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: OpenSIPS 192.168.2.10: Asterisk 70.10.163.44: Public IP 108.59.2.133: Service Provider U 2013/04/09 12:17:02.920454 192.168.2.10:5060 -> 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]>;tag=FCA0BFC0-B585477D. To: <sip:[email protected];user=phone>;tag=as0a76fcde. Call-ID: [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>. 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;call_id= [email protected];code=200;reason=OK U 2013/04/09 12:17:02.939608 192.168.2.5:5060 -> 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]>;tag=FCA0BFC0-B585477D. To: <sip:[email protected];user=phone>;tag=as0a76fcde. Call-ID: [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>. 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 -> 192.168.2.5:5060 BYE sip:[email protected]:5060 SIP/2.0. Max-Forwards: 64. To: "1001" <sip:[email protected]>;tag=as4b40d9b4. From: <sip:[email protected]>;tag=3574513019-870807. Reason: Q.850;cause=16;text="". Call-ID: [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];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] U 2013/04/09 12:17:06.989421 192.168.2.5:5060 -> 108.59.2.133:5060 SIP/2.0 404 Not here. To: "1001" <sip:[email protected]>;tag=as4b40d9b4. From: <sip:[email protected]>;tag=3574513019-870807. Call-ID: [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 -> 108.59.2.133:5060 INVITE 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]>;tag=as234a7f7d. To: <sip:[email protected]>. Contact: <sip:[email protected]:5060>. Call-ID: [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.
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