Re: [OpenSIPS-Users] nathelper vs nat_traversal
Hi Aqs, The two modules are mainly for doing SIP NAT traversal - they both do offer more or less the same functionality, just pick one :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.08.2015 23:34, Aqs Younas wrote: Hi, I am new to opensips. Could someone please tell me the differences between nathelper and nat_traversal? And in what scenario which module or both should be used. Thanks in Advance. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: Opensips 1.10 crash
Hi Denis, First of all, I strongly recommend you to upgrade to 1.11 as 1.10 is no longer supported. Secondly, it looks like you have a memory corruption . See this tutorial: http://www.opensips.org/Documentation/TroubleShooting-OutOfMem on how to enable memory debugging - it will give a more detailed report about the corruption. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.08.2015 07:20, dpa wrote: Hello! Is there any ideas about problem? Thank you. *From:*users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *dpa *Sent:* Wednesday, July 15, 2015 4:11 PM *To:* 'OpenSIPS users mailling list' *Subject:* [OpenSIPS-Users] Opensips 1.10 crash Hello! Opensips1.10.1. Working with rtpproxy2.0. Today Opensips has crashed. Core file was generated. Information from core file is in attachment. The question is why Opensips has crashed? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CARRIERROUTE module
Hi Ben, The carrierroute module is not maintained (for like 6 years). Still, if you want to do a routing based on both caller and caller number, you can do it with the drouting module. Use a set of prefixes to translate the caller prefix into a kind of routing group , which can be used again with drouting as set for searching the callee prefix. Basically you do 2 droutings, one for caller, one for callee. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.08.2015 18:23, Newlin, Ben wrote: Hi all, We have some interest in possibly using the CARRIERROUTE module instead of DROUTING as it allows prefix matching of the originating number as well as the destination, whereas DROUTING must match the originating user exactly. However, we noticed that the CARRIERROUTE module is labeled as alpha, even though it has existed in OpenSIPS for many years and many releases. Is there some reason this module is still alpha? Is it considered to be unstable for production use? Ben Newlin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Call-id issue in Cancel message generated by tm
I am not sure if its some bug or my mistake. I am using topology hiding module (opensips 2.1 version) and I have noticed that Call-id in Cancel message is different than Invite sent to gateway. Invite is sent to gateway and we get session progress but call is not picked up, as per fr_timer opensips triggers cancel but call ID in cancel is not same as sent in Invite INVITE sip:11025@2.2.2.2:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:9096;branch=z9hG4bKe5d5.fca1aeb4.0 Max-Forwards: 26 From: 786786 sip:786786@1.1.1.1:9096;tag=2B6F7HZaNH02F To: sip:11025@2.2.2.2:5060 Call-ID: DLGCH_LkhSV2VxNGNiEgQMMGRhYXxDSF9rcDN+K0QEC2Z7M2ouFFFd CSeq: 79309702 INVITE Contact: sip:sbc@1.1.1.1:9096;did=775.84850bf2 User-Agent: Opensips Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 258 v=0 o=- 22469580 22469580 IN IP4 1.1.1.1 s=Softphone c=IN IP4 1.1.1.1 t=0 0 m=audio 32706 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp:32707 CANCEL sip:11025@2.2.2.2:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:9096;branch=z9hG4bKe5d5.fca1aeb4.0 From: 786786 sip:786786@1.1.1.1:9096;tag=2B6F7HZaNH02F Call-ID: a87968d0-babc-1233-189c-d4ae52c9ad43 To: sip:11025@2.2.2.2:5060 CSeq: 79309702 CANCEL Max-Forwards: 70 Reason: SIP;cause=480;text=NO_ANSWER User-Agent: Opensips Content-Length: 0 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
