1) -> sent Register Opensips1 -> > store in DB > This can change in ttl expire or in re-login of A . A is making > TCP connection with opensips . > > when an Invite comes for A to opensips2 , that send 477 send fail . > > B -> Inv (TCP connection for A ) Opensips2 > > Opensips2 replys with 477 send fail to B . The reason behind > this is A is registered through Opensips1 . > > This is what I want and the problem I am facing . My questing is how > can I solve this . I dont want to add any proxy in between client and > opensips1/2 . Is there any way this will work for me ? > > > > > > > > > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Jul 25 07:53:45 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 25 Jul 2017 17:23:45 +0530 Subject: [OpenSIPS-Users] I have a query regarding loadbalance of opensips . In-Reply-To: References: Message-ID: Is there any way opensips nodes will be connected in TCP ? If what I am expecting is not possible then I will leave this . Then my question is whats the use of opensips cluster using same DB ? In which scenarion I can use this . I just want a cluster of opensips node sharing a single DB . How can I achieve this ? *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Jul 25, 2017 at 5:18 PM, Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > There is an incompatibility between your opensips cluster design and your > network topology. If the end device is TCP connected to one Node, and it is > not able (due network constraints) to receive TCP connections from any > other Node, it makes no sense to share the registration data between the > OpenSIPS Nodes as only the Node that received the registration will be able > to reach back the device (again, due network constraints) > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/24/2017 03:24 PM, Sasmita Panda wrote: > > Hi All , > > I wanted to use a single DB against 2 or more registrar server . > > Below is my requirement : > > Opensips1 (x.x.x.x) -> DB > Opensips2 (x.x.x.y) -> DB > > DB is in different machine and both opensips servers are accessing > the same DB . Opensips is running on Db only mode . db_mode: 3 . > > > I had mapped a domain in route53 ex. " loadbalance.i3clogic.com " and > against this domain I have added both the IPs of opensips1 and 2 as SRV . > > I have added this domain in my opensips config file as aslias in > both the config files . > > what my client do is , it does SRV query in the domain and resolve > 1 Ip at a time and send requests to that . > > example : > A-> SRV query( resolve Ip opensips1) -> sent Register Opensips1 -> store > in DB > This can change in ttl expire or in re-login of A . A is making TCP > connection with opensips . > > when an Invite comes for A to opensips2 , that send 477 send fail . > > B -> Inv (TCP connection for A ) Opensips2 > > Opensips2 replys with 477 send fail to B . The reason behind this is > A is registered through Opensips1 . > > This is what I want and the problem I am facing . My questing is how can I > solve this . I dont want to add any proxy in between client and opensips1/2 > . Is there any way this will work for me ? > > > > > > > > > > > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Jul 25 08:26:44 2017 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 25 Jul 2017 12:26:44 +0000 Subject: [OpenSIPS-Users] TLS_MGM: Multi-domain Client Certificate Validation Message-ID: Hi All, *Running: *opensips-2.3.1-1.el7.x86_64 / CentOS 7 I have been working with new TLS connection and have been having problems validating their client certificate. My OpenSIPs configuration works fine for other providers (i.e. Twilio) however I am seeing the following error messages reported while verify_cert is enabled: Jul 25 13:10:32 proxy.ex.com opensips[4881]: NOTICE:tls_mgm:verify_callback: depth = 0 Jul 25 13:10:32 proxy.ex.com opensips[4881]: NOTICE:tls_mgm:verify_callback: subject = /serialNumber=03379831/1.3.6.1.4.1.311.60.2.1.3=GB/businessCategory=Private Organization/C=GB/postalCode=SO16 7NP/L=Southampton/street=2 Venture Road/O=SIMWOOD ESMS LIMITED/OU=COMODO EV Multi-Domain SSL/CN=simwood.com Jul 25 13:10:32 proxy.ex.com opensips[4881]: NOTICE:tls_mgm:verify_callback: verify error:num=20:unable to get local issuer certificate Jul 25 13:10:32 proxy.ex.com opensips[4881]: NOTICE:tls_mgm:verify_callback: something wrong with the cert ... error code is 20 (check x509_vfy.h) Jul 25 13:10:32 proxy.ex.com opensips[4881]: NOTICE:tls_mgm:verify_callback: verify return:0 Jul 25 13:10:32 proxy.ex.com opensips[4881]: ERROR:proto_tls:tls_accept: New TLS connection from 178.22.140.34:34281 failed to accept Jul 25 13:10:32 proxy.ex.com opensips[4881]: ERROR:proto_tls:tls_print_errstack: TLS errstack: error:140890B2:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned Jul 25 13:10:32 proxy.ex.com opensips[4881]: ERROR:proto_tls:tls_read_req: failed to do pre-tls reading Part of my reason for resorting to the mailing list are old mailing list emails discussing that multi-domain certificates are not supported by OpenSIPs - is anyone able to confirm if this remains a problem? The openssl error code 20 is translated as X509_V_ERR_UNABLE_TO_GET_ISSUER_CERT_LOCALLY I have seen other reports that this issue may be related to an improperly chained certificate - does this sound at all likely? Any tips on debugging would be greatly appreciated, thanks. Callum -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 08:45:09 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 15:45:09 +0300 Subject: [OpenSIPS-Users] pv_proxy_authorize cache In-Reply-To: <12e7b238-9fb3-ba84-c97e-bbb7780b02f0@wsu.com.br> References: <12e7b238-9fb3-ba84-c97e-bbb7780b02f0@wsu.com.br> Message-ID: Hi Mike, depending on your SIP flow, you can use either www_ (if a REGISTER) or proxy_ (if a non-REGISTER) functions. In your script snip, you must populate both auth username and password before the calling the auth function. I do not see the $avp(usuario) set (probably with $fU ??) . Also, if the password is plain/text, be use you properly set the calculate_ha1 parameter. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/24/2017 11:21 PM, Mike Tesliuk wrote: > > Hello there, > > > Im trying to implement a proxy_authorize using cache without success. > > > is that possible to perform the cache using proxy_authorize ? on the > example [1] i see the www_challenge() no proxy_challenge, is that > correct ? > > > on my test im doing this (below): > > > modparam("auth","username_spec", "$avp(usuario)") > modparam("auth","password_spec", "$avp(senha)") > modparam("auth_db", "load_credentials", "$avp(senha)=password") > > > $avp(usuario) = $fU; > > if(cache_fetch("redis","passwd_$fU",$avp(senha))) { > if(!pv_proxy_authorize("")){ > proxy_challenge("","0"); > exit; > } > }else{ > if(!proxy_authorize("")){ > proxy_challenge("","0"); > exit; > } > > > cache_store("redis","passwd_$fU","$avp(senha)",3600); > > } > > But with this rule i do not get the user authenticated. > > what im doing wrong ? :) > > > Thanks in advice > > > > > [1] - https://www.opensips.org/Documentation/Tutorials-MemoryCaching#toc3 > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 08:59:32 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 15:59:32 +0300 Subject: [OpenSIPS-Users] Registered trunks In-Reply-To: <8d4725b82a890a5fc9af8438f2498bdb@voxtelesys.com> References: <8d4725b82a890a5fc9af8438f2498bdb@voxtelesys.com> Message-ID: <2ad662d3-739c-97d7-66ca-1cd31aeb0c5a@opensips.org> Hi Pat, I see 2 scenarios here (in both cases, the calledID and the auth user are completely different, they cannot be calculated one from the other): * if the caller sends in FROM the auth username and in PAI/RPID the CLI, you should be fine. * if the caller sends in FROM the CLI, then it should send in the auth answer the (as "username" attribute) the right auth username. The authentication should be successful, but you should check if the advertised CLI (in FROM) is allowed to use authentication username So, bottom line (I think you are on the second scenario), you have to keep the db_check_from(), re-configure the URI module to perform the check against a predefined set of mappings (auth username as per subscriber and SIP username as per FROM hdr) vi DB table "uri" - see the "db_table" and "use_uri_table" parameters: http://www.opensips.org/html/docs/modules/2.3.x/uri.html#use-uri-table Or using avp_db_query() you can make your one SQL query for checking (via custom table) if the auth username is allowed with a certain CLI (FROM username) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 03:32 AM, Pat Burke wrote: > Hello, > > As a SIP Provider, we implementing the ability to provide SIP trunks > to customers with a PBX or Dialer that require Registration. With > this in mind, > the customer wants to be able to set the CallerID on at least on the > basis of the devices connected tho them, but potentially on a per call > basis. > > For the challenge-response to the non-Register methods, we have > implemented the script as follows (seems to be a very standard way). > My question is > for the case of the CallerID not being the same as the > username/authorization name, how do we do this? Because the "FROM" > user is different from the > authorized user, the db_check_from fails. I don't believe all phone > systems support P-Asserted-ID, so we can't really go that route. So > can we just remove > the "db_check_from"? What risk does that expose us to? > > if ( !(is_method("REGISTER")) ) { > if (is_from_local("$var(reg_domain_attr)")) { # from Registered device > $avp(callee_number_type) := "Registered"; > > # authenticate if from local subscriber > # authenticate all initial non-REGISTER request that pretend to be > # generated by local subscriber (domain from FROM URI is local) > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > exit; > } > > if (!db_check_from()) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > > consume_credentials(); > # caller authenticated > } > } > Regards, > *Pat Burke* > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at wsu.com.br Tue Jul 25 09:01:18 2017 From: mike at wsu.com.br (Mike Tesliuk) Date: Tue, 25 Jul 2017 10:01:18 -0300 Subject: [OpenSIPS-Users] pv_proxy_authorize cache In-Reply-To: References: <12e7b238-9fb3-ba84-c97e-bbb7780b02f0@wsu.com.br> Message-ID: Hello Bogdan, the $avp(usuario) is populated with the $fU as you think, the password is using the calculate parameter , but my question was if the pv_proxy_authorize is supposed to work, because on the example was used the www_ , and on my tests do not work, but i will double check my configuration an try again. thank you very much Em 25/07/17 09:45, Bogdan-Andrei Iancu escreveu: > Hi Mike, > > depending on your SIP flow, you can use either www_ (if a REGISTER) or > proxy_ (if a non-REGISTER) functions. > > In your script snip, you must populate both auth username and password > before the calling the auth function. I do not see the $avp(usuario) > set (probably with $fU ??) . Also, if the password is plain/text, be > use you properly set the calculate_ha1 parameter. > > Best regards, > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > On 07/24/2017 11:21 PM, Mike Tesliuk wrote: >> >> Hello there, >> >> >> Im trying to implement a proxy_authorize using cache without success. >> >> >> is that possible to perform the cache using proxy_authorize ? on the >> example [1] i see the www_challenge() no proxy_challenge, is that >> correct ? >> >> >> on my test im doing this (below): >> >> >> modparam("auth","username_spec", "$avp(usuario)") >> modparam("auth","password_spec", "$avp(senha)") >> modparam("auth_db", "load_credentials", "$avp(senha)=password") >> >> >> $avp(usuario) = $fU; >> >> if(cache_fetch("redis","passwd_$fU",$avp(senha))) { >> if(!pv_proxy_authorize("")){ >> proxy_challenge("","0"); >> exit; >> } >> }else{ >> if(!proxy_authorize("")){ >> proxy_challenge("","0"); >> exit; >> } >> >> >> cache_store("redis","passwd_$fU","$avp(senha)",3600); >> >> } >> >> But with this rule i do not get the user authenticated. >> >> what im doing wrong ? :) >> >> >> Thanks in advice >> >> >> >> >> [1] - https://www.opensips.org/Documentation/Tutorials-MemoryCaching#toc3 >> >> -- >> >> >> ​Atenciosamente, >> WSU TECNOLOGIA >> Mike Tesliuk >> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >> 12387 SW 125th ter, Miami, Florida 33186 - USA >> tel +55 (41) 3941.0650 +1 (786) 719.6253 >> *website | mapa >> | email >> * >> . >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- ​Atenciosamente, WSU TECNOLOGIA Mike Tesliuk Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR 12387 SW 125th ter, Miami, Florida 33186 - USA tel +55 (41) 3941.0650 +1 (786) 719.6253 *website | mapa | email * . -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: OpenPGP digital signature URL: From bogdan at opensips.org Tue Jul 25 09:06:38 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 16:06:38 +0300 Subject: [OpenSIPS-Users] pv_proxy_authorize cache In-Reply-To: References: <12e7b238-9fb3-ba84-c97e-bbb7780b02f0@wsu.com.br> Message-ID: The difference between www_ and proxy_ comes to what SIP headers they are looking for (in the SIP message) for reading the auth answer (Proxy-Authorize versus Authorize). Otherwise, in terms of auth algorithm, they are the same. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 04:01 PM, Mike Tesliuk wrote: > > Hello Bogdan, the $avp(usuario) is populated with the $fU as you > think, the password is using the calculate parameter , but my question > was if the pv_proxy_authorize is supposed to work, because on the > example was used the www_ , and on my tests do not work, but i will > double check my configuration an try again. > > > thank you very much > > > Em 25/07/17 09:45, Bogdan-Andrei Iancu escreveu: >> Hi Mike, >> >> depending on your SIP flow, you can use either www_ (if a REGISTER) >> or proxy_ (if a non-REGISTER) functions. >> >> In your script snip, you must populate both auth username and >> password before the calling the auth function. I do not see the >> $avp(usuario) set (probably with $fU ??) . Also, if the password is >> plain/text, be use you properly set the calculate_ha1 parameter. >> >> Best regards, >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> On 07/24/2017 11:21 PM, Mike Tesliuk wrote: >>> >>> Hello there, >>> >>> >>> Im trying to implement a proxy_authorize using cache without success. >>> >>> >>> is that possible to perform the cache using proxy_authorize ? on the >>> example [1] i see the www_challenge() no proxy_challenge, is that >>> correct ? >>> >>> >>> on my test im doing this (below): >>> >>> >>> modparam("auth","username_spec", "$avp(usuario)") >>> modparam("auth","password_spec", "$avp(senha)") >>> modparam("auth_db", "load_credentials", "$avp(senha)=password") >>> >>> >>> $avp(usuario) = $fU; >>> >>> if(cache_fetch("redis","passwd_$fU",$avp(senha))) { >>> if(!pv_proxy_authorize("")){ >>> proxy_challenge("","0"); >>> exit; >>> } >>> }else{ >>> if(!proxy_authorize("")){ >>> proxy_challenge("","0"); >>> exit; >>> } >>> >>> >>> cache_store("redis","passwd_$fU","$avp(senha)",3600); >>> >>> } >>> >>> But with this rule i do not get the user authenticated. >>> >>> what im doing wrong ? :) >>> >>> >>> Thanks in advice >>> >>> >>> >>> >>> [1] - >>> https://www.opensips.org/Documentation/Tutorials-MemoryCaching#toc3 >>> >>> -- >>> >>> >>> ​Atenciosamente, >>> WSU TECNOLOGIA >>> Mike Tesliuk >>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR >>> 12387 SW 125th ter, Miami, Florida 33186 - USA >>> tel +55 (41) 3941.0650 +1 (786) 719.6253 >>> *website | mapa >>> | >>> email * >>> . >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -- > > > ​Atenciosamente, > WSU TECNOLOGIA > Mike Tesliuk > Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR > 12387 SW 125th ter, Miami, Florida 33186 - USA > tel +55 (41) 3941.0650 +1 (786) 719.6253 > *website | mapa > | > email * > . > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 09:08:25 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 16:08:25 +0300 Subject: [OpenSIPS-Users] I have a query regarding loadbalance of opensips . In-Reply-To: References: Message-ID: <4fc0158c-e2f7-a29b-d04d-de9ce8f468cb@opensips.org> Sasmita, OpenSIPS can open a new TCP connection towards an UAC IF: 1) the registered IP is public 2) the UAC is not behind a NAT. The 477 reply is generated by the inability of OpenSIPS to open a TCP connection - you can see some error messages into the logs too. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 02:53 PM, Sasmita Panda wrote: > Is there any way opensips nodes will be connected in TCP ? If what I > am expecting is not possible then I will leave this . > > Then my question is whats the use of opensips cluster using same DB ? > In which scenarion I can use this . I just want a cluster of opensips > node sharing a single DB . How can I achieve this ? > > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Tue, Jul 25, 2017 at 5:18 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Sasmita, > > There is an incompatibility between your opensips cluster design > and your network topology. If the end device is TCP connected to > one Node, and it is not able (due network constraints) to receive > TCP connections from any other Node, it makes no sense to share > the registration data between the OpenSIPS Nodes as only the Node > that received the registration will be able to reach back the > device (again, due network constraints) > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/24/2017 03:24 PM, Sasmita Panda wrote: >> Hi All , >> I wanted to use a single DB against 2 or more registrar >> server . >> >> Below is my requirement : >> >> Opensips1 (x.x.x.x) -> DB >> Opensips2 (x.x.x.y) -> DB >> >> DB is in different machine and both opensips servers are >> accessing the same DB . Opensips is running on Db only mode . >> db_mode: 3 . >> >> >> I had mapped a domain in route53 ex. " loadbalance.i3clogic.com >> " and against this domain I >> have added both the IPs of opensips1 and 2 as SRV . >> >> I have added this domain in my opensips config file as >> aslias in both the config files . >> >> what my client do is , it does SRV query in the domain and >> resolve 1 Ip at a time and send requests to that . >> >> example : >> A-> SRV query( resolve Ip opensips1) -> sent Register Opensips1 >> -> store in DB >> This can change in ttl expire or in re-login of A . A is >> making TCP connection with opensips . >> >> when an Invite comes for A to opensips2 , that send 477 send fail . >> >> B -> Inv (TCP connection for A ) Opensips2 >> >> Opensips2 replys with 477 send fail to B . The reason >> behind this is A is registered through Opensips1 . >> >> This is what I want and the problem I am facing . My questing is >> how can I solve this . I dont want to add any proxy in between >> client and opensips1/2 . Is there any way this will work for me ? >> >> >> >> >> >> >> >> >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Jul 25 09:16:01 2017 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 25 Jul 2017 18:46:01 +0530 Subject: [OpenSIPS-Users] I have a query regarding loadbalance of opensips . In-Reply-To: <4fc0158c-e2f7-a29b-d04d-de9ce8f468cb@opensips.org> References: <4fc0158c-e2f7-a29b-d04d-de9ce8f468cb@opensips.org> Message-ID: I got your point . I understand why 477 send failed message is coming . Lets I am not using TCP . I am using UDP . my question is how I will use a single DB behind a cluster of opensips . Is this possible in UDP . I dont think so . In UDP i may get "408 request timeout" message from opensips node . Lets say . I have a client and I have given a domain to that . against that domain there are 2 opensips node and both sharing same DB and running in db_only mode . My UAC is behind NAT off-course . According to you I cant use this scenario if my UAC is behind NAT . If opensips nodes and UAC can communicate internally inside a LAN then its possible . Please correct me if I am wrong . *Thanks & Regards* *Sasmita Panda* *Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Jul 25, 2017 at 6:38 PM, Bogdan-Andrei Iancu wrote: > Sasmita, > > OpenSIPS can open a new TCP connection towards an UAC IF: > 1) the registered IP is public > 2) the UAC is not behind a NAT. > > The 477 reply is generated by the inability of OpenSIPS to open a TCP > connection - you can see some error messages into the logs too. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/25/2017 02:53 PM, Sasmita Panda wrote: > > Is there any way opensips nodes will be connected in TCP ? If what I am > expecting is not possible then I will leave this . > > Then my question is whats the use of opensips cluster using same DB ? In > which scenarion I can use this . I just want a cluster of opensips node > sharing a single DB . How can I achieve this ? > > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > On Tue, Jul 25, 2017 at 5:18 PM, Bogdan-Andrei Iancu > wrote: > >> Hi Sasmita, >> >> There is an incompatibility between your opensips cluster design and your >> network topology. If the end device is TCP connected to one Node, and it is >> not able (due network constraints) to receive TCP connections from any >> other Node, it makes no sense to share the registration data between the >> OpenSIPS Nodes as only the Node that received the registration will be able >> to reach back the device (again, due network constraints) >> >> Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/24/2017 03:24 PM, Sasmita Panda wrote: >> >> Hi All , >> >> I wanted to use a single DB against 2 or more registrar server . >> >> Below is my requirement : >> >> Opensips1 (x.x.x.x) -> DB >> Opensips2 (x.x.x.y) -> DB >> >> DB is in different machine and both opensips servers are accessing >> the same DB . Opensips is running on Db only mode . db_mode: 3 . >> >> >> I had mapped a domain in route53 ex. " loadbalance.i3clogic.com " and >> against this domain I have added both the IPs of opensips1 and 2 as SRV . >> >> I have added this domain in my opensips config file as aslias in >> both the config files . >> >> what my client do is , it does SRV query in the domain and resolve >> 1 Ip at a time and send requests to that . >> >> example : >> A-> SRV query( resolve Ip opensips1) -> sent Register Opensips1 -> store >> in DB >> This can change in ttl expire or in re-login of A . A is making TCP >> connection with opensips . >> >> when an Invite comes for A to opensips2 , that send 477 send fail . >> >> B -> Inv (TCP connection for A ) Opensips2 >> >> Opensips2 replys with 477 send fail to B . The reason behind this >> is A is registered through Opensips1 . >> >> This is what I want and the problem I am facing . My questing is how can >> I solve this . I dont want to add any proxy in between client and >> opensips1/2 . Is there any way this will work for me ? >> >> >> >> >> >> >> >> >> >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 09:48:22 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 16:48:22 +0300 Subject: [OpenSIPS-Users] TLS_MGM: Multi-domain Client Certificate Validation In-Reply-To: References: Message-ID: <4762d2e8-8d66-0295-e645-96ddba4d2ec7@opensips.org> Hi Callum, The error may indicate the fact that the TLS client does not present a TLS certificate while connection to your OpenSIPS. This has nothing to do with the TLS multi domain, which anyhow is supported. As the test, you can create a separate TLS domain (server) bound to the IP of that TLS client, TLS domain having the require_certificate option turned off. Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 03:26 PM, Callum Guy wrote: > Hi All, > > *Running: *opensips-2.3.1-1.el7.x86_64 / CentOS 7 > > I have been working with new TLS connection and have been having > problems validating their client certificate. My OpenSIPs > configuration works fine for other providers (i.e. Twilio) however I > am seeing the following error messages reported while verify_cert is > enabled: > > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: depth = 0 > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: subject = > /serialNumber=03379831/1.3.6.1.4.1.311.60.2.1.3=GB/businessCategory=Private > Organization/C=GB/postalCode=SO16 7NP/L=Southampton/street=2 Venture > Road/O=SIMWOOD ESMS LIMITED/OU=COMODO EV Multi-Domain > SSL/CN=simwood.com > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: verify error:num=20:unable to get > local issuer certificate > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: something wrong with the cert ... > error code is 20 (check x509_vfy.h) > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: verify return:0 > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > ERROR:proto_tls:tls_accept: New TLS connection from > 178.22.140.34:34281 failed to accept > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > ERROR:proto_tls:tls_print_errstack: TLS errstack: error:140890B2:SSL > routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > ERROR:proto_tls:tls_read_req: failed to do pre-tls reading > > Part of my reason for resorting to the mailing list are old mailing > list emails discussing that multi-domain certificates are not > supported by OpenSIPs - is anyone able to confirm if this remains a > problem? > > The openssl error code 20 is translated as > X509_V_ERR_UNABLE_TO_GET_ISSUER_CERT_LOCALLY > > I have seen other reports that this issue may be related to an > improperly chained certificate - does this sound at all likely? > > Any tips on debugging would be greatly appreciated, thanks. > > Callum > -- > Callum Guy > Head of Information Security > X-on > > > *^0333 332 0000 | www.x-on.co.uk | > _**_^ > * > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 09:55:55 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 16:55:55 +0300 Subject: [OpenSIPS-Users] I have a query regarding loadbalance of opensips . In-Reply-To: References: <4fc0158c-e2f7-a29b-d04d-de9ce8f468cb@opensips.org> Message-ID: If you have the same kind of network constraints (UAC behind NAT), then in it not work for UDP either. If the UAC is nated, in 99% of the cases, it is able to be reached back only by the SIP node the UAC is registering with. So you have to re-think the internal design of your cluster. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 04:16 PM, Sasmita Panda wrote: > I got your point . I understand why 477 send failed message is coming . > > Lets I am not using TCP . I am using UDP . my question is how I will > use a single DB behind a cluster of opensips . Is this possible in UDP > . I dont think so . In UDP i may get "408 request timeout" message > from opensips node . > > Lets say . I have a client and I have given a domain to that . against > that domain there are 2 opensips node and both sharing same DB and > running in db_only mode . My UAC is behind NAT off-course . > > According to you I cant use this scenario if my UAC is behind NAT > . If opensips nodes and UAC can communicate internally inside a LAN > then its possible . > > Please correct me if I am wrong . > > */Thanks & Regards/* > /Sasmita Panda/ > /Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > On Tue, Jul 25, 2017 at 6:38 PM, Bogdan-Andrei Iancu > > wrote: > > Sasmita, > > OpenSIPS can open a new TCP connection towards an UAC IF: > 1) the registered IP is public > 2) the UAC is not behind a NAT. > > The 477 reply is generated by the inability of OpenSIPS to open a > TCP connection - you can see some error messages into the logs too. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/25/2017 02:53 PM, Sasmita Panda wrote: >> Is there any way opensips nodes will be connected in TCP ? If >> what I am expecting is not possible then I will leave this . >> >> Then my question is whats the use of opensips cluster using same >> DB ? In which scenarion I can use this . I just want a cluster of >> opensips node sharing a single DB . How can I achieve this ? >> >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> On Tue, Jul 25, 2017 at 5:18 PM, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Sasmita, >> >> There is an incompatibility between your opensips cluster >> design and your network topology. If the end device is TCP >> connected to one Node, and it is not able (due network >> constraints) to receive TCP connections from any other Node, >> it makes no sense to share the registration data between the >> OpenSIPS Nodes as only the Node that received the >> registration will be able to reach back the device (again, >> due network constraints) >> >> Best regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> >> On 07/24/2017 03:24 PM, Sasmita Panda wrote: >>> Hi All , >>> I wanted to use a single DB against 2 or more >>> registrar server . >>> >>> Below is my requirement : >>> >>> Opensips1 (x.x.x.x) -> DB >>> Opensips2 (x.x.x.y) -> DB >>> >>> DB is in different machine and both opensips servers >>> are accessing the same DB . Opensips is running on Db only >>> mode . db_mode: 3 . >>> >>> >>> I had mapped a domain in route53 ex. " >>> loadbalance.i3clogic.com " >>> and against this domain I have added both the IPs of >>> opensips1 and 2 as SRV . >>> >>> I have added this domain in my opensips config file >>> as aslias in both the config files . >>> >>> what my client do is , it does SRV query in the >>> domain and resolve 1 Ip at a time and send requests to that . >>> >>> example : >>> A-> SRV query( resolve Ip opensips1) -> sent Register >>> Opensips1 -> store in DB >>> This can change in ttl expire or in re-login of A . A >>> is making TCP connection with opensips . >>> >>> when an Invite comes for A to opensips2 , that send 477 >>> send fail . >>> >>> B -> Inv (TCP connection for A ) Opensips2 >>> >>> Opensips2 replys with 477 send fail to B . The reason >>> behind this is A is registered through Opensips1 . >>> >>> This is what I want and the problem I am facing . My >>> questing is how can I solve this . I dont want to add any >>> proxy in between client and opensips1/2 . Is there any way >>> this will work for me ? >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Jul 25 10:27:20 2017 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 25 Jul 2017 14:27:20 +0000 Subject: [OpenSIPS-Users] TLS_MGM: Multi-domain Client Certificate Validation In-Reply-To: <4762d2e8-8d66-0295-e645-96ddba4d2ec7@opensips.org> References: <4762d2e8-8d66-0295-e645-96ddba4d2ec7@opensips.org> Message-ID: Hi Bogdan, Thanks for your response, based on your advice I performed a full packet capture on the handshake and established that a certificate was indeed being presented. Following up on this I managed to establish that the problem was a missing intermediary CA in the certificate chain, specifically: https://support.comodo.com/index.php?/Knowledgebase/Article/View/975/108/intermediate-2-sha-2-comodo-rsa-extended-validation-secure-server-ca The error message presented by OpenSIPs was certainly misleading in this case. For others benefit the approach for installing a new CA is super simple: 1. create the file in /etc/pki/ca-trust/source/anchors (i.e. comodo-ca-rsa-ev-secure-server.pem) 2. run "update-ca-trust" with root privs Problem solved. Have a good day all! Callum On Tue, Jul 25, 2017 at 2:48 PM Bogdan-Andrei Iancu wrote: > Hi Callum, > > The error may indicate the fact that the TLS client does not present a TLS > certificate while connection to your OpenSIPS. This has nothing to do with > the TLS multi domain, which anyhow is supported. As the test, you can > create a separate TLS domain (server) bound to the IP of that TLS client, > TLS domain having the require_certificate option turned off. > > Best Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/25/2017 03:26 PM, Callum Guy wrote: > > Hi All, > > *Running: *opensips-2.3.1-1.el7.x86_64 / CentOS 7 > > I have been working with new TLS connection and have been having problems > validating their client certificate. My OpenSIPs configuration works fine > for other providers (i.e. Twilio) however I am seeing the following error > messages reported while verify_cert is enabled: > > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: depth = 0 > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: subject = > /serialNumber=03379831/1.3.6.1.4.1.311.60.2.1.3=GB/businessCategory=Private > Organization/C=GB/postalCode=SO16 7NP/L=Southampton/street=2 Venture > Road/O=SIMWOOD ESMS LIMITED/OU=COMODO EV Multi-Domain SSL/CN=simwood.com > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: verify error:num=20:unable to get local > issuer certificate > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: something wrong with the cert ... error > code is 20 (check x509_vfy.h) > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > NOTICE:tls_mgm:verify_callback: verify return:0 > Jul 25 13:10:32 proxy.ex.com opensips[4881]: ERROR:proto_tls:tls_accept: > New TLS connection from 178.22.140.34:34281 failed to accept > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > ERROR:proto_tls:tls_print_errstack: TLS errstack: error:140890B2:SSL > routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned > Jul 25 13:10:32 proxy.ex.com opensips[4881]: > ERROR:proto_tls:tls_read_req: failed to do pre-tls reading > > Part of my reason for resorting to the mailing list are old mailing list > emails discussing that multi-domain certificates are not supported by > OpenSIPs - is anyone able to confirm if this remains a problem? > > The openssl error code 20 is translated as > X509_V_ERR_UNABLE_TO_GET_ISSUER_CERT_LOCALLY > > I have seen other reports that this issue may be related to an improperly > chained certificate - does this sound at all likely? > > Any tips on debugging would be greatly appreciated, thanks. > > Callum > -- > Callum Guy > Head of Information Security > X-on > > > *0333 332 0000 | www.x-on.co.uk | ** > > * > 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 <+44%20333%20332%200000> 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 listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ravi.patel at ecosmob.com Tue Jul 25 10:34:29 2017 From: ravi.patel at ecosmob.com (Ravi Patel) Date: Tue, 25 Jul 2017 20:04:29 +0530 Subject: [OpenSIPS-Users] OpenSIPS reseting issue with $T_fr_inv_timeout while forwarding Message-ID: Hi Team, What is the right way to reset timers *$T_fr_inv_timeout* and *$T_fr_timeout* ?? I am using OpenSIPS-2.2 version The below scenario will help to understand issue, There are 4 SIP users, 1111,2222,3333,4444 What I want to achieve is: 1111 ---> 2222 (FORWARD ON NOANSWER) ---> 3333 (FORWARD ON NOANSWER) ---> 4444 *1st Test Case Scenario:* 1111 2222 (fr_inv_timeout 20 sec) 3333 (fr_inv_timeout 25 sec) 4444 (fr_inv_timeout 30 sec) when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (thats working proper as expexted) and forwards call to 3333 as per my configuration. so in --> 3333 : OpenSIPS generates CANCEL at *20 secs instead of 25 secs* and send 408 to 1111. and not processing the 2nd forwarding. *2nd Test Case Scenario:* 1111 2222 (fr_inv_timeout 20 sec) 3333 (fr_inv_timeout 15 sec) 4444 (fr_inv_timeout 30 sec) when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (that is working proper as expexted) and forwards call to 3333 as per my configuration. now --> 3333 : OpenSIPS generates CANCEL at 15 secs and forwards the call to 4444, Here OpenSIPS generates CANCEL *after 5 secs instead of 30 secs.* We set timeout by using $T_fr_inv_timeout. ------------ route[ring_timeout]{ xlog("L_INFO","------------------- RING_TIMEOUT ---------------\n"); if (!is_method("INVITE")) return; avp_db_load("$rU","$avp(ringtimeout)/usr_preferences"); if($avp(ringtimeout)!=null) { $T_fr_inv_timeout = NULL; xlog("L_INFO","$rU: Ring timeout : $avp(ringtimeout)"); $T_fr_inv_timeout =$(avp(ringtimeout){s.int}) ; xlog("L_INFO","$rU: Ring timeout is setted: [$T_fr_inv_timeout]"); } else { xlog("L_INFO","$rU: Ring timeout is NOT setted"); } } ------------------ >From both the scenarios what we found, it sticks to the first timeout of 2222,that is 20secs in our case. In first scenario it generates CANCEL on 3333 at 20 secs instead of 25 that is 2222's Timeout. In second scenario it generates CANCEL on 3333 at 15sec and on 4444 at 5 sec (15 + 5 = 20 sec) that is also 2222's timeout. Can I know the right method to set $T_fr_inv_timeout ? Let me know if any other information is needed. Thanks, Ravi -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 11:27:22 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 18:27:22 +0300 Subject: [OpenSIPS-Users] TLS_MGM: Multi-domain Client Certificate Validation In-Reply-To: References: <4762d2e8-8d66-0295-e645-96ddba4d2ec7@opensips.org> Message-ID: <19f852eb-91b6-7e93-467b-76a76f767f94@opensips.org> I have to admit that you have to "know how to read the SSL errors" in order to really understand the root problem :) . Now that you find the issue and if we look back at the error description "verify error:num=20:unable to get local issuer certificate", it make sense - SSL complains it did not find the comodo CA in order to validate the certificate presented by the TLS client (which was probably signed by Comodo). Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 05:27 PM, Callum Guy wrote: > Hi Bogdan, > > Thanks for your response, based on your advice I performed a full > packet capture on the handshake and established that a certificate was > indeed being presented. > > Following up on this I managed to establish that the problem was a > missing intermediary CA in the certificate chain, specifically: > > https://support.comodo.com/index.php?/Knowledgebase/Article/View/975/108/intermediate-2-sha-2-comodo-rsa-extended-validation-secure-server-ca > > The error message presented by OpenSIPs was certainly misleading in > this case. For others benefit the approach for installing a new CA is > super simple: > > 1. create the file in /etc/pki/ca-trust/source/anchors > (i.e. comodo-ca-rsa-ev-secure-server.pem) > 2. run "update-ca-trust" with root privs > > Problem solved. > > Have a good day all! > > Callum > > On Tue, Jul 25, 2017 at 2:48 PM Bogdan-Andrei Iancu > > wrote: > > Hi Callum, > > The error may indicate the fact that the TLS client does not > present a TLS certificate while connection to your OpenSIPS. This > has nothing to do with the TLS multi domain, which anyhow is > supported. As the test, you can create a separate TLS domain > (server) bound to the IP of that TLS client, TLS domain having the > require_certificate option turned off. > > Best Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/25/2017 03:26 PM, Callum Guy wrote: >> Hi All, >> >> *Running: *opensips-2.3.1-1.el7.x86_64 / CentOS 7 >> >> I have been working with new TLS connection and have been having >> problems validating their client certificate. My OpenSIPs >> configuration works fine for other providers (i.e. Twilio) >> however I am seeing the following error messages reported while >> verify_cert is enabled: >> >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: NOTICE:tls_mgm:verify_callback: depth = 0 >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: NOTICE:tls_mgm:verify_callback: subject = >> /serialNumber=03379831/1.3.6.1.4.1.311.60.2.1.3=GB/businessCategory=Private >> Organization/C=GB/postalCode=SO16 7NP/L=Southampton/street=2 >> Venture Road/O=SIMWOOD ESMS LIMITED/OU=COMODO EV Multi-Domain >> SSL/CN=simwood.com >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: NOTICE:tls_mgm:verify_callback: verify >> error:num=20:unable to get local issuer certificate >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: NOTICE:tls_mgm:verify_callback: something wrong >> with the cert ... error code is 20 (check x509_vfy.h) >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: NOTICE:tls_mgm:verify_callback: verify return:0 >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: ERROR:proto_tls:tls_accept: New TLS connection >> from 178.22.140.34:34281 failed to >> accept >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: ERROR:proto_tls:tls_print_errstack: TLS errstack: >> error:140890B2:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:no >> certificate returned >> Jul 25 13:10:32 proxy.ex.com >> opensips[4881]: ERROR:proto_tls:tls_read_req: failed to do >> pre-tls reading >> >> Part of my reason for resorting to the mailing list are old >> mailing list emails discussing that multi-domain certificates are >> not supported by OpenSIPs - is anyone able to confirm if this >> remains a problem? >> >> The openssl error code 20 is translated as >> X509_V_ERR_UNABLE_TO_GET_ISSUER_CERT_LOCALLY >> >> I have seen other reports that this issue may be related to an >> improperly chained certificate - does this sound at all likely? >> >> Any tips on debugging would be greatly appreciated, thanks. >> >> Callum >> -- >> Callum Guy >> Head of Information Security >> X-on >> >> >> *^0333 332 0000 | www.x-on.co.uk | >> _**_^ >> * >> 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 > > -- > Callum Guy > Head of Information Security > X-on > > > *^0333 332 0000 | www.x-on.co.uk | > _**_^ > * > 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. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 25 11:37:33 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 25 Jul 2017 18:37:33 +0300 Subject: [OpenSIPS-Users] OpenSIPS reseting issue with $T_fr_inv_timeout while forwarding In-Reply-To: References: Message-ID: <807e815d-060b-598d-89bb-511bf2a0b70c@opensips.org> Hi Ravi, Before each t_rely() you have to set the your custom $T_fr_inv_timeout and $T_fr_timeout, otherwise the default values will be used. As you have a serial forking scenario, you do a new t_relay() at each step. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 05:34 PM, Ravi Patel wrote: > Hi Team, > > What is the right way to reset timers *$T_fr_inv_timeout* and > *$T_fr_timeout* ?? > > I am using OpenSIPS-2.2 version > The below scenario will help to understand issue, > > There are 4 SIP users, > 1111,2222,3333,4444 > > What I want to achieve is: > 1111 ---> 2222 (FORWARD ON NOANSWER) ---> 3333 (FORWARD ON NOANSWER) > ---> 4444 > > *1st Test Case Scenario:* > > 1111 > 2222 (fr_inv_timeout 20 sec) > 3333 (fr_inv_timeout 25 sec) > 4444 (fr_inv_timeout 30 sec) > > > when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (thats > working proper as expexted) and forwards call to 3333 as per my > configuration. > so in --> 3333 : OpenSIPS generates CANCEL at *20 secs instead of 25 > secs* and send 408 to 1111. and not processing the 2nd forwarding. > > *2nd Test Case Scenario:* > 1111 > 2222 (fr_inv_timeout 20 sec) > 3333 (fr_inv_timeout 15 sec) > 4444 (fr_inv_timeout 30 sec) > > when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (that is > working proper as expexted) and forwards call to 3333 as per my > configuration. > now --> 3333 : OpenSIPS generates CANCEL at 15 secs and forwards the > call to 4444, Here OpenSIPS generates CANCEL *after 5 secs instead of > 30 secs.* > > > We set timeout by using $T_fr_inv_timeout. > ------------ > route[ring_timeout]{ > xlog("L_INFO","------------------- RING_TIMEOUT > ---------------\n"); > if (!is_method("INVITE")) > return; > avp_db_load("$rU","$avp(ringtimeout)/usr_preferences"); > if($avp(ringtimeout)!=null) > { > $T_fr_inv_timeout = NULL; > xlog("L_INFO","$rU: Ring timeout : > $avp(ringtimeout)"); > $T_fr_inv_timeout =$(avp(ringtimeout){s.int > }) ; > xlog("L_INFO","$rU: Ring timeout is setted: > [$T_fr_inv_timeout]"); > } > else > { > xlog("L_INFO","$rU: Ring timeout is NOT setted"); > } > } > ------------------ > > From both the scenarios what we found, it sticks to the first timeout > of 2222,that is 20secs in our case. > In first scenario it generates CANCEL on 3333 at 20 secs instead of 25 > that is 2222's Timeout. > In second scenario it generates CANCEL on 3333 at 15sec and on 4444 at > 5 sec (15 + 5 = 20 sec) that is also 2222's timeout. > > > Can I know the right method to set $T_fr_inv_timeout ? > > Let me know if any other information is needed. > > > Thanks, > Ravi > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Jul 25 11:58:28 2017 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 25 Jul 2017 15:58:28 +0000 Subject: [OpenSIPS-Users] TLS_MGM: Multi-domain Client Certificate Validation In-Reply-To: <19f852eb-91b6-7e93-467b-76a76f767f94@opensips.org> References: <4762d2e8-8d66-0295-e645-96ddba4d2ec7@opensips.org> <19f852eb-91b6-7e93-467b-76a76f767f94@opensips.org> Message-ID: It's always easy to overlook the content in errors that haven't been seen before, I agree that on reflection this should have been looked into in more detail as it does cover the scenario. In the context of a Comodo certificate (which we use regularly) it sounded implausible that we wouldn't be able to validate it. TIL - intermediate certificates matter. On Tue, Jul 25, 2017 at 4:27 PM Bogdan-Andrei Iancu wrote: > I have to admit that you have to "know how to read the SSL errors" in > order to really understand the root problem :) . Now that you find the > issue and if we look back at the error description "verify > error:num=20:unable to get local issuer certificate", it make sense - SSL > complains it did not find the comodo CA in order to validate the > certificate presented by the TLS client (which was probably signed by > Comodo). > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/25/2017 05:27 PM, Callum Guy wrote: > > Hi Bogdan, > > Thanks for your response, based on your advice I performed a full packet > capture on the handshake and established that a certificate was indeed > being presented. > > Following up on this I managed to establish that the problem was a missing > intermediary CA in the certificate chain, specifically: > > > https://support.comodo.com/index.php?/Knowledgebase/Article/View/975/108/intermediate-2-sha-2-comodo-rsa-extended-validation-secure-server-ca > > The error message presented by OpenSIPs was certainly misleading in this > case. For others benefit the approach for installing a new CA is super > simple: > > 1. create the file in /etc/pki/ca-trust/source/anchors > (i.e. comodo-ca-rsa-ev-secure-server.pem) > 2. run "update-ca-trust" with root privs > > Problem solved. > > Have a good day all! > > Callum > > On Tue, Jul 25, 2017 at 2:48 PM Bogdan-Andrei Iancu > wrote: > >> Hi Callum, >> >> The error may indicate the fact that the TLS client does not present a >> TLS certificate while connection to your OpenSIPS. This has nothing to do >> with the TLS multi domain, which anyhow is supported. As the test, you can >> create a separate TLS domain (server) bound to the IP of that TLS client, >> TLS domain having the require_certificate option turned off. >> >> Best Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> OpenSIPS Bootcamp 2017, Houston, US >> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html >> >> On 07/25/2017 03:26 PM, Callum Guy wrote: >> >> Hi All, >> >> *Running: *opensips-2.3.1-1.el7.x86_64 / CentOS 7 >> >> I have been working with new TLS connection and have been having problems >> validating their client certificate. My OpenSIPs configuration works fine >> for other providers (i.e. Twilio) however I am seeing the following error >> messages reported while verify_cert is enabled: >> >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> NOTICE:tls_mgm:verify_callback: depth = 0 >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> NOTICE:tls_mgm:verify_callback: subject = >> /serialNumber=03379831/1.3.6.1.4.1.311.60.2.1.3=GB/businessCategory=Private >> Organization/C=GB/postalCode=SO16 7NP/L=Southampton/street=2 Venture >> Road/O=SIMWOOD ESMS LIMITED/OU=COMODO EV Multi-Domain SSL/CN=simwood.com >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> NOTICE:tls_mgm:verify_callback: verify error:num=20:unable to get local >> issuer certificate >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> NOTICE:tls_mgm:verify_callback: something wrong with the cert ... error >> code is 20 (check x509_vfy.h) >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> NOTICE:tls_mgm:verify_callback: verify return:0 >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: ERROR:proto_tls:tls_accept: >> New TLS connection from 178.22.140.34:34281 failed to accept >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> ERROR:proto_tls:tls_print_errstack: TLS errstack: error:140890B2:SSL >> routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned >> Jul 25 13:10:32 proxy.ex.com opensips[4881]: >> ERROR:proto_tls:tls_read_req: failed to do pre-tls reading >> >> Part of my reason for resorting to the mailing list are old mailing list >> emails discussing that multi-domain certificates are not supported by >> OpenSIPs - is anyone able to confirm if this remains a problem? >> >> The openssl error code 20 is translated as >> X509_V_ERR_UNABLE_TO_GET_ISSUER_CERT_LOCALLY >> >> I have seen other reports that this issue may be related to an improperly >> chained certificate - does this sound at all likely? >> >> Any tips on debugging would be greatly appreciated, thanks. >> >> Callum >> -- >> Callum Guy >> Head of Information Security >> X-on >> >> >> *0333 332 0000 | www.x-on.co.uk | ** >> >> * >> 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 <+44%20333%20332%200000> 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 listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> -- > Callum Guy > Head of Information Security > X-on > > > *0333 332 0000 | www.x-on.co.uk | ** > > * > 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 <+44%20333%20332%200000> 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. > > > -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Tue Jul 25 14:18:22 2017 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 25 Jul 2017 14:18:22 -0400 Subject: [OpenSIPS-Users] !rest_get behavior Message-ID: Group I am using the latest opensips 2.3 and I am wondering why a 404 response doesn't invoke this block ? if (!rest_get("http://$avp(api)/cc/authorized/$fU/$rU", "$json(authresponse)", "$var(ct)", "$var(rcode)")) { xlog("Error code $var(rcode) in HTTP GET!\n"); xlog("on account of admittance error we are sending the call to the AS server for processing"); route(ASroute); } the far end response looks like this. HTTP/1.1 404 Not Found. X-Powered-By: Express. Vary: Origin, Accept-Encoding. Access-Control-Allow-Credentials: true. Content-Type: text/plain; charset=utf-8. Content-Length: 9. ETag: W/"9-nR6tc+Z4+i9RpwqTOwvwFw". Date: Tue, 25 Jul 2017 18:11:39 GMT. Connection: keep-alive. . Not Found from the example it looks like other 4XX responses are considered. Thanks, Tito -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 26 02:50:17 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 26 Jul 2017 09:50:17 +0300 Subject: [OpenSIPS-Users] !rest_get behavior In-Reply-To: References: Message-ID: <99752349-90f0-cd56-a155-97f1e3969db8@opensips.org> Hi Tito, It seems that originally, the rest_xxx() functions could return either "false" or "true", depending if the used libcurl was patched or not. This commit [1] fixed the problem, but the doc examples were not updated. I'll have it fixed. [1]: https://github.com/OpenSIPS/opensips/commit/1363a620 Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 25.07.2017 21:18, Tito Cumpen wrote: > > Group > > I am using the latest opensips 2.3 and I am wondering why a 404 > response doesn't invoke this block ? > > > if (!rest_get("http://$avp(api)/cc/authorized/$fU/$rU", > "$json(authresponse)", "$var(ct)", "$var(rcode)")) { > > xlog("Error code $var(rcode) in HTTP GET!\n"); > > > xlog("on account of admittance error we are sending the call to the AS > server for processing"); > > route(ASroute); > > } > > > > the far end response looks like this. > > > HTTP/1.1 404 Not Found. > > X-Powered-By: Express. > > Vary: Origin, Accept-Encoding. > > Access-Control-Allow-Credentials: true. > > Content-Type: text/plain; charset=utf-8. > > Content-Length: 9. > > ETag: W/"9-nR6tc+Z4+i9RpwqTOwvwFw". > > Date: Tue, 25 Jul 2017 18:11:39 GMT. > > Connection: keep-alive. > > . > > Not Found > > from the example it looks like other 4XX responses are considered. > > > Thanks, > Tito > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From goley_ev_sp at mail.ru Wed Jul 26 08:41:06 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Wed, 26 Jul 2017 05:41:06 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1500907583776-7608091.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> <1500907583776-7608091.post@n2.nabble.com> Message-ID: <1501072866468-7608134.post@n2.nabble.com> Hi, Liviu Chircu Has found out in what the reason - the reason was in too big (very big) values ​​of parameters: modparam ("tm", "fr_timeout", 30000) modparam ("tm", "fr_inv_timeout", 120000) modparam ("tm", "wt_timer", 15000) I copied the parameters from an article on the Internet, so I did not pay attention to their values. During debugging, I noticed that after the transaction ended in case of errors, for a very long time in tcpdump I saw the answers that generated the opensips, understood that it should not be so, but postponed the setting. And when I decided to reduce the time for generating answers or requests without an answer, then I found out that the problem with not freeing memory was exactly this. Now I've reduced the values ​​and everything has risen as it should be. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608134.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From goley_ev_sp at mail.ru Wed Jul 26 09:02:12 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Wed, 26 Jul 2017 06:02:12 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1501072866468-7608134.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> <1500907583776-7608091.post@n2.nabble.com> <1501072866468-7608134.post@n2.nabble.com> Message-ID: <1501074132730-7608135.post@n2.nabble.com> Now I'm testing the server under load, 600 calls per minute and watching the situation. I have only two sockets in the configuration: listen = udp: 10.2.1.61: 7062 listen = udp: ХХ.ХХ.ХХХ.XXX: 7062 In the children parameter the value 2 is specified. But I can not understand why 18 processes are created at startup ?... It's interesting that only two of them take a role in the processing of calls, why? ... I found this out, because of the decrease in two processes (pkmem:4 and pkmem:5) of pkg memory . Please explain the work with pkg memory when it is released. I watched the server for 3 hours under load and 1 hour without, as a result, on two processes the free private memory (pkmem) was rectilinearly reduced. After testing, without restarting the server, private memory (pkgmem) is not released, is it the way it should be? [root at sbc sbin]# date Срд Июл 26 09:31:45 MSK 2017 [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: pkmem:0-total_size:: 67108864 pkmem:0-used_size:: 66608 pkmem:0-real_used_size:: 112560 pkmem:0-max_used_size:: 112560 pkmem:0-free_size:: 66996304 pkmem:0-fragments:: 510 pkmem:1-total_size:: 67108864 pkmem:1-used_size:: 87128 pkmem:1-real_used_size:: 135912 pkmem:1-max_used_size:: 146248 pkmem:1-free_size:: 66972952 pkmem:1-fragments:: 526 pkmem:2-total_size:: 67108864 pkmem:2-used_size:: 64832 pkmem:2-real_used_size:: 110664 pkmem:2-max_used_size:: 110672 pkmem:2-free_size:: 66998200 pkmem:2-fragments:: 500 pkmem:3-total_size:: 67108864 pkmem:3-used_size:: 64832 pkmem:3-real_used_size:: 110664 pkmem:3-max_used_size:: 110672 pkmem:3-free_size:: 66998200 pkmem:3-fragments:: 500 pkmem:4-total_size:: 67108864 pkmem:4-used_size:: 1144576 pkmem:4-real_used_size:: 2058560 pkmem:4-max_used_size:: 2069256 pkmem:4-free_size:: 65050304 pkmem:4-fragments:: 36614 pkmem:5-total_size:: 67108864 pkmem:5-used_size:: 1381584 pkmem:5-real_used_size:: 2495896 pkmem:5-max_used_size:: 2506568 pkmem:5-free_size:: 64612968 pkmem:5-fragments:: 44959 pkmem:6-total_size:: 67108864 pkmem:6-used_size:: 119960 pkmem:6-real_used_size:: 166824 pkmem:6-max_used_size:: 173640 pkmem:6-free_size:: 66942040 pkmem:6-fragments:: 518 pkmem:7-total_size:: 67108864 pkmem:7-used_size:: 119960 pkmem:7-real_used_size:: 166824 pkmem:7-max_used_size:: 173808 pkmem:7-free_size:: 66942040 pkmem:7-fragments:: 518 pkmem:8-total_size:: 67108864 pkmem:8-used_size:: 119840 pkmem:8-real_used_size:: 165912 pkmem:8-max_used_size:: 165912 pkmem:8-free_size:: 66942952 pkmem:8-fragments:: 515 pkmem:9-total_size:: 67108864 pkmem:9-used_size:: 119840 pkmem:9-real_used_size:: 165912 pkmem:9-max_used_size:: 165912 pkmem:9-free_size:: 66942952 pkmem:9-fragments:: 515 pkmem:10-total_size:: 67108864 pkmem:10-used_size:: 119840 pkmem:10-real_used_size:: 165912 pkmem:10-max_used_size:: 165912 pkmem:10-free_size:: 66942952 pkmem:10-fragments:: 515 pkmem:11-total_size:: 67108864 pkmem:11-used_size:: 119840 pkmem:11-real_used_size:: 165912 pkmem:11-max_used_size:: 165912 pkmem:11-free_size:: 66942952 pkmem:11-fragments:: 515 pkmem:12-total_size:: 67108864 pkmem:12-used_size:: 119840 pkmem:12-real_used_size:: 165912 pkmem:12-max_used_size:: 165912 pkmem:12-free_size:: 66942952 pkmem:12-fragments:: 515 pkmem:13-total_size:: 67108864 pkmem:13-used_size:: 119840 pkmem:13-real_used_size:: 165912 pkmem:13-max_used_size:: 165912 pkmem:13-free_size:: 66942952 pkmem:13-fragments:: 515 pkmem:14-total_size:: 67108864 pkmem:14-used_size:: 119840 pkmem:14-real_used_size:: 165912 pkmem:14-max_used_size:: 165912 pkmem:14-free_size:: 66942952 pkmem:14-fragments:: 515 pkmem:15-total_size:: 67108864 pkmem:15-used_size:: 119840 pkmem:15-real_used_size:: 165912 pkmem:15-max_used_size:: 165912 pkmem:15-free_size:: 66942952 pkmem:15-fragments:: 515 pkmem:16-total_size:: 67108864 pkmem:16-used_size:: 119840 pkmem:16-real_used_size:: 165912 pkmem:16-max_used_size:: 165912 pkmem:16-free_size:: 66942952 pkmem:16-fragments:: 515 pkmem:17-total_size:: 67108864 pkmem:17-used_size:: 109904 pkmem:17-real_used_size:: 155832 pkmem:17-max_used_size:: 155832 pkmem:17-free_size:: 66953032 pkmem:17-fragments:: 504 [root at sbc sbin]# .... [root at sbc sbin]# date Срд Июл 26 10:35:47 MSK 2017 [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: pkmem:0-total_size:: 67108864 pkmem:0-used_size:: 66608 pkmem:0-real_used_size:: 112560 pkmem:0-max_used_size:: 112560 pkmem:0-free_size:: 66996304 pkmem:0-fragments:: 510 pkmem:1-total_size:: 67108864 pkmem:1-used_size:: 87128 pkmem:1-real_used_size:: 135912 pkmem:1-max_used_size:: 146248 pkmem:1-free_size:: 66972952 pkmem:1-fragments:: 526 pkmem:2-total_size:: 67108864 pkmem:2-used_size:: 64832 pkmem:2-real_used_size:: 110664 pkmem:2-max_used_size:: 110672 pkmem:2-free_size:: 66998200 pkmem:2-fragments:: 500 pkmem:3-total_size:: 67108864 pkmem:3-used_size:: 64832 pkmem:3-real_used_size:: 110664 pkmem:3-max_used_size:: 110672 pkmem:3-free_size:: 66998200 pkmem:3-fragments:: 500 pkmem:4-total_size:: 67108864 pkmem:4-used_size:: 2786296 pkmem:4-real_used_size:: 5088608 pkmem:4-max_used_size:: 5099248 pkmem:4-free_size:: 62020256 pkmem:4-fragments:: 94460 pkmem:5-total_size:: 67108864 pkmem:5-used_size:: 3334952 pkmem:5-real_used_size:: 6101784 pkmem:5-max_used_size:: 6112384 pkmem:5-free_size:: 61007080 pkmem:5-fragments:: 113815 pkmem:6-total_size:: 67108864 pkmem:6-used_size:: 120008 pkmem:6-real_used_size:: 167496 pkmem:6-max_used_size:: 175304 pkmem:6-free_size:: 66941368 pkmem:6-fragments:: 519 pkmem:7-total_size:: 67108864 pkmem:7-used_size:: 120000 pkmem:7-real_used_size:: 167584 pkmem:7-max_used_size:: 175240 pkmem:7-free_size:: 66941280 pkmem:7-fragments:: 519 pkmem:8-total_size:: 67108864 pkmem:8-used_size:: 119880 pkmem:8-real_used_size:: 165976 pkmem:8-max_used_size:: 165976 pkmem:8-free_size:: 66942888 pkmem:8-fragments:: 516 pkmem:9-total_size:: 67108864 pkmem:9-used_size:: 119952 pkmem:9-real_used_size:: 166288 pkmem:9-max_used_size:: 167024 pkmem:9-free_size:: 66942576 pkmem:9-fragments:: 518 pkmem:10-total_size:: 67108864 pkmem:10-used_size:: 119880 pkmem:10-real_used_size:: 165976 pkmem:10-max_used_size:: 165976 pkmem:10-free_size:: 66942888 pkmem:10-fragments:: 516 pkmem:11-total_size:: 67108864 pkmem:11-used_size:: 119880 pkmem:11-real_used_size:: 165976 pkmem:11-max_used_size:: 165976 pkmem:11-free_size:: 66942888 pkmem:11-fragments:: 516 pkmem:12-total_size:: 67108864 pkmem:12-used_size:: 119880 pkmem:12-real_used_size:: 165976 pkmem:12-max_used_size:: 165976 pkmem:12-free_size:: 66942888 pkmem:12-fragments:: 516 pkmem:13-total_size:: 67108864 pkmem:13-used_size:: 119952 pkmem:13-real_used_size:: 166288 pkmem:13-max_used_size:: 167024 pkmem:13-free_size:: 66942576 pkmem:13-fragments:: 518 pkmem:14-total_size:: 67108864 pkmem:14-used_size:: 119952 pkmem:14-real_used_size:: 166288 pkmem:14-max_used_size:: 167024 pkmem:14-free_size:: 66942576 pkmem:14-fragments:: 518 pkmem:15-total_size:: 67108864 pkmem:15-used_size:: 119880 pkmem:15-real_used_size:: 165976 pkmem:15-max_used_size:: 165976 pkmem:15-free_size:: 66942888 pkmem:15-fragments:: 516 pkmem:16-total_size:: 67108864 pkmem:16-used_size:: 119880 pkmem:16-real_used_size:: 165976 pkmem:16-max_used_size:: 165976 pkmem:16-free_size:: 66942888 pkmem:16-fragments:: 516 pkmem:17-total_size:: 67108864 pkmem:17-used_size:: 109904 pkmem:17-real_used_size:: 155832 pkmem:17-max_used_size:: 155832 pkmem:17-free_size:: 66953032 pkmem:17-fragments:: 504 .... [root at sbc sbin]# date Срд Июл 26 11:08:37 MSK 2017 [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: pkmem:0-total_size:: 67108864 pkmem:0-used_size:: 66608 pkmem:0-real_used_size:: 112560 pkmem:0-max_used_size:: 112560 pkmem:0-free_size:: 66996304 pkmem:0-fragments:: 510 pkmem:1-total_size:: 67108864 pkmem:1-used_size:: 87128 pkmem:1-real_used_size:: 135912 pkmem:1-max_used_size:: 146248 pkmem:1-free_size:: 66972952 pkmem:1-fragments:: 526 pkmem:2-total_size:: 67108864 pkmem:2-used_size:: 64832 pkmem:2-real_used_size:: 110664 pkmem:2-max_used_size:: 110672 pkmem:2-free_size:: 66998200 pkmem:2-fragments:: 500 pkmem:3-total_size:: 67108864 pkmem:3-used_size:: 64832 pkmem:3-real_used_size:: 110664 pkmem:3-max_used_size:: 110672 pkmem:3-free_size:: 66998200 pkmem:3-fragments:: 500 pkmem:4-total_size:: 67108864 pkmem:4-used_size:: 4144936 pkmem:4-real_used_size:: 7595432 pkmem:4-max_used_size:: 7606080 pkmem:4-free_size:: 59513432 pkmem:4-fragments:: 142300 pkmem:5-total_size:: 67108864 pkmem:5-used_size:: 4946520 pkmem:5-real_used_size:: 9075448 pkmem:5-max_used_size:: 9086080 pkmem:5-free_size:: 58033416 pkmem:5-fragments:: 170570 pkmem:6-total_size:: 67108864 pkmem:6-used_size:: 120008 pkmem:6-real_used_size:: 167496 pkmem:6-max_used_size:: 175304 pkmem:6-free_size:: 66941368 pkmem:6-fragments:: 519 pkmem:7-total_size:: 67108864 pkmem:7-used_size:: 120000 pkmem:7-real_used_size:: 167584 pkmem:7-max_used_size:: 175272 pkmem:7-free_size:: 66941280 pkmem:7-fragments:: 520 pkmem:8-total_size:: 67108864 pkmem:8-used_size:: 119880 pkmem:8-real_used_size:: 165976 pkmem:8-max_used_size:: 165976 pkmem:8-free_size:: 66942888 pkmem:8-fragments:: 516 pkmem:9-total_size:: 67108864 pkmem:9-used_size:: 119952 pkmem:9-real_used_size:: 166288 pkmem:9-max_used_size:: 167024 pkmem:9-free_size:: 66942576 pkmem:9-fragments:: 518 pkmem:10-total_size:: 67108864 pkmem:10-used_size:: 119880 pkmem:10-real_used_size:: 165976 pkmem:10-max_used_size:: 165976 pkmem:10-free_size:: 66942888 pkmem:10-fragments:: 516 pkmem:11-total_size:: 67108864 pkmem:11-used_size:: 119880 pkmem:11-real_used_size:: 165976 pkmem:11-max_used_size:: 165976 pkmem:11-free_size:: 66942888 pkmem:11-fragments:: 516 pkmem:12-total_size:: 67108864 pkmem:12-used_size:: 119880 pkmem:12-real_used_size:: 165976 pkmem:12-max_used_size:: 165976 pkmem:12-free_size:: 66942888 pkmem:12-fragments:: 516 pkmem:13-total_size:: 67108864 pkmem:13-used_size:: 119952 pkmem:13-real_used_size:: 166288 pkmem:13-max_used_size:: 167024 pkmem:13-free_size:: 66942576 pkmem:13-fragments:: 518 pkmem:14-total_size:: 67108864 pkmem:14-used_size:: 119952 pkmem:14-real_used_size:: 166288 pkmem:14-max_used_size:: 167024 pkmem:14-free_size:: 66942576 pkmem:14-fragments:: 518 pkmem:15-total_size:: 67108864 pkmem:15-used_size:: 119880 pkmem:15-real_used_size:: 165976 pkmem:15-max_used_size:: 165976 pkmem:15-free_size:: 66942888 pkmem:15-fragments:: 516 pkmem:16-total_size:: 67108864 pkmem:16-used_size:: 119880 pkmem:16-real_used_size:: 165976 pkmem:16-max_used_size:: 165976 pkmem:16-free_size:: 66942888 pkmem:16-fragments:: 516 pkmem:17-total_size:: 67108864 pkmem:17-used_size:: 109904 pkmem:17-real_used_size:: 155832 pkmem:17-max_used_size:: 155832 pkmem:17-free_size:: 66953032 pkmem:17-fragments:: 504 [root at sbc sbin]# ... After testing, without restarting the service. [root at sbc sbin]# date Срд Июл 26 12:09:01 MSK 2017 [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: pkmem:0-total_size:: 67108864 pkmem:0-used_size:: 66608 pkmem:0-real_used_size:: 112560 pkmem:0-max_used_size:: 112560 pkmem:0-free_size:: 66996304 pkmem:0-fragments:: 510 pkmem:1-total_size:: 67108864 pkmem:1-used_size:: 87128 pkmem:1-real_used_size:: 135912 pkmem:1-max_used_size:: 146248 pkmem:1-free_size:: 66972952 pkmem:1-fragments:: 526 pkmem:2-total_size:: 67108864 pkmem:2-used_size:: 64832 pkmem:2-real_used_size:: 110664 pkmem:2-max_used_size:: 110672 pkmem:2-free_size:: 66998200 pkmem:2-fragments:: 500 pkmem:3-total_size:: 67108864 pkmem:3-used_size:: 64832 pkmem:3-real_used_size:: 110664 pkmem:3-max_used_size:: 110672 pkmem:3-free_size:: 66998200 pkmem:3-fragments:: 500 pkmem:4-total_size:: 67108864 pkmem:4-used_size:: 6444664 pkmem:4-real_used_size:: 11838512 pkmem:4-max_used_size:: 11849192 pkmem:4-free_size:: 55270352 pkmem:4-fragments:: 223275 pkmem:5-total_size:: 67108864 pkmem:5-used_size:: 7769568 pkmem:5-real_used_size:: 14284264 pkmem:5-max_used_size:: 14294848 pkmem:5-free_size:: 52824600 pkmem:5-fragments:: 269975 pkmem:6-total_size:: 67108864 pkmem:6-used_size:: 120000 pkmem:6-real_used_size:: 167512 pkmem:6-max_used_size:: 175336 pkmem:6-free_size:: 66941352 pkmem:6-fragments:: 520 pkmem:7-total_size:: 67108864 pkmem:7-used_size:: 120000 pkmem:7-real_used_size:: 167584 pkmem:7-max_used_size:: 175272 pkmem:7-free_size:: 66941280 pkmem:7-fragments:: 520 pkmem:8-total_size:: 67108864 pkmem:8-used_size:: 119880 pkmem:8-real_used_size:: 165976 pkmem:8-max_used_size:: 165976 pkmem:8-free_size:: 66942888 pkmem:8-fragments:: 516 pkmem:9-total_size:: 67108864 pkmem:9-used_size:: 119952 pkmem:9-real_used_size:: 166288 pkmem:9-max_used_size:: 167024 pkmem:9-free_size:: 66942576 pkmem:9-fragments:: 518 pkmem:10-total_size:: 67108864 pkmem:10-used_size:: 119880 pkmem:10-real_used_size:: 165976 pkmem:10-max_used_size:: 165976 pkmem:10-free_size:: 66942888 pkmem:10-fragments:: 516 pkmem:11-total_size:: 67108864 pkmem:11-used_size:: 119880 pkmem:11-real_used_size:: 165976 pkmem:11-max_used_size:: 165976 pkmem:11-free_size:: 66942888 pkmem:11-fragments:: 516 pkmem:12-total_size:: 67108864 pkmem:12-used_size:: 119880 pkmem:12-real_used_size:: 165976 pkmem:12-max_used_size:: 165976 pkmem:12-free_size:: 66942888 pkmem:12-fragments:: 516 pkmem:13-total_size:: 67108864 pkmem:13-used_size:: 119952 pkmem:13-real_used_size:: 166288 pkmem:13-max_used_size:: 167024 pkmem:13-free_size:: 66942576 pkmem:13-fragments:: 518 pkmem:14-total_size:: 67108864 pkmem:14-used_size:: 119952 pkmem:14-real_used_size:: 166288 pkmem:14-max_used_size:: 167024 pkmem:14-free_size:: 66942576 pkmem:14-fragments:: 518 pkmem:15-total_size:: 67108864 pkmem:15-used_size:: 119880 pkmem:15-real_used_size:: 165976 pkmem:15-max_used_size:: 165976 pkmem:15-free_size:: 66942888 pkmem:15-fragments:: 516 pkmem:16-total_size:: 67108864 pkmem:16-used_size:: 119880 pkmem:16-real_used_size:: 165976 pkmem:16-max_used_size:: 165976 pkmem:16-free_size:: 66942888 pkmem:16-fragments:: 516 pkmem:17-total_size:: 67108864 pkmem:17-used_size:: 109904 pkmem:17-real_used_size:: 155832 pkmem:17-max_used_size:: 155832 pkmem:17-free_size:: 66953032 pkmem:17-fragments:: 504 [root at sbc sbin]# ... Hour without load. [root at sbc sbin]# date Срд Июл 26 12:54:14 MSK 2017 [root at sbc sbin]# [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: pkmem:0-total_size:: 67108864 pkmem:0-used_size:: 66608 pkmem:0-real_used_size:: 112560 pkmem:0-max_used_size:: 112560 pkmem:0-free_size:: 66996304 pkmem:0-fragments:: 510 pkmem:1-total_size:: 67108864 pkmem:1-used_size:: 87128 pkmem:1-real_used_size:: 135912 pkmem:1-max_used_size:: 146248 pkmem:1-free_size:: 66972952 pkmem:1-fragments:: 526 pkmem:2-total_size:: 67108864 pkmem:2-used_size:: 64832 pkmem:2-real_used_size:: 110664 pkmem:2-max_used_size:: 110672 pkmem:2-free_size:: 66998200 pkmem:2-fragments:: 500 pkmem:3-total_size:: 67108864 pkmem:3-used_size:: 64832 pkmem:3-real_used_size:: 110664 pkmem:3-max_used_size:: 110672 pkmem:3-free_size:: 66998200 pkmem:3-fragments:: 500 pkmem:4-total_size:: 67108864 pkmem:4-used_size:: 6444664 pkmem:4-real_used_size:: 11838512 pkmem:4-max_used_size:: 11849192 pkmem:4-free_size:: 55270352 pkmem:4-fragments:: 223275 pkmem:5-total_size:: 67108864 pkmem:5-used_size:: 7769568 pkmem:5-real_used_size:: 14284264 pkmem:5-max_used_size:: 14294848 pkmem:5-free_size:: 52824600 pkmem:5-fragments:: 269975 pkmem:6-total_size:: 67108864 pkmem:6-used_size:: 120000 pkmem:6-real_used_size:: 167512 pkmem:6-max_used_size:: 175336 pkmem:6-free_size:: 66941352 pkmem:6-fragments:: 520 pkmem:7-total_size:: 67108864 pkmem:7-used_size:: 120000 pkmem:7-real_used_size:: 167584 pkmem:7-max_used_size:: 175272 pkmem:7-free_size:: 66941280 pkmem:7-fragments:: 520 pkmem:8-total_size:: 67108864 pkmem:8-used_size:: 119880 pkmem:8-real_used_size:: 165976 pkmem:8-max_used_size:: 165976 pkmem:8-free_size:: 66942888 pkmem:8-fragments:: 516 pkmem:9-total_size:: 67108864 pkmem:9-used_size:: 119952 pkmem:9-real_used_size:: 166288 pkmem:9-max_used_size:: 167024 pkmem:9-free_size:: 66942576 pkmem:9-fragments:: 518 pkmem:10-total_size:: 67108864 pkmem:10-used_size:: 119880 pkmem:10-real_used_size:: 165976 pkmem:10-max_used_size:: 165976 pkmem:10-free_size:: 66942888 pkmem:10-fragments:: 516 pkmem:11-total_size:: 67108864 pkmem:11-used_size:: 119880 pkmem:11-real_used_size:: 165976 pkmem:11-max_used_size:: 165976 pkmem:11-free_size:: 66942888 pkmem:11-fragments:: 516 pkmem:12-total_size:: 67108864 pkmem:12-used_size:: 119880 pkmem:12-real_used_size:: 165976 pkmem:12-max_used_size:: 165976 pkmem:12-free_size:: 66942888 pkmem:12-fragments:: 516 pkmem:13-total_size:: 67108864 pkmem:13-used_size:: 119952 pkmem:13-real_used_size:: 166288 pkmem:13-max_used_size:: 167024 pkmem:13-free_size:: 66942576 pkmem:13-fragments:: 518 pkmem:14-total_size:: 67108864 pkmem:14-used_size:: 119952 pkmem:14-real_used_size:: 166288 pkmem:14-max_used_size:: 167024 pkmem:14-free_size:: 66942576 pkmem:14-fragments:: 518 pkmem:15-total_size:: 67108864 pkmem:15-used_size:: 119880 pkmem:15-real_used_size:: 165976 pkmem:15-max_used_size:: 165976 pkmem:15-free_size:: 66942888 pkmem:15-fragments:: 516 pkmem:16-total_size:: 67108864 pkmem:16-used_size:: 119880 pkmem:16-real_used_size:: 165976 pkmem:16-max_used_size:: 165976 pkmem:16-free_size:: 66942888 pkmem:16-fragments:: 516 pkmem:17-total_size:: 67108864 pkmem:17-used_size:: 109904 pkmem:17-real_used_size:: 155832 pkmem:17-max_used_size:: 155832 pkmem:17-free_size:: 66953032 pkmem:17-fragments:: 504 [root at sbc sbin]# -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608135.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From liviu at opensips.org Wed Jul 26 09:42:34 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 26 Jul 2017 16:42:34 +0300 Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1501074132730-7608135.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> <1500907583776-7608091.post@n2.nabble.com> <1501072866468-7608134.post@n2.nabble.com> <1501074132730-7608135.post@n2.nabble.com> Message-ID: <7b0172f6-8517-de52-0286-9bf4cd5afbb3@opensips.org> Here are some more MI commands that might be useful: opensipsctl fifo ps -> the type of each process opensipsctl fifo get_statistics dialog: tm: usrloc: -> display in-memory # of transactions / dialogs / registered users Like before, if you suspect a leak, please provide the output of the above MI commands, along with a "kill -SIGUSR1" memory map as instructed earlier, and we should have a solid indication as to whether there's a real problem at hand or not. Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 26.07.2017 16:02, Evgeniy G. via Users wrote: > Now I'm testing the server under load, 600 calls per minute and watching the > situation. I have only two sockets in the configuration: > listen = udp: 10.2.1.61: 7062 > listen = udp: ХХ.ХХ.ХХХ.XXX: 7062 > In the children parameter the value 2 is specified. But I can not understand > why 18 processes are created at startup ?... It's interesting that only two > of them take a role in the processing of calls, why? ... I found this out, > because of the decrease in two processes (pkmem:4 and pkmem:5) of pkg memory > . > Please explain the work with pkg memory when it is released. > I watched the server for 3 hours under load and 1 hour without, as a result, > on two processes the free private memory (pkmem) was rectilinearly reduced. > After testing, without restarting the server, private memory (pkgmem) is not > released, is it the way it should be? > > [root at sbc sbin]# date > Срд Июл 26 09:31:45 MSK 2017 > [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: > [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: > pkmem:0-total_size:: 67108864 > pkmem:0-used_size:: 66608 > pkmem:0-real_used_size:: 112560 > pkmem:0-max_used_size:: 112560 > pkmem:0-free_size:: 66996304 > pkmem:0-fragments:: 510 > pkmem:1-total_size:: 67108864 > pkmem:1-used_size:: 87128 > pkmem:1-real_used_size:: 135912 > pkmem:1-max_used_size:: 146248 > pkmem:1-free_size:: 66972952 > pkmem:1-fragments:: 526 > pkmem:2-total_size:: 67108864 > pkmem:2-used_size:: 64832 > pkmem:2-real_used_size:: 110664 > pkmem:2-max_used_size:: 110672 > pkmem:2-free_size:: 66998200 > pkmem:2-fragments:: 500 > pkmem:3-total_size:: 67108864 > pkmem:3-used_size:: 64832 > pkmem:3-real_used_size:: 110664 > pkmem:3-max_used_size:: 110672 > pkmem:3-free_size:: 66998200 > pkmem:3-fragments:: 500 > pkmem:4-total_size:: 67108864 > pkmem:4-used_size:: 1144576 > pkmem:4-real_used_size:: 2058560 > pkmem:4-max_used_size:: 2069256 > pkmem:4-free_size:: 65050304 > pkmem:4-fragments:: 36614 > pkmem:5-total_size:: 67108864 > pkmem:5-used_size:: 1381584 > pkmem:5-real_used_size:: 2495896 > pkmem:5-max_used_size:: 2506568 > pkmem:5-free_size:: 64612968 > pkmem:5-fragments:: 44959 > pkmem:6-total_size:: 67108864 > pkmem:6-used_size:: 119960 > pkmem:6-real_used_size:: 166824 > pkmem:6-max_used_size:: 173640 > pkmem:6-free_size:: 66942040 > pkmem:6-fragments:: 518 > pkmem:7-total_size:: 67108864 > pkmem:7-used_size:: 119960 > pkmem:7-real_used_size:: 166824 > pkmem:7-max_used_size:: 173808 > pkmem:7-free_size:: 66942040 > pkmem:7-fragments:: 518 > pkmem:8-total_size:: 67108864 > pkmem:8-used_size:: 119840 > pkmem:8-real_used_size:: 165912 > pkmem:8-max_used_size:: 165912 > pkmem:8-free_size:: 66942952 > pkmem:8-fragments:: 515 > pkmem:9-total_size:: 67108864 > pkmem:9-used_size:: 119840 > pkmem:9-real_used_size:: 165912 > pkmem:9-max_used_size:: 165912 > pkmem:9-free_size:: 66942952 > pkmem:9-fragments:: 515 > pkmem:10-total_size:: 67108864 > pkmem:10-used_size:: 119840 > pkmem:10-real_used_size:: 165912 > pkmem:10-max_used_size:: 165912 > pkmem:10-free_size:: 66942952 > pkmem:10-fragments:: 515 > pkmem:11-total_size:: 67108864 > pkmem:11-used_size:: 119840 > pkmem:11-real_used_size:: 165912 > pkmem:11-max_used_size:: 165912 > pkmem:11-free_size:: 66942952 > pkmem:11-fragments:: 515 > pkmem:12-total_size:: 67108864 > pkmem:12-used_size:: 119840 > pkmem:12-real_used_size:: 165912 > pkmem:12-max_used_size:: 165912 > pkmem:12-free_size:: 66942952 > pkmem:12-fragments:: 515 > pkmem:13-total_size:: 67108864 > pkmem:13-used_size:: 119840 > pkmem:13-real_used_size:: 165912 > pkmem:13-max_used_size:: 165912 > pkmem:13-free_size:: 66942952 > pkmem:13-fragments:: 515 > pkmem:14-total_size:: 67108864 > pkmem:14-used_size:: 119840 > pkmem:14-real_used_size:: 165912 > pkmem:14-max_used_size:: 165912 > pkmem:14-free_size:: 66942952 > pkmem:14-fragments:: 515 > pkmem:15-total_size:: 67108864 > pkmem:15-used_size:: 119840 > pkmem:15-real_used_size:: 165912 > pkmem:15-max_used_size:: 165912 > pkmem:15-free_size:: 66942952 > pkmem:15-fragments:: 515 > pkmem:16-total_size:: 67108864 > pkmem:16-used_size:: 119840 > pkmem:16-real_used_size:: 165912 > pkmem:16-max_used_size:: 165912 > pkmem:16-free_size:: 66942952 > pkmem:16-fragments:: 515 > pkmem:17-total_size:: 67108864 > pkmem:17-used_size:: 109904 > pkmem:17-real_used_size:: 155832 > pkmem:17-max_used_size:: 155832 > pkmem:17-free_size:: 66953032 > pkmem:17-fragments:: 504 > [root at sbc sbin]# > > .... > > > [root at sbc sbin]# date > Срд Июл 26 10:35:47 MSK 2017 > > [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: > pkmem:0-total_size:: 67108864 > pkmem:0-used_size:: 66608 > pkmem:0-real_used_size:: 112560 > pkmem:0-max_used_size:: 112560 > pkmem:0-free_size:: 66996304 > pkmem:0-fragments:: 510 > pkmem:1-total_size:: 67108864 > pkmem:1-used_size:: 87128 > pkmem:1-real_used_size:: 135912 > pkmem:1-max_used_size:: 146248 > pkmem:1-free_size:: 66972952 > pkmem:1-fragments:: 526 > pkmem:2-total_size:: 67108864 > pkmem:2-used_size:: 64832 > pkmem:2-real_used_size:: 110664 > pkmem:2-max_used_size:: 110672 > pkmem:2-free_size:: 66998200 > pkmem:2-fragments:: 500 > pkmem:3-total_size:: 67108864 > pkmem:3-used_size:: 64832 > pkmem:3-real_used_size:: 110664 > pkmem:3-max_used_size:: 110672 > pkmem:3-free_size:: 66998200 > pkmem:3-fragments:: 500 > pkmem:4-total_size:: 67108864 > pkmem:4-used_size:: 2786296 > pkmem:4-real_used_size:: 5088608 > pkmem:4-max_used_size:: 5099248 > pkmem:4-free_size:: 62020256 > pkmem:4-fragments:: 94460 > pkmem:5-total_size:: 67108864 > pkmem:5-used_size:: 3334952 > pkmem:5-real_used_size:: 6101784 > pkmem:5-max_used_size:: 6112384 > pkmem:5-free_size:: 61007080 > pkmem:5-fragments:: 113815 > pkmem:6-total_size:: 67108864 > pkmem:6-used_size:: 120008 > pkmem:6-real_used_size:: 167496 > pkmem:6-max_used_size:: 175304 > pkmem:6-free_size:: 66941368 > pkmem:6-fragments:: 519 > pkmem:7-total_size:: 67108864 > pkmem:7-used_size:: 120000 > pkmem:7-real_used_size:: 167584 > pkmem:7-max_used_size:: 175240 > pkmem:7-free_size:: 66941280 > pkmem:7-fragments:: 519 > pkmem:8-total_size:: 67108864 > pkmem:8-used_size:: 119880 > pkmem:8-real_used_size:: 165976 > pkmem:8-max_used_size:: 165976 > pkmem:8-free_size:: 66942888 > pkmem:8-fragments:: 516 > pkmem:9-total_size:: 67108864 > pkmem:9-used_size:: 119952 > pkmem:9-real_used_size:: 166288 > pkmem:9-max_used_size:: 167024 > pkmem:9-free_size:: 66942576 > pkmem:9-fragments:: 518 > pkmem:10-total_size:: 67108864 > pkmem:10-used_size:: 119880 > pkmem:10-real_used_size:: 165976 > pkmem:10-max_used_size:: 165976 > pkmem:10-free_size:: 66942888 > pkmem:10-fragments:: 516 > pkmem:11-total_size:: 67108864 > pkmem:11-used_size:: 119880 > pkmem:11-real_used_size:: 165976 > pkmem:11-max_used_size:: 165976 > pkmem:11-free_size:: 66942888 > pkmem:11-fragments:: 516 > pkmem:12-total_size:: 67108864 > pkmem:12-used_size:: 119880 > pkmem:12-real_used_size:: 165976 > pkmem:12-max_used_size:: 165976 > pkmem:12-free_size:: 66942888 > pkmem:12-fragments:: 516 > pkmem:13-total_size:: 67108864 > pkmem:13-used_size:: 119952 > pkmem:13-real_used_size:: 166288 > pkmem:13-max_used_size:: 167024 > pkmem:13-free_size:: 66942576 > pkmem:13-fragments:: 518 > pkmem:14-total_size:: 67108864 > pkmem:14-used_size:: 119952 > pkmem:14-real_used_size:: 166288 > pkmem:14-max_used_size:: 167024 > pkmem:14-free_size:: 66942576 > pkmem:14-fragments:: 518 > pkmem:15-total_size:: 67108864 > pkmem:15-used_size:: 119880 > pkmem:15-real_used_size:: 165976 > pkmem:15-max_used_size:: 165976 > pkmem:15-free_size:: 66942888 > pkmem:15-fragments:: 516 > pkmem:16-total_size:: 67108864 > pkmem:16-used_size:: 119880 > pkmem:16-real_used_size:: 165976 > pkmem:16-max_used_size:: 165976 > pkmem:16-free_size:: 66942888 > pkmem:16-fragments:: 516 > pkmem:17-total_size:: 67108864 > pkmem:17-used_size:: 109904 > pkmem:17-real_used_size:: 155832 > pkmem:17-max_used_size:: 155832 > pkmem:17-free_size:: 66953032 > pkmem:17-fragments:: 504 > > .... > > [root at sbc sbin]# date > Срд Июл 26 11:08:37 MSK 2017 > [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: > pkmem:0-total_size:: 67108864 > pkmem:0-used_size:: 66608 > pkmem:0-real_used_size:: 112560 > pkmem:0-max_used_size:: 112560 > pkmem:0-free_size:: 66996304 > pkmem:0-fragments:: 510 > pkmem:1-total_size:: 67108864 > pkmem:1-used_size:: 87128 > pkmem:1-real_used_size:: 135912 > pkmem:1-max_used_size:: 146248 > pkmem:1-free_size:: 66972952 > pkmem:1-fragments:: 526 > pkmem:2-total_size:: 67108864 > pkmem:2-used_size:: 64832 > pkmem:2-real_used_size:: 110664 > pkmem:2-max_used_size:: 110672 > pkmem:2-free_size:: 66998200 > pkmem:2-fragments:: 500 > pkmem:3-total_size:: 67108864 > pkmem:3-used_size:: 64832 > pkmem:3-real_used_size:: 110664 > pkmem:3-max_used_size:: 110672 > pkmem:3-free_size:: 66998200 > pkmem:3-fragments:: 500 > pkmem:4-total_size:: 67108864 > pkmem:4-used_size:: 4144936 > pkmem:4-real_used_size:: 7595432 > pkmem:4-max_used_size:: 7606080 > pkmem:4-free_size:: 59513432 > pkmem:4-fragments:: 142300 > pkmem:5-total_size:: 67108864 > pkmem:5-used_size:: 4946520 > pkmem:5-real_used_size:: 9075448 > pkmem:5-max_used_size:: 9086080 > pkmem:5-free_size:: 58033416 > pkmem:5-fragments:: 170570 > pkmem:6-total_size:: 67108864 > pkmem:6-used_size:: 120008 > pkmem:6-real_used_size:: 167496 > pkmem:6-max_used_size:: 175304 > pkmem:6-free_size:: 66941368 > pkmem:6-fragments:: 519 > pkmem:7-total_size:: 67108864 > pkmem:7-used_size:: 120000 > pkmem:7-real_used_size:: 167584 > pkmem:7-max_used_size:: 175272 > pkmem:7-free_size:: 66941280 > pkmem:7-fragments:: 520 > pkmem:8-total_size:: 67108864 > pkmem:8-used_size:: 119880 > pkmem:8-real_used_size:: 165976 > pkmem:8-max_used_size:: 165976 > pkmem:8-free_size:: 66942888 > pkmem:8-fragments:: 516 > pkmem:9-total_size:: 67108864 > pkmem:9-used_size:: 119952 > pkmem:9-real_used_size:: 166288 > pkmem:9-max_used_size:: 167024 > pkmem:9-free_size:: 66942576 > pkmem:9-fragments:: 518 > pkmem:10-total_size:: 67108864 > pkmem:10-used_size:: 119880 > pkmem:10-real_used_size:: 165976 > pkmem:10-max_used_size:: 165976 > pkmem:10-free_size:: 66942888 > pkmem:10-fragments:: 516 > pkmem:11-total_size:: 67108864 > pkmem:11-used_size:: 119880 > pkmem:11-real_used_size:: 165976 > pkmem:11-max_used_size:: 165976 > pkmem:11-free_size:: 66942888 > pkmem:11-fragments:: 516 > pkmem:12-total_size:: 67108864 > pkmem:12-used_size:: 119880 > pkmem:12-real_used_size:: 165976 > pkmem:12-max_used_size:: 165976 > pkmem:12-free_size:: 66942888 > pkmem:12-fragments:: 516 > pkmem:13-total_size:: 67108864 > pkmem:13-used_size:: 119952 > pkmem:13-real_used_size:: 166288 > pkmem:13-max_used_size:: 167024 > pkmem:13-free_size:: 66942576 > pkmem:13-fragments:: 518 > pkmem:14-total_size:: 67108864 > pkmem:14-used_size:: 119952 > pkmem:14-real_used_size:: 166288 > pkmem:14-max_used_size:: 167024 > pkmem:14-free_size:: 66942576 > pkmem:14-fragments:: 518 > pkmem:15-total_size:: 67108864 > pkmem:15-used_size:: 119880 > pkmem:15-real_used_size:: 165976 > pkmem:15-max_used_size:: 165976 > pkmem:15-free_size:: 66942888 > pkmem:15-fragments:: 516 > pkmem:16-total_size:: 67108864 > pkmem:16-used_size:: 119880 > pkmem:16-real_used_size:: 165976 > pkmem:16-max_used_size:: 165976 > pkmem:16-free_size:: 66942888 > pkmem:16-fragments:: 516 > pkmem:17-total_size:: 67108864 > pkmem:17-used_size:: 109904 > pkmem:17-real_used_size:: 155832 > pkmem:17-max_used_size:: 155832 > pkmem:17-free_size:: 66953032 > pkmem:17-fragments:: 504 > [root at sbc sbin]# > > ... > After testing, without restarting the service. > > [root at sbc sbin]# date > Срд Июл 26 12:09:01 MSK 2017 > [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: > pkmem:0-total_size:: 67108864 > pkmem:0-used_size:: 66608 > pkmem:0-real_used_size:: 112560 > pkmem:0-max_used_size:: 112560 > pkmem:0-free_size:: 66996304 > pkmem:0-fragments:: 510 > pkmem:1-total_size:: 67108864 > pkmem:1-used_size:: 87128 > pkmem:1-real_used_size:: 135912 > pkmem:1-max_used_size:: 146248 > pkmem:1-free_size:: 66972952 > pkmem:1-fragments:: 526 > pkmem:2-total_size:: 67108864 > pkmem:2-used_size:: 64832 > pkmem:2-real_used_size:: 110664 > pkmem:2-max_used_size:: 110672 > pkmem:2-free_size:: 66998200 > pkmem:2-fragments:: 500 > pkmem:3-total_size:: 67108864 > pkmem:3-used_size:: 64832 > pkmem:3-real_used_size:: 110664 > pkmem:3-max_used_size:: 110672 > pkmem:3-free_size:: 66998200 > pkmem:3-fragments:: 500 > pkmem:4-total_size:: 67108864 > pkmem:4-used_size:: 6444664 > pkmem:4-real_used_size:: 11838512 > pkmem:4-max_used_size:: 11849192 > pkmem:4-free_size:: 55270352 > pkmem:4-fragments:: 223275 > pkmem:5-total_size:: 67108864 > pkmem:5-used_size:: 7769568 > pkmem:5-real_used_size:: 14284264 > pkmem:5-max_used_size:: 14294848 > pkmem:5-free_size:: 52824600 > pkmem:5-fragments:: 269975 > pkmem:6-total_size:: 67108864 > pkmem:6-used_size:: 120000 > pkmem:6-real_used_size:: 167512 > pkmem:6-max_used_size:: 175336 > pkmem:6-free_size:: 66941352 > pkmem:6-fragments:: 520 > pkmem:7-total_size:: 67108864 > pkmem:7-used_size:: 120000 > pkmem:7-real_used_size:: 167584 > pkmem:7-max_used_size:: 175272 > pkmem:7-free_size:: 66941280 > pkmem:7-fragments:: 520 > pkmem:8-total_size:: 67108864 > pkmem:8-used_size:: 119880 > pkmem:8-real_used_size:: 165976 > pkmem:8-max_used_size:: 165976 > pkmem:8-free_size:: 66942888 > pkmem:8-fragments:: 516 > pkmem:9-total_size:: 67108864 > pkmem:9-used_size:: 119952 > pkmem:9-real_used_size:: 166288 > pkmem:9-max_used_size:: 167024 > pkmem:9-free_size:: 66942576 > pkmem:9-fragments:: 518 > pkmem:10-total_size:: 67108864 > pkmem:10-used_size:: 119880 > pkmem:10-real_used_size:: 165976 > pkmem:10-max_used_size:: 165976 > pkmem:10-free_size:: 66942888 > pkmem:10-fragments:: 516 > pkmem:11-total_size:: 67108864 > pkmem:11-used_size:: 119880 > pkmem:11-real_used_size:: 165976 > pkmem:11-max_used_size:: 165976 > pkmem:11-free_size:: 66942888 > pkmem:11-fragments:: 516 > pkmem:12-total_size:: 67108864 > pkmem:12-used_size:: 119880 > pkmem:12-real_used_size:: 165976 > pkmem:12-max_used_size:: 165976 > pkmem:12-free_size:: 66942888 > pkmem:12-fragments:: 516 > pkmem:13-total_size:: 67108864 > pkmem:13-used_size:: 119952 > pkmem:13-real_used_size:: 166288 > pkmem:13-max_used_size:: 167024 > pkmem:13-free_size:: 66942576 > pkmem:13-fragments:: 518 > pkmem:14-total_size:: 67108864 > pkmem:14-used_size:: 119952 > pkmem:14-real_used_size:: 166288 > pkmem:14-max_used_size:: 167024 > pkmem:14-free_size:: 66942576 > pkmem:14-fragments:: 518 > pkmem:15-total_size:: 67108864 > pkmem:15-used_size:: 119880 > pkmem:15-real_used_size:: 165976 > pkmem:15-max_used_size:: 165976 > pkmem:15-free_size:: 66942888 > pkmem:15-fragments:: 516 > pkmem:16-total_size:: 67108864 > pkmem:16-used_size:: 119880 > pkmem:16-real_used_size:: 165976 > pkmem:16-max_used_size:: 165976 > pkmem:16-free_size:: 66942888 > pkmem:16-fragments:: 516 > pkmem:17-total_size:: 67108864 > pkmem:17-used_size:: 109904 > pkmem:17-real_used_size:: 155832 > pkmem:17-max_used_size:: 155832 > pkmem:17-free_size:: 66953032 > pkmem:17-fragments:: 504 > [root at sbc sbin]# > > ... > > Hour without load. > > [root at sbc sbin]# date > Срд Июл 26 12:54:14 MSK 2017 > [root at sbc sbin]# > [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: > pkmem:0-total_size:: 67108864 > pkmem:0-used_size:: 66608 > pkmem:0-real_used_size:: 112560 > pkmem:0-max_used_size:: 112560 > pkmem:0-free_size:: 66996304 > pkmem:0-fragments:: 510 > pkmem:1-total_size:: 67108864 > pkmem:1-used_size:: 87128 > pkmem:1-real_used_size:: 135912 > pkmem:1-max_used_size:: 146248 > pkmem:1-free_size:: 66972952 > pkmem:1-fragments:: 526 > pkmem:2-total_size:: 67108864 > pkmem:2-used_size:: 64832 > pkmem:2-real_used_size:: 110664 > pkmem:2-max_used_size:: 110672 > pkmem:2-free_size:: 66998200 > pkmem:2-fragments:: 500 > pkmem:3-total_size:: 67108864 > pkmem:3-used_size:: 64832 > pkmem:3-real_used_size:: 110664 > pkmem:3-max_used_size:: 110672 > pkmem:3-free_size:: 66998200 > pkmem:3-fragments:: 500 > pkmem:4-total_size:: 67108864 > pkmem:4-used_size:: 6444664 > pkmem:4-real_used_size:: 11838512 > pkmem:4-max_used_size:: 11849192 > pkmem:4-free_size:: 55270352 > pkmem:4-fragments:: 223275 > pkmem:5-total_size:: 67108864 > pkmem:5-used_size:: 7769568 > pkmem:5-real_used_size:: 14284264 > pkmem:5-max_used_size:: 14294848 > pkmem:5-free_size:: 52824600 > pkmem:5-fragments:: 269975 > pkmem:6-total_size:: 67108864 > pkmem:6-used_size:: 120000 > pkmem:6-real_used_size:: 167512 > pkmem:6-max_used_size:: 175336 > pkmem:6-free_size:: 66941352 > pkmem:6-fragments:: 520 > pkmem:7-total_size:: 67108864 > pkmem:7-used_size:: 120000 > pkmem:7-real_used_size:: 167584 > pkmem:7-max_used_size:: 175272 > pkmem:7-free_size:: 66941280 > pkmem:7-fragments:: 520 > pkmem:8-total_size:: 67108864 > pkmem:8-used_size:: 119880 > pkmem:8-real_used_size:: 165976 > pkmem:8-max_used_size:: 165976 > pkmem:8-free_size:: 66942888 > pkmem:8-fragments:: 516 > pkmem:9-total_size:: 67108864 > pkmem:9-used_size:: 119952 > pkmem:9-real_used_size:: 166288 > pkmem:9-max_used_size:: 167024 > pkmem:9-free_size:: 66942576 > pkmem:9-fragments:: 518 > pkmem:10-total_size:: 67108864 > pkmem:10-used_size:: 119880 > pkmem:10-real_used_size:: 165976 > pkmem:10-max_used_size:: 165976 > pkmem:10-free_size:: 66942888 > pkmem:10-fragments:: 516 > pkmem:11-total_size:: 67108864 > pkmem:11-used_size:: 119880 > pkmem:11-real_used_size:: 165976 > pkmem:11-max_used_size:: 165976 > pkmem:11-free_size:: 66942888 > pkmem:11-fragments:: 516 > pkmem:12-total_size:: 67108864 > pkmem:12-used_size:: 119880 > pkmem:12-real_used_size:: 165976 > pkmem:12-max_used_size:: 165976 > pkmem:12-free_size:: 66942888 > pkmem:12-fragments:: 516 > pkmem:13-total_size:: 67108864 > pkmem:13-used_size:: 119952 > pkmem:13-real_used_size:: 166288 > pkmem:13-max_used_size:: 167024 > pkmem:13-free_size:: 66942576 > pkmem:13-fragments:: 518 > pkmem:14-total_size:: 67108864 > pkmem:14-used_size:: 119952 > pkmem:14-real_used_size:: 166288 > pkmem:14-max_used_size:: 167024 > pkmem:14-free_size:: 66942576 > pkmem:14-fragments:: 518 > pkmem:15-total_size:: 67108864 > pkmem:15-used_size:: 119880 > pkmem:15-real_used_size:: 165976 > pkmem:15-max_used_size:: 165976 > pkmem:15-free_size:: 66942888 > pkmem:15-fragments:: 516 > pkmem:16-total_size:: 67108864 > pkmem:16-used_size:: 119880 > pkmem:16-real_used_size:: 165976 > pkmem:16-max_used_size:: 165976 > pkmem:16-free_size:: 66942888 > pkmem:16-fragments:: 516 > pkmem:17-total_size:: 67108864 > pkmem:17-used_size:: 109904 > pkmem:17-real_used_size:: 155832 > pkmem:17-max_used_size:: 155832 > pkmem:17-free_size:: 66953032 > pkmem:17-fragments:: 504 > [root at sbc sbin]# > > > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608135.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From goley_ev_sp at mail.ru Wed Jul 26 09:43:26 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Wed, 26 Jul 2017 06:43:26 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1501074132730-7608135.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> <1500907583776-7608091.post@n2.nabble.com> <1501072866468-7608134.post@n2.nabble.com> <1501074132730-7608135.post@n2.nabble.com> Message-ID: <1501076606080-7608136.post@n2.nabble.com> Thank you for opensipsctl fifo ps [root at sbc sbin]# ./opensipsctl fifo ps Process:: ID=0 PID=15438 Type=attendant Process:: ID=1 PID=15440 Type=MI FIFO Process:: ID=2 PID=15441 Type=time_keeper Process:: ID=3 PID=15442 Type=timer Process:: ID=4 PID=15444 Type=SIP receiver udp:10.2.1.52:7062 Process:: ID=5 PID=15445 Type=SIP receiver udp:10.2.1.52:7062 Process:: ID=6 PID=15446 Type=SIP receiver udp:XXX.XXX.XXX.XXX:7062 Process:: ID=7 PID=15448 Type=SIP receiver udp:XXX.XXX.XXX.XXX:7062 Process:: ID=8 PID=15449 Type=TCP receiver Process:: ID=9 PID=15451 Type=TCP receiver Process:: ID=10 PID=15452 Type=TCP receiver Process:: ID=11 PID=15454 Type=TCP receiver Process:: ID=12 PID=15455 Type=TCP receiver Process:: ID=13 PID=15457 Type=TCP receiver Process:: ID=14 PID=15458 Type=TCP receiver Process:: ID=15 PID=15466 Type=TCP receiver Process:: ID=16 PID=15467 Type=Timer handler Process:: ID=17 PID=15468 Type=TCP main [root at sbc sbin]# -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608136.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From goley_ev_sp at mail.ru Wed Jul 26 11:45:49 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Wed, 26 Jul 2017 08:45:49 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <1501076606080-7608136.post@n2.nabble.com> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> <1500907583776-7608091.post@n2.nabble.com> <1501072866468-7608134.post@n2.nabble.com> <1501074132730-7608135.post@n2.nabble.com> <1501076606080-7608136.post@n2.nabble.com> Message-ID: <1501083949069-7608138.post@n2.nabble.com> I hope someone will help in the future when a similar situation arises. Found the reason for the leak of private memory (pkmem). When the memory ends in the logs of the message, a message is displayed. 2017-07-26T03: 42: 10.1251 B2BUA INFO: core: fm_malloc: unable to allocate a large enough fragment! 2017-07-26T03: 42: 10.1252 B2BUA ERROR: core: do_action: memory allocation failure 2017-07-26T03: 42: 10.1253 B2BUA ERROR: core: pv_set_ruri_port: do action failed 2017-07-26T03: 42: 10.1254 B2BUA ERROR: core: do_assign: setting PV failed 2017-07-26T03: 42: 10.1255 B2BUA ERROR: core: do_assign: error at / usr/local/opensips231//etc/opensips/opensips.cfg:221 2017-07-26T03: 42: 10.1256 B2BUA ERROR: core: fm_malloc: not enough free pkg memory (0 bytes left, need 64), please increase the "-M" command line parameter! 2017-07-26T03: 42: 10.1257 B2BUA INFO: core: fm_malloc: attempting defragmentation ... I was wondering why he displays an error in line 221. Here it is: $rd = $avp(dest_domain); $rp = $ avp(dest_port); Where I set the values ​​above $avp dest_domain) = $dd; $avp(dest_port) = $dp; I commented out the line: #$rd = $avp(dest_domain); $rp = $ avp (dest_port); Pointed out to use for private memory (pkmem) 1 MB and restarted opensips. Before loading, I looked at the statistics [Root @ sbc sbin] # ./opensipsctl fifo get_statistics pkmem: | Grep pkmem: [45] Pkmem: 4-total_size :: 1048576 Pkmem: 4-used_size :: 103224 Pkmem: 4-real_used_size :: 146368 Pkmem: 4-max_used_size :: 146368 Pkmem: 4-free_size :: 902208 Pkmem: 4-fragments :: 393 Pkmem: 5-total_size :: 1048576 Pkmem: 5-used_size :: 103224 Pkmem: 5-real_used_size :: 146368 Pkmem: 5-max_used_size :: 146368 Pkmem: 5-free_size :: 902208 Pkmem: 5-fragments :: 393 Load included: [Root @ sbc sbin] # ./opensipsctl fifo get_statistics pkmem: | Grep pkmem: [45] Pkmem: 4-total_size :: 1048576 Pkmem: 4-used_size :: 103272 Pkmem: 4-real_used_size :: 147976 Pkmem: 4-max_used_size :: 156968 Pkmem: 4-free_size :: 900600 Pkmem: 4-fragments :: 395 Pkmem: 5-total_size :: 1048576 Pkmem: 5-used_size :: 103224 Pkmem: 5-real_used_size :: 146368 Pkmem: 5-max_used_size :: 146368 Pkmem: 5-free_size :: 902208 Pkmem: 5-fragments :: 393 .... [Root @ sbc sbin] # ./opensipsctl fifo get_statistics pkmem: | Grep pkmem: [45] Pkmem: 4-total_size :: 1048576 Pkmem: 4-used_size :: 103272 Pkmem: 4-real_used_size :: 148024 Pkmem: 4-max_used_size :: 157056 Pkmem: 4-free_size :: 900552 Pkmem: 4-fragments :: 395 Pkmem: 5-total_size :: 1048576 Pkmem: 5-used_size :: 103224 Pkmem: 5-real_used_size :: 146368 Pkmem: 5-max_used_size :: 146368 Pkmem: 5-free_size :: 902208 Pkmem: 5-fragments :: 393 [Root @ sbc sbin] # .... [Root @ sbc sbin] # ./opensipsctl fifo get_statistics pkmem: | Grep pkmem: [45] Pkmem: 4-total_size :: 1048576 Pkmem: 4-used_size :: 103272 Pkmem: 4-real_used_size :: 148024 Pkmem: 4-max_used_size :: 157056 Pkmem: 4-free_size :: 900552 Pkmem: 4-fragments :: 395 Pkmem: 5-total_size :: 1048576 Pkmem: 5-used_size :: 103280 Pkmem: 5-real_used_size :: 147624 Pkmem: 5-max_used_size :: 155416 Pkmem: 5-free_size :: 900952 Pkmem: 5-fragments :: 395 .... [Root @ sbc sbin] # ./opensipsctl fifo get_statistics pkmem: | Grep pkmem: [45] Pkmem: 4-total_size :: 1048576 Pkmem: 4-used_size :: 103272 Pkmem: 4-real_used_size :: 148024 Pkmem: 4-max_used_size :: 157056 Pkmem: 4-free_size :: 900552 Pkmem: 4-fragments :: 395 Pkmem: 5-total_size :: 1048576 Pkmem: 5-used_size :: 103280 Pkmem: 5-real_used_size :: 147984 Pkmem: 5-max_used_size :: 157256 Pkmem: 5-free_size :: 900592 Pkmem: 5-fragments :: 395 [Root @ sbc sbin] # For 1100 seconds opensips successfully processed 11500 calls, statistics [Root @ sbc sbin] # ./opensipsctl fifo get_statistics pkmem: | Grep pkmem: [45] Pkmem: 4-total_size :: 1048576 Pkmem: 4-used_size :: 103272 Pkmem: 4-real_used_size :: 148024 Pkmem: 4-max_used_size :: 157064 Pkmem: 4-free_size :: 900552 Pkmem: 4-fragments :: 395 Pkmem: 5-total_size :: 1048576 Pkmem: 5-used_size :: 103280 Pkmem: 5-real_used_size :: 148080 Pkmem: 5-max_used_size :: 157256 Pkmem: 5-free_size :: 900496 Pkmem: 5-fragments :: 395 [Root @ sbc sbin] # The reason was in assigning the value of rd and rp. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608138.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From dawid.mielnik at gmail.com Wed Jul 26 12:11:55 2017 From: dawid.mielnik at gmail.com (Dawid Mielnik) Date: Wed, 26 Jul 2017 18:11:55 +0200 Subject: [OpenSIPS-Users] ERROR:core:proto_udp_send Message-ID: Hi All, Can you help me understand the cause of these error messages ? [...] Jul 26 09:56:00.259767 ERR 32258 ERROR:core:proto_udp_send: sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily unavailable(11) Jul 26 09:56:00.259813 ERR 32258 ERROR:core:proto_udp_send: sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily unavailable(11) Jul 26 09:56:00.259822 ERR 32258 ERROR:core:proto_udp_send: sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily unavailable(11) Jul 26 09:56:00.259831 ERR 32258 ERROR:core:proto_udp_send: sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily unavailable(11) Jul 26 09:56:00.259839 ERR 32258 ERROR:core:proto_udp_send: sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily unavailable(11) Jul 26 09:56:00.259847 ERR 32258 ERROR:core:proto_udp_send: sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily unavailable(11) [...] Opensips 2.2.2. I am also using rest_client - for each call, before being proxied, an http request is made by rest_client. Before and during this time the http server was overloaded and took quite long time. I am suspecting that this is linked to it but would like to understand how exactly. Thanks, Dawid -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jul 26 13:09:13 2017 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 26 Jul 2017 20:09:13 +0300 Subject: [OpenSIPS-Users] ERROR:core:proto_udp_send In-Reply-To: References: Message-ID: <1baaf52d-6b39-1bfa-89a9-1f828bb931b8@opensips.org> Hi Dawid, This seems to have been fixed a long time ago [1], so I'm a bit puzzled. Just to be 100% sure, can you try to locate the exact binary path your OpenSIPS service is running on, and figure out the exact version that way? (e.g. output of "/usr/local/sbin/opensips -V") [1]: https://github.com/OpenSIPS/opensips/commit/d3aaf44c Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 26.07.2017 19:11, Dawid Mielnik wrote: > Hi All, > > Can you help me understand the cause of these error messages ? > > [...] > Jul 26 09:56:00.259767 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource > temporarily unavailable(11) > Jul 26 09:56:00.259813 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource > temporarily unavailable(11) > Jul 26 09:56:00.259822 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource > temporarily unavailable(11) > Jul 26 09:56:00.259831 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource > temporarily unavailable(11) > Jul 26 09:56:00.259839 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource > temporarily unavailable(11) > Jul 26 09:56:00.259847 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource > temporarily unavailable(11) > [...] > > > Opensips 2.2.2. > I am also using rest_client - for each call, before being proxied, an > http request is made by rest_client. Before and during this time the > http server was overloaded and took quite long time. I am suspecting > that this is linked to it but would like to understand how exactly. > > Thanks, > Dawid > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From dawid.mielnik at gmail.com Wed Jul 26 13:21:05 2017 From: dawid.mielnik at gmail.com (Dawid Mielnik) Date: Wed, 26 Jul 2017 19:21:05 +0200 Subject: [OpenSIPS-Users] ERROR:core:proto_udp_send In-Reply-To: <1baaf52d-6b39-1bfa-89a9-1f828bb931b8@opensips.org> References: <1baaf52d-6b39-1bfa-89a9-1f828bb931b8@opensips.org> Message-ID: Hi Liviu, Exact version info: version: opensips 2.2.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 9eb177d main.c compiled on 11:43:55 Nov 3 2016 with gcc 4.4.7 BR, Dawid On Wed, Jul 26, 2017 at 7:09 PM, Liviu Chircu wrote: > Hi Dawid, > > This seems to have been fixed a long time ago [1], so I'm a bit puzzled. > Just to be 100% sure, can you try to locate the exact binary path your > OpenSIPS service is running on, and figure out the exact version that way? > (e.g. output of "/usr/local/sbin/opensips -V") > > [1]: https://github.com/OpenSIPS/opensips/commit/d3aaf44c > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 26.07.2017 19:11, Dawid Mielnik wrote: > > Hi All, > > Can you help me understand the cause of these error messages ? > > [...] > Jul 26 09:56:00.259767 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily > unavailable(11) > Jul 26 09:56:00.259813 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily > unavailable(11) > Jul 26 09:56:00.259822 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily > unavailable(11) > Jul 26 09:56:00.259831 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily > unavailable(11) > Jul 26 09:56:00.259839 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily > unavailable(11) > Jul 26 09:56:00.259847 ERR 32258 ERROR:core:proto_udp_send: > sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource temporarily > unavailable(11) > [...] > > > Opensips 2.2.2. > I am also using rest_client - for each call, before being proxied, an http > request is made by rest_client. Before and during this time the http server > was overloaded and took quite long time. I am suspecting that this is > linked to it but would like to understand how exactly. > > Thanks, > Dawid > > > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From scott at gotbrew.org Wed Jul 26 17:37:30 2017 From: scott at gotbrew.org (Scott Fertig) Date: Wed, 26 Jul 2017 17:37:30 -0400 (EDT) Subject: [OpenSIPS-Users] Small problem with opensips-cp MI Message-ID: <724076238.1647.1501105050122.JavaMail.zimbra@gotbrew.org> HI, first I apologize if this has been gone over already but I have been searching and have not been able to find an answer as of yet. Simply my problem is that I cannot seem to get the MI connection setup properly in opensips-cp. I have tried json as well as FIFO with no success and was wondering if someone could point out what I may be doing wrong here. I am using Opensips 2.2.5 and Opensips-cp 6.2. When trying json my opensips-cp boxes.global.inc.php MI section looks as so: $boxes[$box_id]['mi']['conn']="json:127.0.0.1:8888/JSON"; I have mi_json and json loaded after the http and httpd modules in opensips.cfg: #### HTTPD module loadmodule "httpd.so" modparam("httpd", "port", 8888) #### MI_HTTP module loadmodule "json.so" loadmodule "mi_http.so" loadmodule "mi_json.so" When I try to use the MI commands section in the control panel, I issue a command and get back the error: "MI command failed with 400" Also I can see opensips listening on port 8888: tcp 0 0 0.0.0.0:8888 0.0.0.0:* LISTEN 15232/opensips When I try to use fifo I get a different error, my boxes config looks like: $boxes[$box_id]['mi']['conn']="fifo:/tmp/opensips_fifo"; I can verify that the file exits in /tmp with the correct name and that the fifo module is loaded: ls -al /tmp/opensips_fifo prw-rw-rw- 1 root root 0 Jul 26 16:40 /tmp/opensips_fifo #### FIFO Management Interface loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") modparam("mi_fifo", "fifo_mode", 0666) When trying to use fifo I get this error: Unknwon/Unsupported type[fifo] for MI URL Hopefully someone can point me in the right direction, I'm not exactly sure where to go next on this problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: From scott at gotbrew.org Wed Jul 26 18:08:14 2017 From: scott at gotbrew.org (Scott Fertig) Date: Wed, 26 Jul 2017 18:08:14 -0400 (EDT) Subject: [OpenSIPS-Users] Small problem with opensips-cp MI In-Reply-To: <724076238.1647.1501105050122.JavaMail.zimbra@gotbrew.org> References: <724076238.1647.1501105050122.JavaMail.zimbra@gotbrew.org> Message-ID: <800357350.1663.1501106894120.JavaMail.zimbra@gotbrew.org> I apologize, I just found the issue on my own. Looks like it was because I left the default "JSON" in the json MI portion of the config, but I just found in the documentation the default is "json". I changed to all lower case and the issue is resolved. From: "Scott Fertig" To: "users" Sent: Wednesday, July 26, 2017 5:37:30 PM Subject: [OpenSIPS-Users] Small problem with opensips-cp MI HI, first I apologize if this has been gone over already but I have been searching and have not been able to find an answer as of yet. Simply my problem is that I cannot seem to get the MI connection setup properly in opensips-cp. I have tried json as well as FIFO with no success and was wondering if someone could point out what I may be doing wrong here. I am using Opensips 2.2.5 and Opensips-cp 6.2. When trying json my opensips-cp boxes.global.inc.php MI section looks as so: $boxes[$box_id]['mi']['conn']="json:127.0.0.1:8888/JSON"; I have mi_json and json loaded after the http and httpd modules in opensips.cfg: #### HTTPD module loadmodule "httpd.so" modparam("httpd", "port", 8888) #### MI_HTTP module loadmodule "json.so" loadmodule "mi_http.so" loadmodule "mi_json.so" When I try to use the MI commands section in the control panel, I issue a command and get back the error: "MI command failed with 400" Also I can see opensips listening on port 8888: tcp 0 0 0.0.0.0:8888 0.0.0.0:* LISTEN 15232/opensips When I try to use fifo I get a different error, my boxes config looks like: $boxes[$box_id]['mi']['conn']="fifo:/tmp/opensips_fifo"; I can verify that the file exits in /tmp with the correct name and that the fifo module is loaded: ls -al /tmp/opensips_fifo prw-rw-rw- 1 root root 0 Jul 26 16:40 /tmp/opensips_fifo #### FIFO Management Interface loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") modparam("mi_fifo", "fifo_mode", 0666) When trying to use fifo I get this error: Unknwon/Unsupported type[fifo] for MI URL Hopefully someone can point me in the right direction, I'm not exactly sure where to go next on this problem. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Wed Jul 26 17:05:00 2017 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 26 Jul 2017 18:05:00 -0300 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: References: <20170720184547.GA11475@tlaquepaque.localdomain> <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> Message-ID: Hi Bogdan So on transaction accounting, the record is written after transaction receives final response. So it's not written as soon I receive BYE, but when I receive the 200OK of the BYE. My customer is complaining that the call is taking 200ms more and our system is charging 1 sec more than on his billing. On million calls, this is generating a some thousands difference. Thanks for the information! Regards On Tue, Jul 25, 2017 at 6:36 AM, Bogdan-Andrei Iancu wrote: > Hi Daniel, > > There are 3 types of accountings in OpenSIPS - per message, per > transaction, per dialog. > > For the per message, it is clear :) . When doing per-transaction > accounting, the ACC record is written when the transaction is completed > with a final response (>=200) on the UAS side (towards caller). For the > dialog based accounting, the time reference (for ending the call) is the > reception of BYE request; still the CDR is written on the BYE final reply > (as OpenSIPS allows you to collect CDR info from the BYE replies too). > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/20/2017 10:49 PM, Daniel Zanutti wrote: > > Hi Alex > > I'm having a billing problem from receiving BYE to 200 OK is taking more > than 500ms. If BYE is accounted when it's received, great! > > Are you absolutely sure it works this way? > > Thanks > > On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov > wrote: > >> My understanding is that this is a rather simple module without >> sophisticated state componentry, and that it logs things immediately as >> received, in the same iteration of message processing. >> >> -- Alex >> >> -- >> Principal, Evariste Systems LLC (www.evaristesys.com) >> >> Sent from my Google Nexus. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eahaselhoff at gmail.com Thu Jul 27 03:04:48 2017 From: eahaselhoff at gmail.com (Edwin) Date: Thu, 27 Jul 2017 00:04:48 -0700 (MST) Subject: [OpenSIPS-Users] clusterer sync node Message-ID: <1501139088595-7608149.post@n2.nabble.com> I build a opensips HA cluster with a b2b front-end and two back-end nodes. The front-end connects to the clients, handles the registrations and load balance the invites with a 'b2b_init_request("top hiding/a")' to one of the two back-end nodes. The backend nodes do the actual processing, accounting etc. If I take one of the backend nodes out of production (for maybe a config update) and bring it up again, it has to 'learn' al the sessions and registrations (usrloc, dialog) again. Question: Is it posible to do a sync the clusterer node on startup with the front node so all the registrations are in memory again (or should i add a feature request?) Gr. Edwin -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/clusterer-sync-node-tp7608149.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From razvan at opensips.org Thu Jul 27 05:07:27 2017 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 27 Jul 2017 12:07:27 +0300 Subject: [OpenSIPS-Users] ERROR:core:proto_udp_send In-Reply-To: References: <1baaf52d-6b39-1bfa-89a9-1f828bb931b8@opensips.org> Message-ID: Hi, Dawid! I think the error appears when the Networking send buffers are full. If your system has enough resources, you can try to increase the memory used by the buffers[1]. [1] https://www.cyberciti.biz/faq/linux-tcp-tuning/ Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 07/26/2017 08:21 PM, Dawid Mielnik wrote: > Hi Liviu, > > Exact version info: > > version: opensips 2.2.2 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 9eb177d > main.c compiled on 11:43:55 Nov 3 2016 with gcc 4.4.7 > > BR, > Dawid > > On Wed, Jul 26, 2017 at 7:09 PM, Liviu Chircu > wrote: > > Hi Dawid, > > This seems to have been fixed a long time ago [1], so I'm a bit > puzzled. Just to be 100% sure, can you try to locate the exact > binary path your OpenSIPS service is running on, and figure out > the exact version that way? (e.g. output of > "/usr/local/sbin/opensips -V") > > [1]: https://github.com/OpenSIPS/opensips/commit/d3aaf44c > > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 26.07.2017 19:11, Dawid Mielnik wrote: >> Hi All, >> >> Can you help me understand the cause of these error messages ? >> >> [...] >> Jul 26 09:56:00.259767 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259813 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259822 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259831 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259839 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259847 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> [...] >> >> >> Opensips 2.2.2. >> I am also using rest_client - for each call, before being >> proxied, an http request is made by rest_client. Before and >> during this time the http server was overloaded and took quite >> long time. I am suspecting that this is linked to it but would >> like to understand how exactly. >> >> Thanks, >> Dawid >> >> >> >> _______________________________________________ >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From dawid.mielnik at gmail.com Thu Jul 27 05:58:39 2017 From: dawid.mielnik at gmail.com (Dawid Mielnik) Date: Thu, 27 Jul 2017 11:58:39 +0200 Subject: [OpenSIPS-Users] ERROR:core:proto_udp_send In-Reply-To: References: <1baaf52d-6b39-1bfa-89a9-1f828bb931b8@opensips.org> Message-ID: Hi Liviu, Thanks for your answer! I am still somehow trying to relate this to the rest_client problems context as it happened in exact same time (the errors have not been oberved in other situations even with higher traffic). I do not know internals of OpenSIPS - but is it possible that this is somehow related to receive buffers ? That is, OpenSIPS threads are busy waiting for rest_get responses (which have long delays) which leads to slow processing of incoming traffic and hence these errors ? Or is this not related in any way? Thanks, Dawid On Thu, Jul 27, 2017 at 11:07 AM, Răzvan Crainea wrote: > Hi, Dawid! > > I think the error appears when the Networking send buffers are full. If > your system has enough resources, you can try to increase the memory used > by the buffers[1]. > > [1] https://www.cyberciti.biz/faq/linux-tcp-tuning/ > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 07/26/2017 08:21 PM, Dawid Mielnik wrote: > > Hi Liviu, > > Exact version info: > > version: opensips 2.2.2 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 9eb177d > main.c compiled on 11:43:55 Nov 3 2016 with gcc 4.4.7 > > BR, > Dawid > > On Wed, Jul 26, 2017 at 7:09 PM, Liviu Chircu wrote: > >> Hi Dawid, >> >> This seems to have been fixed a long time ago [1], so I'm a bit puzzled. >> Just to be 100% sure, can you try to locate the exact binary path your >> OpenSIPS service is running on, and figure out the exact version that way? >> (e.g. output of "/usr/local/sbin/opensips -V") >> >> [1]: https://github.com/OpenSIPS/opensips/commit/d3aaf44c >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 26.07.2017 19:11, Dawid Mielnik wrote: >> >> Hi All, >> >> Can you help me understand the cause of these error messages ? >> >> [...] >> Jul 26 09:56:00.259767 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259813 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259822 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259831 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259839 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> Jul 26 09:56:00.259847 ERR 32258 ERROR:core:proto_udp_send: >> sendto(sock,0x7fb4817b3ba8,563,0,0x7fb484c23f10,16): Resource >> temporarily unavailable(11) >> [...] >> >> >> Opensips 2.2.2. >> I am also using rest_client - for each call, before being proxied, an >> http request is made by rest_client. Before and during this time the http >> server was overloaded and took quite long time. I am suspecting that this >> is linked to it but would like to understand how exactly. >> >> Thanks, >> Dawid >> >> >> >> _______________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jul 27 09:48:58 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 27 Jul 2017 16:48:58 +0300 Subject: [OpenSIPS-Users] 2 weeks left to OpenSIPS Training @ ClueCon Message-ID: Hi all, AS you already know, this year we are running again an OpenSIPS training @ ClueCon 2017 - about how to do front-ending and clustering for FreeSWITCH system, including some of the latest cool features available in OpenSIPS 2.3, like mid-registration for registration throttling and aggregation, FreeSWITCH driven load balancing and more. This year will also show how to use the OpenSIPS front-end to add more services (like presence and BLF) to your system, independently to FreeSWITCH. See full description here https://goo.gl/7NtYVF . So, do not mis it. Available seats are getting low, so get your registration done via https://freeswitch.com/cart.php Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html From jhchinn at thenavisway.com Thu Jul 27 15:03:49 2017 From: jhchinn at thenavisway.com (jhchinn) Date: Thu, 27 Jul 2017 12:03:49 -0700 (MST) Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display Message-ID: <1501182229494-7608166.post@n2.nabble.com> I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 following the directions from http://controlpanel.opensips.org/htmldoc/INSTALL.html.Everything loaded and I can get to the CP and all frames via multiple browsers.The display for the User Management doesn't display anything while all others do. Any ideas why and what to do to fix it?Thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/User-Management-frame-in-OpenSIPS-CP-has-no-display-tp7608166.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wsimon at stratusvideo.com Thu Jul 27 20:14:23 2017 From: wsimon at stratusvideo.com (William Simon) Date: Fri, 28 Jul 2017 00:14:23 +0000 Subject: [OpenSIPS-Users] Multiple rtpproxy on same server, same network interfaces Message-ID: <8051674B-0EA4-4552-BD18-FE1104DA9DC7@stratusvideo.com> I am using multi-core processors (who isn't) and want to get the most out of opensips + rtpproxy running on the same server. According to opensips docs I can tell opensips to load balance between two instances of rtpproxy on the same machine, controlled through different UDP sockets: (From http://www.opensips.org/html/docs/modules/2.2.x/rtpproxy.html) # multiple rtproxies for LB modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221 udp:localhost:12222") Using rtpproxy 2.0 it looks like I should have two cores per rtpproxy. Is it enough then to set up (CORES / 2) instances of rtpproxy, each with the same parameters on the server but different control sockets, and then tell opensips about them using the rtpproxy load balance syntax shown above? Do they need to be assigned different RTP ranges, IP addresses or anything like that? I have set up a test box as I just described but cannot tell whether I will have resource conflicts under load. "The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer." -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jul 28 06:42:14 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 13:42:14 +0300 Subject: [OpenSIPS-Users] OpenSIPS reseting issue with $T_fr_inv_timeout while forwarding In-Reply-To: References: <807e815d-060b-598d-89bb-511bf2a0b70c@opensips.org> Message-ID: <8f2dfaa4-b7fd-719e-3317-0f9acb5a7bc8@opensips.org> Hello Ravi, It looks like the failed call to 3333 has no failure route set (this is why you directly get the 408 default timeout, as there is no failure route to handle the timeout event). Are you sure you t_on_failure() for each call attempt (especially when handling the call failure to 2222) ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/26/2017 04:35 PM, Ravi Patel wrote: > Dear Bogdan, > > I am Grateful for your reply. > > I applied *$T_fr_inv_timeout* before doing each *t_relay().* by > applying it , I am able to achieve it at 1st forwarding but > unfortunately not working for 2nd forwarding. > > The scenario is: > 1111 > 2222 (fr_inv_timeout 10 sec) > 3333 (fr_inv_timeout 5 sec) > 4444 (fr_inv_timeout 20 sec) > > when 1111 calls 2222 : OpenSIPS generates CANCEL at 10 secs and > forwards call to 3333. > now --> 3333 : OpenSIPS generates CANCEL at 5 secs**but does not > forward call to 4444 instead it sends *408 to Caller(1111)* and drops > call. > > I am attaching packets where sip.client.com > refers to the SIP clients and sip.server.com > refers to the OpenSIPS Server. > > Also find the attached snapshots of the call flow. > > Please guide what can be done or where I am doing wrong ? > Let me know if you need any other information. > > Best Regards, > Ravi Patel > > > > > On Tue, Jul 25, 2017 at 9:07 PM, Bogdan-Andrei Iancu > > wrote: > > Hi Ravi, > > Before each t_rely() you have to set the your custom > $T_fr_inv_timeout and $T_fr_timeout, otherwise the default values > will be used. As you have a serial forking scenario, you do a new > t_relay() at each step. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/25/2017 05:34 PM, Ravi Patel wrote: >> Hi Team, >> >> What is the right way to reset timers *$T_fr_inv_timeout* and >> *$T_fr_timeout* ?? >> >> I am using OpenSIPS-2.2 version >> The below scenario will help to understand issue, >> >> There are 4 SIP users, >> 1111,2222,3333,4444 >> >> What I want to achieve is: >> 1111 ---> 2222 (FORWARD ON NOANSWER) ---> 3333 (FORWARD ON >> NOANSWER) ---> 4444 >> >> *1st Test Case Scenario:* >> >> 1111 >> 2222 (fr_inv_timeout 20 sec) >> 3333 (fr_inv_timeout 25 sec) >> 4444 (fr_inv_timeout 30 sec) >> >> >> when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs >> (thats working proper as expexted) and forwards call to 3333 as >> per my configuration. >> so in --> 3333 : OpenSIPS generates CANCEL at *20 secs instead of >> 25 secs* and send 408 to 1111. and not processing the 2nd forwarding. >> >> *2nd Test Case Scenario:* >> 1111 >> 2222 (fr_inv_timeout 20 sec) >> 3333 (fr_inv_timeout 15 sec) >> 4444 (fr_inv_timeout 30 sec) >> >> when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (that >> is working proper as expexted) and forwards call to 3333 as per >> my configuration. >> now --> 3333 : OpenSIPS generates CANCEL at 15 secs and forwards >> the call to 4444, Here OpenSIPS generates CANCEL *after 5 secs >> instead of 30 secs.* >> >> >> We set timeout by using $T_fr_inv_timeout. >> ------------ >> route[ring_timeout]{ >> xlog("L_INFO","------------------- RING_TIMEOUT ---------------\n"); >> if (!is_method("INVITE")) >> return; >> avp_db_load("$rU","$avp(ringtimeout)/usr_preferences"); >> if($avp(ringtimeout)!=null) >> { >> $T_fr_inv_timeout = NULL; >> xlog("L_INFO","$rU: Ring timeout : >> $avp(ringtimeout)"); >> $T_fr_inv_timeout >> =$(avp(ringtimeout){s.int }) ; >> xlog("L_INFO","$rU: Ring timeout is >> setted: [$T_fr_inv_timeout]"); >> } >> else >> { >> xlog("L_INFO","$rU: Ring timeout is NOT >> setted"); >> } >> } >> ------------------ >> >> From both the scenarios what we found, it sticks to the first >> timeout of 2222,that is 20secs in our case. >> In first scenario it generates CANCEL on 3333 at 20 secs instead >> of 25 that is 2222's Timeout. >> In second scenario it generates CANCEL on 3333 at 15sec and on >> 4444 at 5 sec (15 + 5 = 20 sec) that is also 2222's timeout. >> >> >> Can I know the right method to set $T_fr_inv_timeout ? >> >> Let me know if any other information is needed. >> >> >> Thanks, >> Ravi >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jul 28 06:58:41 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 13:58:41 +0300 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <1501182229494-7608166.post@n2.nabble.com> References: <1501182229494-7608166.post@n2.nabble.com> Message-ID: <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> Hello, Have you configured the DB support, so that the User Management can list the users ? Do you see any errors in the apache logs ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/27/2017 10:03 PM, jhchinn wrote: > I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 following > the directions from > http://controlpanel.opensips.org/htmldoc/INSTALL.html. Everything > loaded and I can get to the CP and all frames via multiple browsers. > The display for the User Management doesn't display anything while all > others do. Any ideas why and what to do to fix it? Thanks > ------------------------------------------------------------------------ > View this message in context: User Management frame in OpenSIPS-CP has > no display > > Sent from the OpenSIPS - Users mailing list archive > > at Nabble.com. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jul 28 07:01:06 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 14:01:06 +0300 Subject: [OpenSIPS-Users] Accounting of 200 OK and BYE In-Reply-To: References: <20170720184547.GA11475@tlaquepaque.localdomain> <27A3D6D1-6475-49B5-B9E8-EE511F814B48@evaristesys.com> Message-ID: <49fa4c4c-f1e7-5295-0e0c-eb3ad3a90114@opensips.org> Hi Daniel, Yes, transaction is on the final response - fro INVITE you want to account when the call was accepted, not when it was dialed. If you want for the BYE to account when the request was received, use extra accounting to store the timestamp (ms also) of receiving the BYE (via script operations). When you correlate the INVITE with the BYE, you will calculate the duration from the 200 OK INVITE to the BYE request. Or much simpler just use dialog based accounting which is already doing this. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/27/2017 12:05 AM, Daniel Zanutti wrote: > Hi Bogdan > > So on transaction accounting, the record is written after transaction > receives final response. So it's not written as soon I receive BYE, > but when I receive the 200OK of the BYE. > > My customer is complaining that the call is taking 200ms more and our > system is charging 1 sec more than on his billing. On million calls, > this is generating a some thousands difference. > > Thanks for the information! > > Regards > > On Tue, Jul 25, 2017 at 6:36 AM, Bogdan-Andrei Iancu > > wrote: > > Hi Daniel, > > There are 3 types of accountings in OpenSIPS - per message, per > transaction, per dialog. > > For the per message, it is clear :) . When doing per-transaction > accounting, the ACC record is written when the transaction is > completed with a final response (>=200) on the UAS side (towards > caller). For the dialog based accounting, the time reference (for > ending the call) is the reception of BYE request; still the CDR is > written on the BYE final reply (as OpenSIPS allows you to collect > CDR info from the BYE replies too). > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 07/20/2017 10:49 PM, Daniel Zanutti wrote: >> Hi Alex >> >> I'm having a billing problem from receiving BYE to 200 OK is >> taking more than 500ms. If BYE is accounted when it's received, >> great! >> >> Are you absolutely sure it works this way? >> >> Thanks >> >> On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov >> > wrote: >> >> My understanding is that this is a rather simple module >> without sophisticated state componentry, and that it logs >> things immediately as received, in the same iteration of >> message processing. >> >> -- Alex >> >> -- >> Principal, Evariste Systems LLC (www.evaristesys.com >> ) >> >> Sent from my Google Nexus. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jul 28 07:04:38 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 14:04:38 +0300 Subject: [OpenSIPS-Users] Multiple rtpproxy on same server, same network interfaces In-Reply-To: <8051674B-0EA4-4552-BD18-FE1104DA9DC7@stratusvideo.com> References: <8051674B-0EA4-4552-BD18-FE1104DA9DC7@stratusvideo.com> Message-ID: <855d8fce-0c34-0bb2-363d-2194b48651d4@opensips.org> Hello William, In 2.0 there are indeed multiple threads, but as worker (doing the RTP relay) is still one. The rest of the treads are light processing ones. So I would keep a one to one mapping, IMHO. If multiple rtpproxies are using the IP for RTP relay, then you have to partition the port rage to avoid overlapping between them. If there are different IPs, you do not need to do this. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 03:14 AM, William Simon wrote: > I am using multi-core processors (who isn't) and want to get the most > out of opensips + rtpproxy running on the same server. > > According to opensips docs I can tell opensips to load balance between > two instances of rtpproxy on the same machine, controlled through > different UDP sockets: > > (From http://www.opensips.org/html/docs/modules/2.2.x/rtpproxy.html) > > # multiple rtproxies for LB > modparam("rtpproxy", "rtpproxy_sock", > "udp:localhost:12221 udp:localhost:12222") > > > Using rtpproxy 2.0 it looks like I should have two cores per rtpproxy. > Is it enough then to set up (CORES / 2) instances of rtpproxy, each > with the same parameters on the server but different control sockets, > and then tell opensips about them using the rtpproxy load balance > syntax shown above? > > Do they need to be assigned different RTP ranges, IP addresses or > anything like that? I have set up a test box as I just described but > cannot tell whether I will have resource conflicts under load. > > > “The information transmitted is intended only for the person or entity > to which it is addressed and may contain proprietary, > business-confidential and/or privileged material. If you are not the > intended recipient of this message you are hereby notified that any > use, review, retransmission, dissemination, distribution, reproduction > or any action taken in reliance upon this message is prohibited. If > you received this in error, please contact the sender and delete the > material from any computer.” > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From wsimon at stratusvideo.com Fri Jul 28 09:41:16 2017 From: wsimon at stratusvideo.com (William Simon) Date: Fri, 28 Jul 2017 13:41:16 +0000 Subject: [OpenSIPS-Users] Multiple rtpproxy on same server, same network interfaces In-Reply-To: <855d8fce-0c34-0bb2-363d-2194b48651d4@opensips.org> References: <8051674B-0EA4-4552-BD18-FE1104DA9DC7@stratusvideo.com> <855d8fce-0c34-0bb2-363d-2194b48651d4@opensips.org> Message-ID: <9600020E-6F27-45F6-8401-4A1ED9A6E594@stratusvideo.com> Thanks for this advice. I didn't see any clear instructions on the web about how to do this. My intent is to use multiple rtpproxies on the same IPs, so I will partition the RTP port range as suggested. On Jul 28, 2017, at 7:04 AM, Bogdan-Andrei Iancu > wrote: Hello William, In 2.0 there are indeed multiple threads, but as worker (doing the RTP relay) is still one. The rest of the treads are light processing ones. So I would keep a one to one mapping, IMHO. If multiple rtpproxies are using the IP for RTP relay, then you have to partition the port rage to avoid overlapping between them. If there are different IPs, you do not need to do this. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 03:14 AM, William Simon wrote: I am using multi-core processors (who isn't) and want to get the most out of opensips + rtpproxy running on the same server. According to opensips docs I can tell opensips to load balance between two instances of rtpproxy on the same machine, controlled through different UDP sockets: (From http://www.opensips.org/html/docs/modules/2.2.x/rtpproxy.html) # multiple rtproxies for LB modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221 udp:localhost:12222") Using rtpproxy 2.0 it looks like I should have two cores per rtpproxy. Is it enough then to set up (CORES / 2) instances of rtpproxy, each with the same parameters on the server but different control sockets, and then tell opensips about them using the rtpproxy load balance syntax shown above? Do they need to be assigned different RTP ranges, IP addresses or anything like that? I have set up a test box as I just described but cannot tell whether I will have resource conflicts under load. “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” -------------- next part -------------- An HTML attachment was scrubbed... URL: From tribest at gmail.com Tue Jul 25 00:08:19 2017 From: tribest at gmail.com (Nickylin) Date: Mon, 24 Jul 2017 21:08:19 -0700 (MST) Subject: [OpenSIPS-Users] problem with parallel forking Message-ID: <1500955699787-7608097.post@n2.nabble.com> I know the parallel forking it means an INVITE comes into Opensips. Opensips fork multiple INVITE and sends them out to multiple places at once. Once it receives a 200 OK from one of those places, it sends CANCELs to the others. But I have observation , sometimes , if opensips receive more on 200OK from callee at the same , it will not send CANCEL to devices which send 200ok . SO , it means more than one callee's status is connection , but only one caller. Have any solution , callee can receive BYE , if opensips receive 200OK at the same time. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/problem-with-parallel-forking-tp7608097.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From devang.nathwani31589 at gmail.com Wed Jul 26 07:17:08 2017 From: devang.nathwani31589 at gmail.com (devangn) Date: Wed, 26 Jul 2017 04:17:08 -0700 (MST) Subject: [OpenSIPS-Users] timer between 100 trying AND 18X ringing In-Reply-To: <1425067726211-7595524.post@n2.nabble.com> References: <1425067726211-7595524.post@n2.nabble.com> Message-ID: <1501067828614-7608133.post@n2.nabble.com> Hello, Have you resolve this issue? if yes how? i also need to wait for 18X after 100, i have tried, $avp(final_reply_timer) = 4; in external_to_internal_relay route Any help would be appreciated -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/timer-between-100-trying-AND-18X-ringing-tp7595524p7608133.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From ravi.patel at ecosmob.com Wed Jul 26 09:35:40 2017 From: ravi.patel at ecosmob.com (Ravi Patel) Date: Wed, 26 Jul 2017 19:05:40 +0530 Subject: [OpenSIPS-Users] OpenSIPS reseting issue with $T_fr_inv_timeout while forwarding In-Reply-To: <807e815d-060b-598d-89bb-511bf2a0b70c@opensips.org> References: <807e815d-060b-598d-89bb-511bf2a0b70c@opensips.org> Message-ID: Dear Bogdan, I am Grateful for your reply. I applied *$T_fr_inv_timeout* before doing each *t_relay().* by applying it , I am able to achieve it at 1st forwarding but unfortunately not working for 2nd forwarding. The scenario is: 1111 2222 (fr_inv_timeout 10 sec) 3333 (fr_inv_timeout 5 sec) 4444 (fr_inv_timeout 20 sec) when 1111 calls 2222 : OpenSIPS generates CANCEL at 10 secs and forwards call to 3333. now --> 3333 : OpenSIPS generates CANCEL at 5 secs but does not forward call to 4444 instead it sends *408 to Caller(1111)* and drops call. I am attaching packets where sip.client.com refers to the SIP clients and sip.server.com refers to the OpenSIPS Server. Also find the attached snapshots of the call flow. Please guide what can be done or where I am doing wrong ? Let me know if you need any other information. Best Regards, Ravi Patel On Tue, Jul 25, 2017 at 9:07 PM, Bogdan-Andrei Iancu wrote: > Hi Ravi, > > Before each t_rely() you have to set the your custom $T_fr_inv_timeout and > $T_fr_timeout, otherwise the default values will be used. As you have a > serial forking scenario, you do a new t_relay() at each step. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/25/2017 05:34 PM, Ravi Patel wrote: > > Hi Team, > > What is the right way to reset timers *$T_fr_inv_timeout* and > *$T_fr_timeout* ?? > > I am using OpenSIPS-2.2 version > The below scenario will help to understand issue, > > There are 4 SIP users, > 1111,2222,3333,4444 > > What I want to achieve is: > 1111 ---> 2222 (FORWARD ON NOANSWER) ---> 3333 (FORWARD ON NOANSWER) ---> > 4444 > > *1st Test Case Scenario:* > > 1111 > 2222 (fr_inv_timeout 20 sec) > 3333 (fr_inv_timeout 25 sec) > 4444 (fr_inv_timeout 30 sec) > > > when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (thats working > proper as expexted) and forwards call to 3333 as per my configuration. > so in --> 3333 : OpenSIPS generates CANCEL at *20 secs instead of 25 secs* > and send 408 to 1111. and not processing the 2nd forwarding. > > *2nd Test Case Scenario:* > 1111 > 2222 (fr_inv_timeout 20 sec) > 3333 (fr_inv_timeout 15 sec) > 4444 (fr_inv_timeout 30 sec) > > when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (that is > working proper as expexted) and forwards call to 3333 as per my > configuration. > now --> 3333 : OpenSIPS generates CANCEL at 15 secs and forwards the call > to 4444, Here OpenSIPS generates CANCEL *after 5 secs instead of 30 secs.* > > > We set timeout by using $T_fr_inv_timeout. > ------------ > route[ring_timeout]{ > xlog("L_INFO","------------------- RING_TIMEOUT > ---------------\n"); > if (!is_method("INVITE")) > return; > avp_db_load("$rU","$avp(ringtimeout)/usr_preferences"); > if($avp(ringtimeout)!=null) > { > $T_fr_inv_timeout = NULL; > xlog("L_INFO","$rU: Ring timeout : > $avp(ringtimeout)"); > $T_fr_inv_timeout =$(avp(ringtimeout){s.int}) ; > xlog("L_INFO","$rU: Ring timeout is setted: > [$T_fr_inv_timeout]"); > } > else > { > xlog("L_INFO","$rU: Ring timeout is NOT setted"); > } > } > ------------------ > > From both the scenarios what we found, it sticks to the first timeout of > 2222,that is 20secs in our case. > In first scenario it generates CANCEL on 3333 at 20 secs instead of 25 > that is 2222's Timeout. > In second scenario it generates CANCEL on 3333 at 15sec and on 4444 at 5 > sec (15 + 5 = 20 sec) that is also 2222's timeout. > > > Can I know the right method to set $T_fr_inv_timeout ? > > Let me know if any other information is needed. > > > Thanks, > Ravi > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: forwardissue_ngrep Type: application/octet-stream Size: 13919 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 1 Type: application/octet-stream Size: 79855 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 2 Type: application/octet-stream Size: 76279 bytes Desc: not available URL: From Ben.Newlin at genesys.com Fri Jul 28 11:39:55 2017 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 28 Jul 2017 15:39:55 +0000 Subject: [OpenSIPS-Users] OpenSIPS reseting issue with $T_fr_inv_timeout while forwarding In-Reply-To: References: <807e815d-060b-598d-89bb-511bf2a0b70c@opensips.org> Message-ID: Ravi, Are you sure you are arming the failure route after each step using t_on_failure? It sounds like you are only doing this on the call to 2222, which allows you to failover to 3333. But when you send to 3333 you have to arm the failure route again. Ben Newlin From: Users on behalf of Ravi Patel Reply-To: OpenSIPS users mailling list Date: Friday, July 28, 2017 at 11:36 AM To: Bogdan-Andrei Iancu Cc: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS reseting issue with $T_fr_inv_timeout while forwarding Dear Bogdan, I am Grateful for your reply. I applied $T_fr_inv_timeout before doing each t_relay(). by applying it , I am able to achieve it at 1st forwarding but unfortunately not working for 2nd forwarding. The scenario is: 1111 2222 (fr_inv_timeout 10 sec) 3333 (fr_inv_timeout 5 sec) 4444 (fr_inv_timeout 20 sec) when 1111 calls 2222 : OpenSIPS generates CANCEL at 10 secs and forwards call to 3333. now --> 3333 : OpenSIPS generates CANCEL at 5 secs but does not forward call to 4444 instead it sends 408 to Caller(1111) and drops call. I am attaching packets where sip.client.com refers to the SIP clients and sip.server.com refers to the OpenSIPS Server. Also find the attached snapshots of the call flow. Please guide what can be done or where I am doing wrong ? Let me know if you need any other information. Best Regards, Ravi Patel On Tue, Jul 25, 2017 at 9:07 PM, Bogdan-Andrei Iancu > wrote: Hi Ravi, Before each t_rely() you have to set the your custom $T_fr_inv_timeout and $T_fr_timeout, otherwise the default values will be used. As you have a serial forking scenario, you do a new t_relay() at each step. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 05:34 PM, Ravi Patel wrote: Hi Team, What is the right way to reset timers $T_fr_inv_timeout and $T_fr_timeout ?? I am using OpenSIPS-2.2 version The below scenario will help to understand issue, There are 4 SIP users, 1111,2222,3333,4444 What I want to achieve is: 1111 ---> 2222 (FORWARD ON NOANSWER) ---> 3333 (FORWARD ON NOANSWER) ---> 4444 1st Test Case Scenario: 1111 2222 (fr_inv_timeout 20 sec) 3333 (fr_inv_timeout 25 sec) 4444 (fr_inv_timeout 30 sec) when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (thats working proper as expexted) and forwards call to 3333 as per my configuration. so in --> 3333 : OpenSIPS generates CANCEL at 20 secs instead of 25 secs and send 408 to 1111. and not processing the 2nd forwarding. 2nd Test Case Scenario: 1111 2222 (fr_inv_timeout 20 sec) 3333 (fr_inv_timeout 15 sec) 4444 (fr_inv_timeout 30 sec) when 1111 calls 2222 : OpenSIPS generates CANCEL at 20 secs (that is working proper as expexted) and forwards call to 3333 as per my configuration. now --> 3333 : OpenSIPS generates CANCEL at 15 secs and forwards the call to 4444, Here OpenSIPS generates CANCEL after 5 secs instead of 30 secs. We set timeout by using $T_fr_inv_timeout. ------------ route[ring_timeout]{ xlog("L_INFO","------------------- RING_TIMEOUT ---------------\n"); if (!is_method("INVITE")) return; avp_db_load("$rU","$avp(ringtimeout)/usr_preferences"); if($avp(ringtimeout)!=null) { $T_fr_inv_timeout = NULL; xlog("L_INFO","$rU: Ring timeout : $avp(ringtimeout)"); $T_fr_inv_timeout =$(avp(ringtimeout){s.int}) ; xlog("L_INFO","$rU: Ring timeout is setted: [$T_fr_inv_timeout]"); } else { xlog("L_INFO","$rU: Ring timeout is NOT setted"); } } ------------------ From both the scenarios what we found, it sticks to the first timeout of 2222,that is 20secs in our case. In first scenario it generates CANCEL on 3333 at 20 secs instead of 25 that is 2222's Timeout. In second scenario it generates CANCEL on 3333 at 15sec and on 4444 at 5 sec (15 + 5 = 20 sec) that is also 2222's timeout. Can I know the right method to set $T_fr_inv_timeout ? Let me know if any other information is needed. Thanks, Ravi _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jul 28 13:08:38 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 20:08:38 +0300 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: References: <1501182229494-7608166.post@n2.nabble.com> <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> Message-ID: That is the typical error if you forgot turn ON the short_open_tag PHP option in php.ini for apache (not for cli). And be sure you restart apache after doing the change. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 07:52 PM, Jerry Chinn wrote: > > Bogdan, > > I found an error in the > /var/www/opensips-cp/web/tools/users/user_management/user_management.php > file > > On line 273 and 274 there is an unexpected } character. > > I removed them and I am now getting the attached display in the User > Management display > > > Your thoughts? > > *Jerry Chinn* > > *Telecom VoIP Specialist* > > *NAVIS *More Performance. More Profit. > > tel 541-330-3562 > > www.TheNavisWay.com > > Facebook | Twitter > | LinkedIn > | Blog > > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Friday, July 28, 2017 3:59 AM > *To:* OpenSIPS users mailling list; Jerry Chinn > *Subject:* Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP > has no display > > Hello, > > Have you configured the DB support, so that the User Management can > list the users ? Do you see any errors in the apache logs ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/27/2017 10:03 PM, jhchinn wrote: > > I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 > following the directions from > http://controlpanel.opensips.org/htmldoc/INSTALL.html. Everything > loaded and I can get to the CP and all frames via multiple > browsers. The display for the User Management doesn't display > anything while all others do. Image removed by sender.Any ideas > why and what to do to fix it? Thanks > > ------------------------------------------------------------------------ > > View this message in context: User Management frame in OpenSIPS-CP > has no display > > Sent from the OpenSIPS - Users mailing list archive > > at Nabble.com. > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: not available URL: From JHChinn at TheNavisWay.com Fri Jul 28 12:52:13 2017 From: JHChinn at TheNavisWay.com (Jerry Chinn) Date: Fri, 28 Jul 2017 16:52:13 +0000 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> References: <1501182229494-7608166.post@n2.nabble.com> <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> Message-ID: Bogdan, I found an error in the /var/www/opensips-cp/web/tools/users/user_management/user_management.php file On line 273 and 274 there is an unexpected } character. I removed them and I am now getting the attached display in the User Management display [cid:image001.png at 01D30787.30137FC0] Your thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 3:59 AM To: OpenSIPS users mailling list; Jerry Chinn Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display Hello, Have you configured the DB support, so that the User Management can list the users ? Do you see any errors in the apache logs ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/27/2017 10:03 PM, jhchinn wrote: I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 following the directions from http://controlpanel.opensips.org/htmldoc/INSTALL.html. Everything loaded and I can get to the CP and all frames via multiple browsers. The display for the User Management doesn't display anything while all others do. [Image removed by sender.] Any ideas why and what to do to fix it? Thanks ________________________________ View this message in context: User Management frame in OpenSIPS-CP has no display Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 116142 bytes Desc: image001.png URL: From JHChinn at TheNavisWay.com Fri Jul 28 13:37:41 2017 From: JHChinn at TheNavisWay.com (Jerry Chinn) Date: Fri, 28 Jul 2017 17:37:41 +0000 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: References: <1501182229494-7608166.post@n2.nabble.com> <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> Message-ID: <760a1bac1a49498d92802cc17424267c@hil-vs-exdag02.buehner-fry.com> Bogdan, I fixed the short_open_tag in the php.ini file (I Had not removed the ;). I restarted the httpd.service and still see either the blank screen with the } left in the /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php file OR the below view when the } is removed [cid:image001.png at 01D30787.30137FC0] Any thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 10:09 AM To: Jerry Chinn; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display That is the typical error if you forgot turn ON the short_open_tag PHP option in php.ini for apache (not for cli). And be sure you restart apache after doing the change. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 07:52 PM, Jerry Chinn wrote: Bogdan, I found an error in the /var/www/opensips-cp/web/tools/users/user_management/user_management.php file On line 273 and 274 there is an unexpected } character. I removed them and I am now getting the attached display in the User Management display Your thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 3:59 AM To: OpenSIPS users mailling list; Jerry Chinn Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display Hello, Have you configured the DB support, so that the User Management can list the users ? Do you see any errors in the apache logs ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/27/2017 10:03 PM, jhchinn wrote: I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 following the directions from http://controlpanel.opensips.org/htmldoc/INSTALL.html. Everything loaded and I can get to the CP and all frames via multiple browsers. The display for the User Management doesn't display anything while all others do. [Image removed by sender.] Any ideas why and what to do to fix it? Thanks ________________________________ View this message in context: User Management frame in OpenSIPS-CP has no display Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 823 bytes Desc: image001.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 116142 bytes Desc: image002.png URL: From bogdan at opensips.org Fri Jul 28 13:40:15 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 20:40:15 +0300 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <760a1bac1a49498d92802cc17424267c@hil-vs-exdag02.buehner-fry.com> References: <1501182229494-7608166.post@n2.nabble.com> <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> <760a1bac1a49498d92802cc17424267c@hil-vs-exdag02.buehner-fry.com> Message-ID: <806c830e-c46d-8033-c03e-6b21410a7184@opensips.org> What is the file you fixed ? Have you restarted apache ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 08:37 PM, Jerry Chinn wrote: > > Bogdan, > > I fixed the short_open_tag in the php.ini file (I Had not removed the ;). > > I restarted the httpd.service and still see either the blank screen > with the } left in the > /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php > file > > OR > > the below view when the } is removed > > cid:image001.png at 01D30787.30137FC0 > > Any thoughts? > > *Jerry Chinn* > > *Telecom VoIP Specialist* > > *NAVIS *More Performance. More Profit. > > tel 541-330-3562 > > www.TheNavisWay.com > > Facebook | Twitter > | LinkedIn > | Blog > > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Friday, July 28, 2017 10:09 AM > *To:* Jerry Chinn; OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP > has no display > > That is the typical error if you forgot turn ON the short_open_tag PHP > option in php.ini for apache (not for cli). And be sure you restart > apache after doing the change. > > Regards, > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/28/2017 07:52 PM, Jerry Chinn wrote: > > Bogdan, > > I found an error in the > /var/www/opensips-cp/web/tools/users/user_management/user_management.php > file > > On line 273 and 274 there is an unexpected } character. > > I removed them and I am now getting the attached display in the > User Management display > > > > Your thoughts? > > *Jerry Chinn* > > *Telecom VoIP Specialist* > > *NAVIS *More Performance. More Profit. > > tel 541-330-3562 > > www.TheNavisWay.com > > Facebook | Twitter > | LinkedIn > | Blog > > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Friday, July 28, 2017 3:59 AM > *To:* OpenSIPS users mailling list; Jerry Chinn > *Subject:* Re: [OpenSIPS-Users] User Management frame in > OpenSIPS-CP has no display > > Hello, > > Have you configured the DB support, so that the User Management > can list the users ? Do you see any errors in the apache logs ? > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > > > OpenSIPS Bootcamp 2017, Houston, US > > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/27/2017 10:03 PM, jhchinn wrote: > > I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 > following the directions from > http://controlpanel.opensips.org/htmldoc/INSTALL.html. > Everything loaded and I can get to the CP and all frames via > multiple browsers. The display for the User Management doesn't > display anything while all others do. Image removed by > sender.Any ideas why and what to do to fix it? Thanks > > ------------------------------------------------------------------------ > > View this message in context: User Management frame in > OpenSIPS-CP has no display > > Sent from the OpenSIPS - Users mailing list archive > > at Nabble.com. > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 116142 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 823 bytes Desc: not available URL: From bogdan at opensips.org Fri Jul 28 13:46:42 2017 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 28 Jul 2017 20:46:42 +0300 Subject: [OpenSIPS-Users] problem with parallel forking In-Reply-To: <1500955699787-7608097.post@n2.nabble.com> References: <1500955699787-7608097.post@n2.nabble.com> Message-ID: Hi, According to RFC3261, if multiple 200 OK are received (even from different branches during parallel forking), a proxy *MUST* relay them back all to caller. And the caller must accept all 200 OK for INVITE, but to decide to keep only one (and send BYE to the other branches). And note that CANCEL is sent only to branches/legs in early stage (which did not return a final >=200 reply) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/25/2017 07:08 AM, Nickylin wrote: > I know the parallel forking it means an INVITE comes into Opensips. Opensips > fork multiple INVITE and sends them out to multiple places at once. Once > it receives a 200 OK from one of those places, it sends CANCELs to the > others. But I have observation , sometimes , if opensips receive more on > 200OK from callee at the same , it will not send CANCEL to devices which > send 200ok . SO , it means more than one callee's status is connection , but > only one caller. Have any solution , callee can receive BYE , if opensips > receive 200OK at the same time. > > > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/problem-with-parallel-forking-tp7608097.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From michaelsmith201708 at hotmail.com Sun Jul 30 14:01:29 2017 From: michaelsmith201708 at hotmail.com (Michael Smith) Date: Sun, 30 Jul 2017 18:01:29 +0000 Subject: [OpenSIPS-Users] Use Gstreamer RTP packets as source Message-ID: Hello, I need to stream audio in many different encoding algorithms (G711, G722, MPEG4, etc) and I thought to use Gstreamer to encode the audio and OpenSIPS to send using a SIP communication. Will this work? Can I send the RTP encoded packets over a SIP communication using OpenSIPS? Sorry for the rookie question. Any tip will be very helpful, Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.pavlov1987 at gmail.com Mon Jul 31 00:32:26 2017 From: igor.pavlov1987 at gmail.com (Igor Pavlov) Date: Mon, 31 Jul 2017 08:32:26 +0400 Subject: [OpenSIPS-Users] Separated registrar and proxy Message-ID: <6845f1a5-6b33-c8d3-e23a-3ceaff13815a@gmail.com> Hi, all. I'm trying to separate registrar and proxy functions between several hosts. Registrations successfully saves at registrar, but sip proxy could not find record when trying to lookup at "location" table on INVITE. I can't see any querys to my DB from proxy. *registrar conf:* loadmodule "usrloc.so" modparam("usrloc", "db_url", "mysql://opensips:open_sips at db1.example.com/opensips") modparam("usrloc", "db_mode", 2) if (!save("location","f")) sl_reply_error(); // *proxy conf:* loadmodule "usrloc.so" modparam("usrloc", "db_url", "mysql://opensips:open_sips at db1.example.com/opensips") modparam("usrloc", "db_mode", 2) route[location] { xlog("L_INFO","Lookup for $ru"); lookup("location"); switch($retcode) { case -1: case -3: sl_send_reply("404","Not found"); exit; case -2: sl_send_reply("405","Not found"); exit; }; } // -- ____________ Best regards, Igor Pavlov -------------- next part -------------- An HTML attachment was scrubbed... URL: From basit.engg at gmail.com Mon Jul 31 11:37:48 2017 From: basit.engg at gmail.com (Abdul Basit) Date: Mon, 31 Jul 2017 20:37:48 +0500 Subject: [OpenSIPS-Users] SIP password auth mechanism In-Reply-To: References: <1fa97094-00aa-8e54-c7ba-0073d441cae8@opensips.org> Message-ID: Hi Bogdan, Sorry for very late reply. I couldn't find any implementation if *EC-SRP *yet. However, Ejabbered implemented https://en.wikipedia.org/wiki/ Salted_Challenge_Response_Authentication_Mechanism *(SCRAM)* This is interesting model and can be adopted for SIP based services as well. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Fri, Mar 10, 2017 at 8:29 PM, Bogdan-Andrei Iancu wrote: > Hi Abdul, > > I see that's a draft, so hard to judge on how far it will get. And > something like this is not on our roadmap, maybe because of its very, very > low priority in terms of needs. Do you have any idea if anyone actually > implemented this ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Summit May 2017 Amsterdam > http://www.opensips.org/events/Summit-2017Amsterdam.html > > On 03/09/2017 12:37 PM, Abdul Basit wrote: > > Hi Geeks, > > While exploring further I found a draft explaining elliptic curve secure > remote protocol (*EC-SRP*) for SIP authentication > https://tools.ietf.org/html/draft-liu-sipcore-ec-srp5-03 > > This explanation seems align with my requirements of not storing password > in database. > UAC and UAS both should support EC-SRP. > > Do we have any road-map of opensips implementing of EC-RSP or similar > authentication mechanism? > I will check the same with PJSIP because i couldn't find any traces on > their forum as well. > > -- > regards, > > abdul basit > > > On Wed, Mar 8, 2017 at 9:53 PM, Abdul Basit wrote: > >> Hi Bogdan, >> >> I am using PJSIP as UAC and Opensips as UAS with radius for AAA. >> I wanted to avoid getting into the code but let me check the flexibility. >> >> Thank you for your reply :) >> >> -- >> regards, >> >> abdul basit >> >> On Wed, Mar 8, 2017 at 1:34 AM, Bogdan-Andrei Iancu < >> bogdan at opensips.org> wrote: >> >>> Hi Abdul, >>> >>> Besides the digest auth, there is no other standard auth mechanism for >>> SIP, AFAIK. >>> >>> If you have control over the SIP UAC, of course, you could try to build >>> your own auth mechanism - OpenSIPS offers enough flexibility in terms of >>> both header manipulation and data computing. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> OpenSIPS Summit May 2017 Amsterdam >>> http://www.opensips.org/events/Summit-2017Amsterdam.html >>> >>> On 03/07/2017 10:26 AM, Abdul Basit wrote: >>> >>> Hi, >>> I have a scenario where I will create password HASH = SALT + STRING and >>> save SALT and resulted HASH only in DB. I will transport random STRING >>> value to my custom sip application as password. >>> Digest authentication is not comply with this requirement. Is that any >>> supported authentication mechanism that can fulfill this requirement. >>> or is there any more appropriate authentication mechanism by >>> opensips/kamailio? >>> One of the objectives is in case DB will compromise, users passwords >>> will not available because random STRING will not store in DB. >>> Looking forward for suggestions and comments. >>> -- regards, >>> abdul basit >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From jhchinn at thenavisway.com Mon Jul 31 13:37:59 2017 From: jhchinn at thenavisway.com (jhchinn) Date: Mon, 31 Jul 2017 10:37:59 -0700 (MST) Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <1501182229494-7608166.post@n2.nabble.com> References: <1501182229494-7608166.post@n2.nabble.com> Message-ID: <1501522679591-7608194.post@n2.nabble.com> Bogdan, Is there any other advice on how to proceed? Thanks, Jerry -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/User-Management-frame-in-OpenSIPS-CP-has-no-display-tp7608166p7608194.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. From jennifer.hashimoto at caztel.com Mon Jul 31 13:52:44 2017 From: jennifer.hashimoto at caztel.com (Jennifer Hashimoto) Date: Mon, 31 Jul 2017 13:52:44 -0400 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <1501522679591-7608194.post@n2.nabble.com> References: <1501182229494-7608166.post@n2.nabble.com> <1501522679591-7608194.post@n2.nabble.com> Message-ID: <7DDAA334-6A11-4FB6-9310-7C415A819A4D@caztel.com> Jerry, I use opensips-cp maybe I can help. A couple things - I have not found there is an extraneous } that needed to be removed, I would suggest putting it back. What database are you using? I am using postgres instead of mysql and I found that there was a query where it did a count of users to see if it had anything to display and it was mistakenly finding that it had no users to display. I suggest finding the part where it does a count on users and modifying the php file to echo the sql it uses (just add a few lines like echo $sql;) then test the sql against your database. I had to change something like < $data_no=$resultset[0]['count']; --- > $data_no=$resultset[0]['count(*)’]; Jennifer --------------------------------------------------- Jennifer Akemi Hashimoto Caztel Communications jennifer.hashimoto at caztel.com 905-836-5445 > On Jul 31, 2017, at 1:37 PM, jhchinn wrote: > > Bogdan, > > Is there any other advice on how to proceed? > > Thanks, > > Jerry > > > > -- > View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/User-Management-frame-in-OpenSIPS-CP-has-no-display-tp7608166p7608194.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From JHChinn at TheNavisWay.com Mon Jul 31 17:24:54 2017 From: JHChinn at TheNavisWay.com (Jerry Chinn) Date: Mon, 31 Jul 2017 21:24:54 +0000 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <7DDAA334-6A11-4FB6-9310-7C415A819A4D@caztel.com> References: <1501182229494-7608166.post@n2.nabble.com> <1501522679591-7608194.post@n2.nabble.com> <7DDAA334-6A11-4FB6-9310-7C415A819A4D@caztel.com> Message-ID: <14e950c87ac044b3b3ff12810a088e9c@hil-vs-exdag02.buehner-fry.com> Jennifer, Thanks for your response. I did identify the issue and am posting what I found to resolve this. In the php.ini file there are (2) locations where the short_open_tag is defined. ;short_open_tag ; Default Value: On ; Development Value: Off ; Production Value: On Again in a few lines below ; This directive determines whether or not PHP will recognize code between ; tags as PHP source which should be processed as such. It's been ; recommended for several years that you not use the short tag "short cut" and ; instead to use the full tag combination. With the wide spread use ; of XML and use of these tags by other languages, the server can become easily ; confused and end up parsing the wrong code in the wrong context. But because ; this short cut has been a feature for such a long time, it's currently still ; supported for backwards compatibility, but we recommend you don't use them. ; Default Value: On ; Development Value: Off ; Production Value: Off ; http://php.net/short-open-tag short_open_tag = OFF When I searched for the parameter, I uncommented the first, and did not see the second. Once I corrected this error, all works fine. Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Jennifer Hashimoto Sent: Monday, July 31, 2017 10:53 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display Jerry, I use opensips-cp maybe I can help. A couple things - I have not found there is an extraneous } that needed to be removed, I would suggest putting it back. What database are you using? I am using postgres instead of mysql and I found that there was a query where it did a count of users to see if it had anything to display and it was mistakenly finding that it had no users to display. I suggest finding the part where it does a count on users and modifying the php file to echo the sql it uses (just add a few lines like echo $sql;) then test the sql against your database. I had to change something like < $data_no=$resultset[0]['count']; --- > $data_no=$resultset[0]['count(*)’]; Jennifer --------------------------------------------------- Jennifer Akemi Hashimoto Caztel Communications jennifer.hashimoto at caztel.com 905-836-5445 On Jul 31, 2017, at 1:37 PM, jhchinn > wrote: Bogdan, Is there any other advice on how to proceed? Thanks, Jerry -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/User-Management-frame-in-OpenSIPS-CP-has-no-display-tp7608166p7608194.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From JHChinn at TheNavisWay.com Fri Jul 28 13:50:48 2017 From: JHChinn at TheNavisWay.com (Jerry Chinn) Date: Fri, 28 Jul 2017 17:50:48 +0000 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <806c830e-c46d-8033-c03e-6b21410a7184@opensips.org> References: <1501182229494-7608166.post@n2.nabble.com> <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> <760a1bac1a49498d92802cc17424267c@hil-vs-exdag02.buehner-fry.com> <806c830e-c46d-8033-c03e-6b21410a7184@opensips.org> Message-ID: <2584ee1156d34a1396a0faa6df4b36a4@hil-vs-exdag02.buehner-fry.com> /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php And I did restart apache. Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 10:40 AM To: Jerry Chinn; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display What is the file you fixed ? Have you restarted apache ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 08:37 PM, Jerry Chinn wrote: Bogdan, I fixed the short_open_tag in the php.ini file (I Had not removed the ;). I restarted the httpd.service and still see either the blank screen with the } left in the /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php file OR the below view when the } is removed [cid:image001.png at 01D30787.30137FC0] Any thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 10:09 AM To: Jerry Chinn; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display That is the typical error if you forgot turn ON the short_open_tag PHP option in php.ini for apache (not for cli). And be sure you restart apache after doing the change. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 07:52 PM, Jerry Chinn wrote: Bogdan, I found an error in the /var/www/opensips-cp/web/tools/users/user_management/user_management.php file On line 273 and 274 there is an unexpected } character. I removed them and I am now getting the attached display in the User Management display Your thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 3:59 AM To: OpenSIPS users mailling list; Jerry Chinn Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display Hello, Have you configured the DB support, so that the User Management can list the users ? Do you see any errors in the apache logs ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/27/2017 10:03 PM, jhchinn wrote: I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 following the directions from http://controlpanel.opensips.org/htmldoc/INSTALL.html. Everything loaded and I can get to the CP and all frames via multiple browsers. The display for the User Management doesn't display anything while all others do. [Image removed by sender.] Any ideas why and what to do to fix it? Thanks ________________________________ View this message in context: User Management frame in OpenSIPS-CP has no display Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 116142 bytes Desc: image001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 823 bytes Desc: image002.jpg URL: From JHChinn at TheNavisWay.com Fri Jul 28 13:54:49 2017 From: JHChinn at TheNavisWay.com (Jerry Chinn) Date: Fri, 28 Jul 2017 17:54:49 +0000 Subject: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display In-Reply-To: <806c830e-c46d-8033-c03e-6b21410a7184@opensips.org> References: <1501182229494-7608166.post@n2.nabble.com> <221f3c8f-971b-2b66-ac59-f68a6db476d1@opensips.org> <760a1bac1a49498d92802cc17424267c@hil-vs-exdag02.buehner-fry.com> <806c830e-c46d-8033-c03e-6b21410a7184@opensips.org> Message-ID: This is what shows up in the error log after a restart: [Fri Jul 28 10:30:02.600483 2017] [mpm_prefork:notice] [pid 3227] AH00170: caught SIGWINCH, shutting down gracefully [Fri Jul 28 10:30:07.456104 2017] [core:notice] [pid 3246] SELinux policy enabled; httpd running as context system_u:system_r:httpd_t:s0 [Fri Jul 28 10:30:07.457197 2017] [suexec:notice] [pid 3246] AH01232: suEXEC mechanism enabled (wrapper: /usr/sbin/suexec) [Fri Jul 28 10:30:10.396994 2017] [auth_digest:notice] [pid 3246] AH01757: generating secret for digest authentication ... [Fri Jul 28 10:30:10.399234 2017] [lbmethod_heartbeat:notice] [pid 3246] AH02282: No slotmem from mod_heartmonitor [Fri Jul 28 10:30:10.448825 2017] [mpm_prefork:notice] [pid 3246] AH00163: Apache/2.4.6 (CentOS) PHP/5.4.16 configured -- resuming normal operations [Fri Jul 28 10:30:10.448882 2017] [core:notice] [pid 3246] AH00094: Command line: '/usr/sbin/httpd -D FOREGROUND' [Fri Jul 28 10:30:21.538198 2017] [:error] [pid 3251] [client 10.2.112.22:53033] PHP Warning: Creating default object from empty value in /var/www/opensips-cp/config/db.inc.php on line 24, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:30:21.556313 2017] [:error] [pid 3251] [client 10.2.112.22:53033] PHP Parse error: syntax error, unexpected '}' in /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php on line 273, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:30:24.139244 2017] [:error] [pid 3247] [client 10.2.112.22:53037] PHP Notice: Undefined variable: i in /var/www/opensips-cp/web/main.php on line 42, referer: http://10.2.112.29/cp/ [Fri Jul 28 10:30:24.193831 2017] [:error] [pid 3249] [client 10.2.112.22:53043] PHP Warning: Creating default object from empty value in /var/www/opensips-cp/config/db.inc.php on line 24, referer: http://10.2.112.29/cp/main.php [Fri Jul 28 10:30:24.231075 2017] [:error] [pid 3249] [client 10.2.112.22:53043] PHP Parse error: syntax error, unexpected '}' in /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php on line 273, referer: http://10.2.112.29/cp/main.php [Fri Jul 28 10:35:25.738895 2017] [:error] [pid 3251] [client 10.2.112.22:54191] PHP Notice: Undefined variable: i in /var/www/opensips-cp/web/main.php on line 42, referer: http://10.2.112.29/cp/ [Fri Jul 28 10:35:25.817481 2017] [:error] [pid 3251] [client 10.2.112.22:54191] PHP Warning: Creating default object from empty value in /var/www/opensips-cp/config/db.inc.php on line 24, referer: http://10.2.112.29/cp/main.php [Fri Jul 28 10:35:25.844710 2017] [:error] [pid 3251] [client 10.2.112.22:54191] PHP Parse error: syntax error, unexpected '}' in /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php on line 273, referer: http://10.2.112.29/cp/main.php [Fri Jul 28 10:35:30.676243 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Warning: Creating default object from empty value in /var/www/opensips-cp/config/db.inc.php on line 24, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.700095 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined index: username in /var/www/opensips-cp/web/tools/users/alias_management/template/alias_management.main.php on line 24, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.700198 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined index: alias_username in /var/www/opensips-cp/web/tools/users/alias_management/template/alias_management.main.php on line 25, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.700221 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined index: alias_domain in /var/www/opensips-cp/web/tools/users/alias_management/template/alias_management.main.php on line 26, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.700236 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined index: alias_type in /var/www/opensips-cp/web/tools/users/alias_management/template/alias_management.main.php on line 27, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.703075 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined variable: temp in /var/www/opensips-cp/web/tools/users/alias_management/lib/functions.inc.php on line 60, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.703179 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined variable: temp in /var/www/opensips-cp/web/tools/users/alias_management/lib/functions.inc.php on line 60, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.703410 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined variable: temp in /var/www/opensips-cp/web/tools/users/alias_management/lib/functions.inc.php on line 91, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.703439 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined variable: temp in /var/www/opensips-cp/web/tools/users/alias_management/lib/functions.inc.php on line 91, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.704470 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined variable: errors in /var/www/opensips-cp/web/tools/users/alias_management/alias_management.php on line 304, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:30.704514 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Notice: Undefined variable: errors in /var/www/opensips-cp/web/tools/users/alias_management/alias_management.php on line 304, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:31.767117 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Warning: Creating default object from empty value in /var/www/opensips-cp/config/db.inc.php on line 24, referer: http://10.2.112.29/cp/menu.php [Fri Jul 28 10:35:31.786457 2017] [:error] [pid 3254] [client 10.2.112.22:54227] PHP Parse error: syntax error, unexpected '}' in /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php on line 273, referer: http://10.2.112.29/cp/menu.php Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 10:40 AM To: Jerry Chinn; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display What is the file you fixed ? Have you restarted apache ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 08:37 PM, Jerry Chinn wrote: Bogdan, I fixed the short_open_tag in the php.ini file (I Had not removed the ;). I restarted the httpd.service and still see either the blank screen with the } left in the /var/www/opensips-cp/web/tools/users/user_management/template/user_management.main.php file OR the below view when the } is removed [cid:image001.png at 01D30787.30137FC0] Any thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 10:09 AM To: Jerry Chinn; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display That is the typical error if you forgot turn ON the short_open_tag PHP option in php.ini for apache (not for cli). And be sure you restart apache after doing the change. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/28/2017 07:52 PM, Jerry Chinn wrote: Bogdan, I found an error in the /var/www/opensips-cp/web/tools/users/user_management/user_management.php file On line 273 and 274 there is an unexpected } character. I removed them and I am now getting the attached display in the User Management display Your thoughts? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com Facebook | Twitter | LinkedIn | Blog From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, July 28, 2017 3:59 AM To: OpenSIPS users mailling list; Jerry Chinn Subject: Re: [OpenSIPS-Users] User Management frame in OpenSIPS-CP has no display Hello, Have you configured the DB support, so that the User Management can list the users ? Do you see any errors in the apache logs ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 07/27/2017 10:03 PM, jhchinn wrote: I just loaded OpenSIPS-CP 7.2.3 on a server running CentOS7 following the directions from http://controlpanel.opensips.org/htmldoc/INSTALL.html. Everything loaded and I can get to the CP and all frames via multiple browsers. The display for the User Management doesn't display anything while all others do. [Image removed by sender.] Any ideas why and what to do to fix it? Thanks ________________________________ View this message in context: User Management frame in OpenSIPS-CP has no display Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 116142 bytes Desc: image001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 823 bytes Desc: image002.jpg URL: From goley_ev_sp at mail.ru Mon Jul 31 10:48:52 2017 From: goley_ev_sp at mail.ru (Evgeniy G.) Date: Mon, 31 Jul 2017 07:48:52 -0700 (MST) Subject: [OpenSIPS-Users] Does not release shared memory In-Reply-To: <7b0172f6-8517-de52-0286-9bf4cd5afbb3@opensips.org> References: <1500661901795-7608065.post@n2.nabble.com> <186d936f-7690-0be0-494c-cd5cec92296c@opensips.org> <1500884933063-7608080.post@n2.nabble.com> <1500896228657-7608082.post@n2.nabble.com> <1500896281600-7608083.post@n2.nabble.com> <1faa0fac-e42f-2722-f798-8ca04554c1e2@opensips.org> <1500907583776-7608091.post@n2.nabble.com> <1501072866468-7608134.post@n2.nabble.com> <1501074132730-7608135.post@n2.nabble.com> <7b0172f6-8517-de52-0286-9bf4cd5afbb3@opensips.org> Message-ID: <1501512514.443817723@f313.i.mail.ru> Hi, Liviu Chircu! Many thanks for your desire to help, well done that the project is alive and you support it. Regarding my problem, everything turned out to be easier than I expected. I already figured out, including thanks to your hint. Thank you for your assistance! Best regards, E.Goley >Среда, 26 июля 2017, 16:43 +03:00 от "Liviu Chircu [via OpenSIPS (Open SIP Server)]" : > >Here are some more MI commands that might be useful: > >opensipsctl fifo ps -> the type of each process > >opensipsctl fifo get_statistics dialog: tm: usrloc: -> display in-memory ># of transactions / dialogs / registered users > >Like before, if you suspect a leak, please provide the output of the >above MI commands, along with a "kill -SIGUSR1" memory map as instructed >earlier, and we should have a solid indication as to whether there's a >real problem at hand or not. > >Best regards, > >Liviu Chircu >OpenSIPS Developer >http://www.opensips-solutions.com > >On 26.07.2017 16:02, Evgeniy G. via Users wrote: > >> Now I'm testing the server under load, 600 calls per minute and watching the >> situation. I have only two sockets in the configuration: >> listen = udp: 10.2.1.61: 7062 >> listen = udp: ХХ.ХХ.ХХХ.XXX: 7062 >> In the children parameter the value 2 is specified. But I can not understand >> why 18 processes are created at startup ?... It's interesting that only two >> of them take a role in the processing of calls, why? ... I found this out, >> because of the decrease in two processes (pkmem:4 and pkmem:5) of pkg memory >> . >> Please explain the work with pkg memory when it is released. >> I watched the server for 3 hours under load and 1 hour without, as a result, >> on two processes the free private memory (pkmem) was rectilinearly reduced. >> After testing, without restarting the server, private memory (pkgmem) is not >> released, is it the way it should be? >> >> [root at sbc sbin]# date >> Срд Июл 26 09:31:45 MSK 2017 >> [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: >> [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: >> pkmem:0-total_size:: 67108864 >> pkmem:0-used_size:: 66608 >> pkmem:0-real_used_size:: 112560 >> pkmem:0-max_used_size:: 112560 >> pkmem:0-free_size:: 66996304 >> pkmem:0-fragments:: 510 >> pkmem:1-total_size:: 67108864 >> pkmem:1-used_size:: 87128 >> pkmem:1-real_used_size:: 135912 >> pkmem:1-max_used_size:: 146248 >> pkmem:1-free_size:: 66972952 >> pkmem:1-fragments:: 526 >> pkmem:2-total_size:: 67108864 >> pkmem:2-used_size:: 64832 >> pkmem:2-real_used_size:: 110664 >> pkmem:2-max_used_size:: 110672 >> pkmem:2-free_size:: 66998200 >> pkmem:2-fragments:: 500 >> pkmem:3-total_size:: 67108864 >> pkmem:3-used_size:: 64832 >> pkmem:3-real_used_size:: 110664 >> pkmem:3-max_used_size:: 110672 >> pkmem:3-free_size:: 66998200 >> pkmem:3-fragments:: 500 >> pkmem:4-total_size:: 67108864 >> pkmem:4-used_size:: 1144576 >> pkmem:4-real_used_size:: 2058560 >> pkmem:4-max_used_size:: 2069256 >> pkmem:4-free_size:: 65050304 >> pkmem:4-fragments:: 36614 >> pkmem:5-total_size:: 67108864 >> pkmem:5-used_size:: 1381584 >> pkmem:5-real_used_size:: 2495896 >> pkmem:5-max_used_size:: 2506568 >> pkmem:5-free_size:: 64612968 >> pkmem:5-fragments:: 44959 >> pkmem:6-total_size:: 67108864 >> pkmem:6-used_size:: 119960 >> pkmem:6-real_used_size:: 166824 >> pkmem:6-max_used_size:: 173640 >> pkmem:6-free_size:: 66942040 >> pkmem:6-fragments:: 518 >> pkmem:7-total_size:: 67108864 >> pkmem:7-used_size:: 119960 >> pkmem:7-real_used_size:: 166824 >> pkmem:7-max_used_size:: 173808 >> pkmem:7-free_size:: 66942040 >> pkmem:7-fragments:: 518 >> pkmem:8-total_size:: 67108864 >> pkmem:8-used_size:: 119840 >> pkmem:8-real_used_size:: 165912 >> pkmem:8-max_used_size:: 165912 >> pkmem:8-free_size:: 66942952 >> pkmem:8-fragments:: 515 >> pkmem:9-total_size:: 67108864 >> pkmem:9-used_size:: 119840 >> pkmem:9-real_used_size:: 165912 >> pkmem:9-max_used_size:: 165912 >> pkmem:9-free_size:: 66942952 >> pkmem:9-fragments:: 515 >> pkmem:10-total_size:: 67108864 >> pkmem:10-used_size:: 119840 >> pkmem:10-real_used_size:: 165912 >> pkmem:10-max_used_size:: 165912 >> pkmem:10-free_size:: 66942952 >> pkmem:10-fragments:: 515 >> pkmem:11-total_size:: 67108864 >> pkmem:11-used_size:: 119840 >> pkmem:11-real_used_size:: 165912 >> pkmem:11-max_used_size:: 165912 >> pkmem:11-free_size:: 66942952 >> pkmem:11-fragments:: 515 >> pkmem:12-total_size:: 67108864 >> pkmem:12-used_size:: 119840 >> pkmem:12-real_used_size:: 165912 >> pkmem:12-max_used_size:: 165912 >> pkmem:12-free_size:: 66942952 >> pkmem:12-fragments:: 515 >> pkmem:13-total_size:: 67108864 >> pkmem:13-used_size:: 119840 >> pkmem:13-real_used_size:: 165912 >> pkmem:13-max_used_size:: 165912 >> pkmem:13-free_size:: 66942952 >> pkmem:13-fragments:: 515 >> pkmem:14-total_size:: 67108864 >> pkmem:14-used_size:: 119840 >> pkmem:14-real_used_size:: 165912 >> pkmem:14-max_used_size:: 165912 >> pkmem:14-free_size:: 66942952 >> pkmem:14-fragments:: 515 >> pkmem:15-total_size:: 67108864 >> pkmem:15-used_size:: 119840 >> pkmem:15-real_used_size:: 165912 >> pkmem:15-max_used_size:: 165912 >> pkmem:15-free_size:: 66942952 >> pkmem:15-fragments:: 515 >> pkmem:16-total_size:: 67108864 >> pkmem:16-used_size:: 119840 >> pkmem:16-real_used_size:: 165912 >> pkmem:16-max_used_size:: 165912 >> pkmem:16-free_size:: 66942952 >> pkmem:16-fragments:: 515 >> pkmem:17-total_size:: 67108864 >> pkmem:17-used_size:: 109904 >> pkmem:17-real_used_size:: 155832 >> pkmem:17-max_used_size:: 155832 >> pkmem:17-free_size:: 66953032 >> pkmem:17-fragments:: 504 >> [root at sbc sbin]# >> >> .... >> >> >> [root at sbc sbin]# date >> Срд Июл 26 10:35:47 MSK 2017 >> >> [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: >> pkmem:0-total_size:: 67108864 >> pkmem:0-used_size:: 66608 >> pkmem:0-real_used_size:: 112560 >> pkmem:0-max_used_size:: 112560 >> pkmem:0-free_size:: 66996304 >> pkmem:0-fragments:: 510 >> pkmem:1-total_size:: 67108864 >> pkmem:1-used_size:: 87128 >> pkmem:1-real_used_size:: 135912 >> pkmem:1-max_used_size:: 146248 >> pkmem:1-free_size:: 66972952 >> pkmem:1-fragments:: 526 >> pkmem:2-total_size:: 67108864 >> pkmem:2-used_size:: 64832 >> pkmem:2-real_used_size:: 110664 >> pkmem:2-max_used_size:: 110672 >> pkmem:2-free_size:: 66998200 >> pkmem:2-fragments:: 500 >> pkmem:3-total_size:: 67108864 >> pkmem:3-used_size:: 64832 >> pkmem:3-real_used_size:: 110664 >> pkmem:3-max_used_size:: 110672 >> pkmem:3-free_size:: 66998200 >> pkmem:3-fragments:: 500 >> pkmem:4-total_size:: 67108864 >> pkmem:4-used_size:: 2786296 >> pkmem:4-real_used_size:: 5088608 >> pkmem:4-max_used_size:: 5099248 >> pkmem:4-free_size:: 62020256 >> pkmem:4-fragments:: 94460 >> pkmem:5-total_size:: 67108864 >> pkmem:5-used_size:: 3334952 >> pkmem:5-real_used_size:: 6101784 >> pkmem:5-max_used_size:: 6112384 >> pkmem:5-free_size:: 61007080 >> pkmem:5-fragments:: 113815 >> pkmem:6-total_size:: 67108864 >> pkmem:6-used_size:: 120008 >> pkmem:6-real_used_size:: 167496 >> pkmem:6-max_used_size:: 175304 >> pkmem:6-free_size:: 66941368 >> pkmem:6-fragments:: 519 >> pkmem:7-total_size:: 67108864 >> pkmem:7-used_size:: 120000 >> pkmem:7-real_used_size:: 167584 >> pkmem:7-max_used_size:: 175240 >> pkmem:7-free_size:: 66941280 >> pkmem:7-fragments:: 519 >> pkmem:8-total_size:: 67108864 >> pkmem:8-used_size:: 119880 >> pkmem:8-real_used_size:: 165976 >> pkmem:8-max_used_size:: 165976 >> pkmem:8-free_size:: 66942888 >> pkmem:8-fragments:: 516 >> pkmem:9-total_size:: 67108864 >> pkmem:9-used_size:: 119952 >> pkmem:9-real_used_size:: 166288 >> pkmem:9-max_used_size:: 167024 >> pkmem:9-free_size:: 66942576 >> pkmem:9-fragments:: 518 >> pkmem:10-total_size:: 67108864 >> pkmem:10-used_size:: 119880 >> pkmem:10-real_used_size:: 165976 >> pkmem:10-max_used_size:: 165976 >> pkmem:10-free_size:: 66942888 >> pkmem:10-fragments:: 516 >> pkmem:11-total_size:: 67108864 >> pkmem:11-used_size:: 119880 >> pkmem:11-real_used_size:: 165976 >> pkmem:11-max_used_size:: 165976 >> pkmem:11-free_size:: 66942888 >> pkmem:11-fragments:: 516 >> pkmem:12-total_size:: 67108864 >> pkmem:12-used_size:: 119880 >> pkmem:12-real_used_size:: 165976 >> pkmem:12-max_used_size:: 165976 >> pkmem:12-free_size:: 66942888 >> pkmem:12-fragments:: 516 >> pkmem:13-total_size:: 67108864 >> pkmem:13-used_size:: 119952 >> pkmem:13-real_used_size:: 166288 >> pkmem:13-max_used_size:: 167024 >> pkmem:13-free_size:: 66942576 >> pkmem:13-fragments:: 518 >> pkmem:14-total_size:: 67108864 >> pkmem:14-used_size:: 119952 >> pkmem:14-real_used_size:: 166288 >> pkmem:14-max_used_size:: 167024 >> pkmem:14-free_size:: 66942576 >> pkmem:14-fragments:: 518 >> pkmem:15-total_size:: 67108864 >> pkmem:15-used_size:: 119880 >> pkmem:15-real_used_size:: 165976 >> pkmem:15-max_used_size:: 165976 >> pkmem:15-free_size:: 66942888 >> pkmem:15-fragments:: 516 >> pkmem:16-total_size:: 67108864 >> pkmem:16-used_size:: 119880 >> pkmem:16-real_used_size:: 165976 >> pkmem:16-max_used_size:: 165976 >> pkmem:16-free_size:: 66942888 >> pkmem:16-fragments:: 516 >> pkmem:17-total_size:: 67108864 >> pkmem:17-used_size:: 109904 >> pkmem:17-real_used_size:: 155832 >> pkmem:17-max_used_size:: 155832 >> pkmem:17-free_size:: 66953032 >> pkmem:17-fragments:: 504 >> >> .... >> >> [root at sbc sbin]# date >> Срд Июл 26 11:08:37 MSK 2017 >> [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: >> pkmem:0-total_size:: 67108864 >> pkmem:0-used_size:: 66608 >> pkmem:0-real_used_size:: 112560 >> pkmem:0-max_used_size:: 112560 >> pkmem:0-free_size:: 66996304 >> pkmem:0-fragments:: 510 >> pkmem:1-total_size:: 67108864 >> pkmem:1-used_size:: 87128 >> pkmem:1-real_used_size:: 135912 >> pkmem:1-max_used_size:: 146248 >> pkmem:1-free_size:: 66972952 >> pkmem:1-fragments:: 526 >> pkmem:2-total_size:: 67108864 >> pkmem:2-used_size:: 64832 >> pkmem:2-real_used_size:: 110664 >> pkmem:2-max_used_size:: 110672 >> pkmem:2-free_size:: 66998200 >> pkmem:2-fragments:: 500 >> pkmem:3-total_size:: 67108864 >> pkmem:3-used_size:: 64832 >> pkmem:3-real_used_size:: 110664 >> pkmem:3-max_used_size:: 110672 >> pkmem:3-free_size:: 66998200 >> pkmem:3-fragments:: 500 >> pkmem:4-total_size:: 67108864 >> pkmem:4-used_size:: 4144936 >> pkmem:4-real_used_size:: 7595432 >> pkmem:4-max_used_size:: 7606080 >> pkmem:4-free_size:: 59513432 >> pkmem:4-fragments:: 142300 >> pkmem:5-total_size:: 67108864 >> pkmem:5-used_size:: 4946520 >> pkmem:5-real_used_size:: 9075448 >> pkmem:5-max_used_size:: 9086080 >> pkmem:5-free_size:: 58033416 >> pkmem:5-fragments:: 170570 >> pkmem:6-total_size:: 67108864 >> pkmem:6-used_size:: 120008 >> pkmem:6-real_used_size:: 167496 >> pkmem:6-max_used_size:: 175304 >> pkmem:6-free_size:: 66941368 >> pkmem:6-fragments:: 519 >> pkmem:7-total_size:: 67108864 >> pkmem:7-used_size:: 120000 >> pkmem:7-real_used_size:: 167584 >> pkmem:7-max_used_size:: 175272 >> pkmem:7-free_size:: 66941280 >> pkmem:7-fragments:: 520 >> pkmem:8-total_size:: 67108864 >> pkmem:8-used_size:: 119880 >> pkmem:8-real_used_size:: 165976 >> pkmem:8-max_used_size:: 165976 >> pkmem:8-free_size:: 66942888 >> pkmem:8-fragments:: 516 >> pkmem:9-total_size:: 67108864 >> pkmem:9-used_size:: 119952 >> pkmem:9-real_used_size:: 166288 >> pkmem:9-max_used_size:: 167024 >> pkmem:9-free_size:: 66942576 >> pkmem:9-fragments:: 518 >> pkmem:10-total_size:: 67108864 >> pkmem:10-used_size:: 119880 >> pkmem:10-real_used_size:: 165976 >> pkmem:10-max_used_size:: 165976 >> pkmem:10-free_size:: 66942888 >> pkmem:10-fragments:: 516 >> pkmem:11-total_size:: 67108864 >> pkmem:11-used_size:: 119880 >> pkmem:11-real_used_size:: 165976 >> pkmem:11-max_used_size:: 165976 >> pkmem:11-free_size:: 66942888 >> pkmem:11-fragments:: 516 >> pkmem:12-total_size:: 67108864 >> pkmem:12-used_size:: 119880 >> pkmem:12-real_used_size:: 165976 >> pkmem:12-max_used_size:: 165976 >> pkmem:12-free_size:: 66942888 >> pkmem:12-fragments:: 516 >> pkmem:13-total_size:: 67108864 >> pkmem:13-used_size:: 119952 >> pkmem:13-real_used_size:: 166288 >> pkmem:13-max_used_size:: 167024 >> pkmem:13-free_size:: 66942576 >> pkmem:13-fragments:: 518 >> pkmem:14-total_size:: 67108864 >> pkmem:14-used_size:: 119952 >> pkmem:14-real_used_size:: 166288 >> pkmem:14-max_used_size:: 167024 >> pkmem:14-free_size:: 66942576 >> pkmem:14-fragments:: 518 >> pkmem:15-total_size:: 67108864 >> pkmem:15-used_size:: 119880 >> pkmem:15-real_used_size:: 165976 >> pkmem:15-max_used_size:: 165976 >> pkmem:15-free_size:: 66942888 >> pkmem:15-fragments:: 516 >> pkmem:16-total_size:: 67108864 >> pkmem:16-used_size:: 119880 >> pkmem:16-real_used_size:: 165976 >> pkmem:16-max_used_size:: 165976 >> pkmem:16-free_size:: 66942888 >> pkmem:16-fragments:: 516 >> pkmem:17-total_size:: 67108864 >> pkmem:17-used_size:: 109904 >> pkmem:17-real_used_size:: 155832 >> pkmem:17-max_used_size:: 155832 >> pkmem:17-free_size:: 66953032 >> pkmem:17-fragments:: 504 >> [root at sbc sbin]# >> >> ... >> After testing, without restarting the service. >> >> [root at sbc sbin]# date >> Срд Июл 26 12:09:01 MSK 2017 >> [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: >> pkmem:0-total_size:: 67108864 >> pkmem:0-used_size:: 66608 >> pkmem:0-real_used_size:: 112560 >> pkmem:0-max_used_size:: 112560 >> pkmem:0-free_size:: 66996304 >> pkmem:0-fragments:: 510 >> pkmem:1-total_size:: 67108864 >> pkmem:1-used_size:: 87128 >> pkmem:1-real_used_size:: 135912 >> pkmem:1-max_used_size:: 146248 >> pkmem:1-free_size:: 66972952 >> pkmem:1-fragments:: 526 >> pkmem:2-total_size:: 67108864 >> pkmem:2-used_size:: 64832 >> pkmem:2-real_used_size:: 110664 >> pkmem:2-max_used_size:: 110672 >> pkmem:2-free_size:: 66998200 >> pkmem:2-fragments:: 500 >> pkmem:3-total_size:: 67108864 >> pkmem:3-used_size:: 64832 >> pkmem:3-real_used_size:: 110664 >> pkmem:3-max_used_size:: 110672 >> pkmem:3-free_size:: 66998200 >> pkmem:3-fragments:: 500 >> pkmem:4-total_size:: 67108864 >> pkmem:4-used_size:: 6444664 >> pkmem:4-real_used_size:: 11838512 >> pkmem:4-max_used_size:: 11849192 >> pkmem:4-free_size:: 55270352 >> pkmem:4-fragments:: 223275 >> pkmem:5-total_size:: 67108864 >> pkmem:5-used_size:: 7769568 >> pkmem:5-real_used_size:: 14284264 >> pkmem:5-max_used_size:: 14294848 >> pkmem:5-free_size:: 52824600 >> pkmem:5-fragments:: 269975 >> pkmem:6-total_size:: 67108864 >> pkmem:6-used_size:: 120000 >> pkmem:6-real_used_size:: 167512 >> pkmem:6-max_used_size:: 175336 >> pkmem:6-free_size:: 66941352 >> pkmem:6-fragments:: 520 >> pkmem:7-total_size:: 67108864 >> pkmem:7-used_size:: 120000 >> pkmem:7-real_used_size:: 167584 >> pkmem:7-max_used_size:: 175272 >> pkmem:7-free_size:: 66941280 >> pkmem:7-fragments:: 520 >> pkmem:8-total_size:: 67108864 >> pkmem:8-used_size:: 119880 >> pkmem:8-real_used_size:: 165976 >> pkmem:8-max_used_size:: 165976 >> pkmem:8-free_size:: 66942888 >> pkmem:8-fragments:: 516 >> pkmem:9-total_size:: 67108864 >> pkmem:9-used_size:: 119952 >> pkmem:9-real_used_size:: 166288 >> pkmem:9-max_used_size:: 167024 >> pkmem:9-free_size:: 66942576 >> pkmem:9-fragments:: 518 >> pkmem:10-total_size:: 67108864 >> pkmem:10-used_size:: 119880 >> pkmem:10-real_used_size:: 165976 >> pkmem:10-max_used_size:: 165976 >> pkmem:10-free_size:: 66942888 >> pkmem:10-fragments:: 516 >> pkmem:11-total_size:: 67108864 >> pkmem:11-used_size:: 119880 >> pkmem:11-real_used_size:: 165976 >> pkmem:11-max_used_size:: 165976 >> pkmem:11-free_size:: 66942888 >> pkmem:11-fragments:: 516 >> pkmem:12-total_size:: 67108864 >> pkmem:12-used_size:: 119880 >> pkmem:12-real_used_size:: 165976 >> pkmem:12-max_used_size:: 165976 >> pkmem:12-free_size:: 66942888 >> pkmem:12-fragments:: 516 >> pkmem:13-total_size:: 67108864 >> pkmem:13-used_size:: 119952 >> pkmem:13-real_used_size:: 166288 >> pkmem:13-max_used_size:: 167024 >> pkmem:13-free_size:: 66942576 >> pkmem:13-fragments:: 518 >> pkmem:14-total_size:: 67108864 >> pkmem:14-used_size:: 119952 >> pkmem:14-real_used_size:: 166288 >> pkmem:14-max_used_size:: 167024 >> pkmem:14-free_size:: 66942576 >> pkmem:14-fragments:: 518 >> pkmem:15-total_size:: 67108864 >> pkmem:15-used_size:: 119880 >> pkmem:15-real_used_size:: 165976 >> pkmem:15-max_used_size:: 165976 >> pkmem:15-free_size:: 66942888 >> pkmem:15-fragments:: 516 >> pkmem:16-total_size:: 67108864 >> pkmem:16-used_size:: 119880 >> pkmem:16-real_used_size:: 165976 >> pkmem:16-max_used_size:: 165976 >> pkmem:16-free_size:: 66942888 >> pkmem:16-fragments:: 516 >> pkmem:17-total_size:: 67108864 >> pkmem:17-used_size:: 109904 >> pkmem:17-real_used_size:: 155832 >> pkmem:17-max_used_size:: 155832 >> pkmem:17-free_size:: 66953032 >> pkmem:17-fragments:: 504 >> [root at sbc sbin]# >> >> ... >> >> Hour without load. >> >> [root at sbc sbin]# date >> Срд Июл 26 12:54:14 MSK 2017 >> [root at sbc sbin]# >> [root at sbc sbin]# ./opensipsctl fifo get_statistics pkmem: >> pkmem:0-total_size:: 67108864 >> pkmem:0-used_size:: 66608 >> pkmem:0-real_used_size:: 112560 >> pkmem:0-max_used_size:: 112560 >> pkmem:0-free_size:: 66996304 >> pkmem:0-fragments:: 510 >> pkmem:1-total_size:: 67108864 >> pkmem:1-used_size:: 87128 >> pkmem:1-real_used_size:: 135912 >> pkmem:1-max_used_size:: 146248 >> pkmem:1-free_size:: 66972952 >> pkmem:1-fragments:: 526 >> pkmem:2-total_size:: 67108864 >> pkmem:2-used_size:: 64832 >> pkmem:2-real_used_size:: 110664 >> pkmem:2-max_used_size:: 110672 >> pkmem:2-free_size:: 66998200 >> pkmem:2-fragments:: 500 >> pkmem:3-total_size:: 67108864 >> pkmem:3-used_size:: 64832 >> pkmem:3-real_used_size:: 110664 >> pkmem:3-max_used_size:: 110672 >> pkmem:3-free_size:: 66998200 >> pkmem:3-fragments:: 500 >> pkmem:4-total_size:: 67108864 >> pkmem:4-used_size:: 6444664 >> pkmem:4-real_used_size:: 11838512 >> pkmem:4-max_used_size:: 11849192 >> pkmem:4-free_size:: 55270352 >> pkmem:4-fragments:: 223275 >> pkmem:5-total_size:: 67108864 >> pkmem:5-used_size:: 7769568 >> pkmem:5-real_used_size:: 14284264 >> pkmem:5-max_used_size:: 14294848 >> pkmem:5-free_size:: 52824600 >> pkmem:5-fragments:: 269975 >> pkmem:6-total_size:: 67108864 >> pkmem:6-used_size:: 120000 >> pkmem:6-real_used_size:: 167512 >> pkmem:6-max_used_size:: 175336 >> pkmem:6-free_size:: 66941352 >> pkmem:6-fragments:: 520 >> pkmem:7-total_size:: 67108864 >> pkmem:7-used_size:: 120000 >> pkmem:7-real_used_size:: 167584 >> pkmem:7-max_used_size:: 175272 >> pkmem:7-free_size:: 66941280 >> pkmem:7-fragments:: 520 >> pkmem:8-total_size:: 67108864 >> pkmem:8-used_size:: 119880 >> pkmem:8-real_used_size:: 165976 >> pkmem:8-max_used_size:: 165976 >> pkmem:8-free_size:: 66942888 >> pkmem:8-fragments:: 516 >> pkmem:9-total_size:: 67108864 >> pkmem:9-used_size:: 119952 >> pkmem:9-real_used_size:: 166288 >> pkmem:9-max_used_size:: 167024 >> pkmem:9-free_size:: 66942576 >> pkmem:9-fragments:: 518 >> pkmem:10-total_size:: 67108864 >> pkmem:10-used_size:: 119880 >> pkmem:10-real_used_size:: 165976 >> pkmem:10-max_used_size:: 165976 >> pkmem:10-free_size:: 66942888 >> pkmem:10-fragments:: 516 >> pkmem:11-total_size:: 67108864 >> pkmem:11-used_size:: 119880 >> pkmem:11-real_used_size:: 165976 >> pkmem:11-max_used_size:: 165976 >> pkmem:11-free_size:: 66942888 >> pkmem:11-fragments:: 516 >> pkmem:12-total_size:: 67108864 >> pkmem:12-used_size:: 119880 >> pkmem:12-real_used_size:: 165976 >> pkmem:12-max_used_size:: 165976 >> pkmem:12-free_size:: 66942888 >> pkmem:12-fragments:: 516 >> pkmem:13-total_size:: 67108864 >> pkmem:13-used_size:: 119952 >> pkmem:13-real_used_size:: 166288 >> pkmem:13-max_used_size:: 167024 >> pkmem:13-free_size:: 66942576 >> pkmem:13-fragments:: 518 >> pkmem:14-total_size:: 67108864 >> pkmem:14-used_size:: 119952 >> pkmem:14-real_used_size:: 166288 >> pkmem:14-max_used_size:: 167024 >> pkmem:14-free_size:: 66942576 >> pkmem:14-fragments:: 518 >> pkmem:15-total_size:: 67108864 >> pkmem:15-used_size:: 119880 >> pkmem:15-real_used_size:: 165976 >> pkmem:15-max_used_size:: 165976 >> pkmem:15-free_size:: 66942888 >> pkmem:15-fragments:: 516 >> pkmem:16-total_size:: 67108864 >> pkmem:16-used_size:: 119880 >> pkmem:16-real_used_size:: 165976 >> pkmem:16-max_used_size:: 165976 >> pkmem:16-free_size:: 66942888 >> pkmem:16-fragments:: 516 >> pkmem:17-total_size:: 67108864 >> pkmem:17-used_size:: 109904 >> pkmem:17-real_used_size:: 155832 >> pkmem:17-max_used_size:: 155832 >> pkmem:17-free_size:: 66953032 >> pkmem:17-fragments:: 504 >> [root at sbc sbin]# >> >> >> >> >> >> -- >> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608135.html >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> Users mailing list >> [hidden email] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >_______________________________________________ >Users mailing list >[hidden email] >http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >---------------------------------------------------------------------- >If you reply to this email, your message will be added to the discussion below: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608137.html >To unsubscribe from Does not release shared memory, click here . >NAML -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Does-not-release-shared-memory-tp7608065p7608192.html Sent from the 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