Sorry for the top post!!!! N
On 4/10/13, Nick Khamis <[email protected]> wrote: > Hello Bogdan, > > Sorry for the missing info. The topology is the simple > > NAT Box <--> OpenSIPS <--> Asterisk > (192.168.2.1) (192.168.2.5) 192.168.2.10) > > I have pointed the problem to the regenerated asterisk invite: > > U 2013/04/09 15:43:24.396204 192.168.2.10:5060 -> 108.59.2.133:5060 > INVITE sip:[email protected] SIP/2.0. > Call-ID: [email protected]:5060. > > Where the callid was changed, and the RR was lost. The original INVITE > request > from the UA was as follows: > > U 2013/04/09 15:44:00.549096 192.168.2.11:5060 -> 192.168.2.5:5060 > INVITE sip:[email protected]:5060;user=phone SIP/2.0. > Call-ID: [email protected]. > > Surely others with OpenSIPS/Asterisk integrations experienced this > issue in the past? I > have found little solutions outside of implementing top hiding. As > mentioned earlier, asterisk has mapped the two Call-IDs together: > > U 2013/04/09 15:43:32.211016 108.59.2.133:5060 -> 192.168.2.10:5060 > SIP/2.0 183 Session Progress. > Call-ID: [email protected]:5060. > > U 2013/04/09 15:43:32.214127 192.168.2.10:5060 -> 192.168.2.5:5060 > SIP/2.0 183 Session Progress. > Call-ID: [email protected]. > > Would relaying the non-loose BYE to asterisk who has record of the > newly created callid work? > > > Thanks in Advance, > > N. > > > > > > On 4/10/13, Bogdan-Andrei Iancu <[email protected]> wrote: >> Nick, >> >> I do not know what is the topology of your SIP network, but the idea is >> that the BYE received by OpenSIPS does not contain proper routing >> information - now, either the BYE was wrongly generated by the end >> point, either it was wrongly changed on the way (if there are more hops >> between that end point and opensips). >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> On 04/09/2013 09:23 PM, Nick Khamis wrote: >>> On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu >>> <[email protected] <mailto:[email protected]>> wrote: >>> >>> Hi Nick, >>> >>> The BYE is not properly formed and rejected by script - in the 200 >>> OK of the INVITE, you can see that your opensips is doing >>> Record-Routing, but the BYE does not contain the corresponding >>> Route hdr, so SIP routing is impossible. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> >>> On 04/09/2013 08:05 PM, Nick Khamis wrote: >>>> Hello Everyone, >>>> >>>> I saw an earlier post about this issue: >>>> http://www.mail-archive.com/[email protected]/msg23052.html >>>> >>>> And was wondering if there was anything we can do on our end to >>>> fix this problem? It seems that providers are not obligated to >>>> maintain RR? When the caller (internal) initiates the BYE >>>> everything is ok, but not the case when the callee (external) >>>> initiates the BYE. >>>> >>>> 192.168.2.5 <http://192.168.2.5>: OpenSIPS >>>> 192.168.2.10 <http://192.168.2.10>: Asterisk >>>> 70.10.163.44 <http://70.10.163.44>: Public IP >>>> 108.59.2.133 <http://108.59.2.133>: Service Provider >>>> >>>> >>>> U 2013/04/09 12:17:02.920454 192.168.2.10:5060 >>>> <http://192.168.2.10:5060> -> 192.168.2.5:5060 >>>> <http://192.168.2.5:5060> >>>> SIP/2.0 200 OK. >>>> Via: SIP/2.0/UDP >>>> >>>> 192.168.2.5;branch=z9hG4bKac2e.554c6e93.0;received=192.168.2.5;rport=5060. >>>> Via: SIP/2.0/UDP >>>> >>>> 192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1. >>>> Record-Route: <sip:192.168.2.5;lr;did=392.62562fb2>. >>>> From: "1001" <sip:[email protected] >>>> <mailto:sip%[email protected]>>;tag=FCA0BFC0-B585477D. >>>> To: <sip:[email protected] >>>> >>>> <mailto:sip%[email protected]>;user=phone>;tag=as0a76fcde. >>>> Call-ID: [email protected] >>>> <mailto:[email protected]>. >>>> CSeq: 1 INVITE. >>>> Server: Asterisk PBX UNKNOWN__and_probably_unsupported. >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >>>> NOTIFY, INFO, PUBLISH. >>>> Supported: replaces, timer. >>>> Contact: <sip:[email protected]:5060 >>>> <http://sip:[email protected]:5060>>. >>>> Content-Type: application/sdp. >>>> Content-Length: 312. >>>> . >>>> v=0. >>>> o=root 1860889533 1860889534 IN IP4 192.168.2.10. >>>> s=Asterisk PBX UNKNOWN__and_probably_unsupported. >>>> c=IN IP4 192.168.2.10. >>>> t=0 0. >>>> m=audio 60646 RTP/AVP 18 101. >>>> a=rtpmap:18 G729/8000. >>>> a=fmtp:18 annexb=no. >>>> a=rtpmap:101 telephone-event/8000. >>>> a=fmtp:101 0-16. >>>> a=silenceSupp:off - - - -. >>>> a=ptime:20. >>>> a=sendrecv. >>>> >>>> ACC: transaction answered: >>>> >>>> timestamp=1365524222;method=INVITE;from_tag=FCA0BFC0-B585477D;to_tag=as0a76fcde;[email protected] >>>> <mailto:[email protected]>;code=200;reason=OK >>>> >>>> U 2013/04/09 12:17:02.939608 192.168.2.5:5060 >>>> <http://192.168.2.5:5060> -> 192.168.2.11:5060 >>>> <http://192.168.2.11:5060> >>>> SIP/2.0 200 OK. >>>> Via: SIP/2.0/UDP >>>> >>>> 192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1. >>>> Record-Route: <sip:192.168.2.5;lr;did=392.62562fb2>. >>>> From: "1001" <sip:[email protected] >>>> <mailto:sip%[email protected]>>;tag=FCA0BFC0-B585477D. >>>> To: <sip:[email protected] >>>> >>>> <mailto:sip%[email protected]>;user=phone>;tag=as0a76fcde. >>>> Call-ID: [email protected] >>>> <mailto:[email protected]>. >>>> CSeq: 1 INVITE. >>>> Server: Asterisk PBX UNKNOWN__and_probably_unsupported. >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >>>> NOTIFY, INFO, PUBLISH. >>>> Supported: replaces, timer. >>>> Contact: <sip:[email protected]:5060 >>>> <http://sip:[email protected]:5060>>. >>>> Content-Type: application/sdp. >>>> Content-Length: 329. >>>> . >>>> v=0. >>>> o=root 1860889533 1860889534 IN IP4 192.168.2.10. >>>> s=Asterisk PBX UNKNOWN__and_probably_unsupported. >>>> c=IN IP4 192.168.2.5. >>>> t=0 0. >>>> m=audio 31148 RTP/AVP 18 101. >>>> a=rtpmap:18 G729/8000. >>>> a=fmtp:18 annexb=no. >>>> a=rtpmap:101 telephone-event/8000. >>>> a=fmtp:101 0-16. >>>> a=silenceSupp:off - - - -. >>>> a=ptime:20. >>>> a=sendrecv. >>>> a=nortpproxy:yes. >>>> >>>> >>>> >>>> U 2013/04/09 12:17:06.988918 108.59.2.133:5060 >>>> <http://108.59.2.133:5060> -> 192.168.2.5:5060 >>>> <http://192.168.2.5:5060> >>>> BYE sip:[email protected]:5060 >>>> <http://sip:[email protected]:5060> SIP/2.0. >>>> Max-Forwards: 64. >>>> To: "1001" <sip:[email protected] >>>> <mailto:sip%[email protected]>>;tag=as4b40d9b4. >>>> From: <sip:[email protected] >>>> >>>> <mailto:sip%[email protected]>>;tag=3574513019-870807. >>>> Reason: Q.850;cause=16;text="". >>>> Call-ID: [email protected]:5060 >>>> <http://[email protected]:5060>. >>>> CSeq: 2 BYE. >>>> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, >>>> REFER, SUBSCRIBE, PRACK, UPDATE. >>>> Via: SIP/2.0/UDP 108.59.2.133;branch=z9hG4bK2deb.8bfd0b06.0. >>>> Contact: <sip:[email protected] >>>> <mailto:sip%[email protected]>;did=e9e.a6618961>. >>>> Allow-Events: as-feature-event. >>>> Allow-Events: call-info. >>>> Allow-Events: presence. >>>> Allow-Events: line-seize. >>>> Allow-Events: dialog. >>>> Allow-Events: refer. >>>> Allow-Events: message-summary. >>>> Content-Length: 0. >>>> . >>>> >>>> Forcing RPORT: sip:[email protected] >>>> <mailto:sip%[email protected]> >>>> >>>> U 2013/04/09 12:17:06.989421 192.168.2.5:5060 >>>> <http://192.168.2.5:5060> -> 108.59.2.133:5060 >>>> <http://108.59.2.133:5060> >>>> SIP/2.0 404 Not here. >>>> To: "1001" <sip:[email protected] >>>> <mailto:sip%[email protected]>>;tag=as4b40d9b4. >>>> From: <sip:[email protected] >>>> >>>> <mailto:sip%[email protected]>>;tag=3574513019-870807. >>>> Call-ID: [email protected]:5060 >>>> <http://[email protected]:5060>. >>>> CSeq: 2 BYE. >>>> Via: SIP/2.0/UDP >>>> >>>> 108.59.2.133;received=108.59.2.133;rport=5060;branch=z9hG4bK2deb.8bfd0b06.0. >>>> Content-Length: 0. >>>> >>>> >>>> Or is asterisk the culprit? Looking at the forwarded INVITE (on >>>> the asterisk server), I see that the RR has been re-written, as >>>> opposed to appended when contacting the provider: >>>> >>>> >>>> U 2013/04/09 12:52:52.109611 192.168.2.10:5060 >>>> <http://192.168.2.10:5060> -> 108.59.2.133:5060 >>>> <http://108.59.2.133:5060> >>>> INVITE sip:[email protected] >>>> <mailto:sip%[email protected]> SIP/2.0. >>>> Via: SIP/2.0/UDP 70.10.163.44:5060;branch=z9hG4bK75a764b9;rport. >>>> Max-Forwards: 70. >>>> From: "1001" <sip:[email protected] >>>> <mailto:sip%[email protected]>>;tag=as234a7f7d. >>>> To: <sip:[email protected] >>>> <mailto:sip%[email protected]>>. >>>> Contact: <sip:[email protected]:5060 >>>> <http://sip:[email protected]:5060>>. >>>> Call-ID: [email protected]:5060 >>>> <http://[email protected]:5060>. >>>> CSeq: 102 INVITE. >>>> User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported. >>>> Date: Tue, 09 Apr 2013 16:52:52 GMT. >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >>>> NOTIFY, INFO, PUBLISH. >>>> Supported: replaces, timer. >>>> Content-Type: application/sdp. >>>> Content-Length: 310. >>>> . >>>> v=0. >>>> o=root 731333659 731333659 IN IP4 70.10.163.44. >>>> s=Asterisk PBX UNKNOWN__and_probably_unsupported. >>>> c=IN IP4 70.10.163.44. >>>> t=0 0. >>>> m=audio 30434 RTP/AVP 18 101. >>>> a=rtpmap:18 G729/8000. >>>> a=fmtp:18 annexb=no. >>>> a=rtpmap:101 telephone-event/8000. >>>> a=fmtp:101 0-16. >>>> a=silenceSupp:off - - - -. >>>> a=ptime:20. >>>> a=sendrecv. >>>> >>>> >>>> Can we get an externally initiated BYE working in an >>>> OpenSIPS->Asterisk integration? If so, some suggestions would be >>>> appreciated. Maybe just really the non-loose route BYE to asterisk? >>>> Is adding topology hiding functionality a cumbersome task... >>>> >>>> Thanks in Advance, >>>> >>>> N. >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] <mailto:[email protected]> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> Is our asterisk server not relaying the RR along with the INVITE? If >>> so, can we configure the PBX to do so using one of it's variables? * >>> Mailing list CC'ed in this email... >>> >>> >>> N. >> > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
