<div dir="ltr">Hello Everyone,<div><br></div><div style>I saw an earlier post about this issue: <a href="http://www.mail-archive.com/users@lists.opensips.org/msg23052.html">http://www.mail-archive.com/users@lists.opensips.org/msg23052.html</a></div>
<div style><br></div><div style>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.</div>
<div style><br></div><div style><a href="http://192.168.2.5">192.168.2.5</a>: OpenSIPS</div><div style><a href="http://192.168.2.10">192.168.2.10</a>: Asterisk</div><div style><a href="http://70.10.163.44">70.10.163.44</a>: Public IP<br>
</div><div style><a href="http://108.59.2.133">108.59.2.133</a>: Service Provider<br></div><div style><br></div><div style><br></div><div style><div>U 2013/04/09 12:17:02.920454 <a href="http://192.168.2.10:5060">192.168.2.10:5060</a> -> <a href="http://192.168.2.5:5060">192.168.2.5:5060</a></div>
<div>SIP/2.0 200 OK.</div><div>Via: SIP/2.0/UDP 192.168.2.5;branch=z9hG4bKac2e.554c6e93.0;received=192.168.2.5;rport=5060.</div><div>Via: SIP/2.0/UDP 192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1.</div>
<div>Record-Route: <sip:192.168.2.5;lr;did=392.62562fb2>.</div><div>From: "1001" <<a href="mailto:sip%3A1001@server.example.com">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div><div>To: <<a href="mailto:sip%3A15178342008@server.example.com">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11">595ad334-f06e97fa-3bbc8137@192.168.2.11</a>.</div><div>CSeq: 1 INVITE.</div><div>Server: Asterisk PBX UNKNOWN__and_probably_unsupported.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.</div>
<div>Supported: replaces, timer.</div><div>Contact: <<a href="http://sip:15178342008@192.168.2.10:5060">sip:15178342008@192.168.2.10:5060</a>>.</div><div>Content-Type: application/sdp.</div><div>Content-Length: 312.</div>
<div>.</div><div>v=0.</div><div>o=root 1860889533 1860889534 IN IP4 192.168.2.10.</div><div>s=Asterisk PBX UNKNOWN__and_probably_unsupported.</div><div>c=IN IP4 192.168.2.10.</div><div>t=0 0.</div><div>m=audio 60646 RTP/AVP 18 101.</div>
<div>a=rtpmap:18 G729/8000.</div><div>a=fmtp:18 annexb=no.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=silenceSupp:off - - - -.</div><div>a=ptime:20.</div><div>a=sendrecv.</div><div>
<br></div><div>ACC: transaction answered: timestamp=1365524222;method=INVITE;from_tag=FCA0BFC0-B585477D;to_tag=as0a76fcde;call_id=<a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11">595ad334-f06e97fa-3bbc8137@192.168.2.11</a>;code=200;reason=OK</div>
<div><br></div><div>U 2013/04/09 12:17:02.939608 <a href="http://192.168.2.5:5060">192.168.2.5:5060</a> -> <a href="http://192.168.2.11:5060">192.168.2.11:5060</a></div><div>SIP/2.0 200 OK.</div><div>Via: SIP/2.0/UDP 192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1.</div>
<div>Record-Route: <sip:192.168.2.5;lr;did=392.62562fb2>.</div><div>From: "1001" <<a href="mailto:sip%3A1001@server.example.com">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div><div>To: <<a href="mailto:sip%3A15178342008@server.example.com">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11">595ad334-f06e97fa-3bbc8137@192.168.2.11</a>.</div><div>CSeq: 1 INVITE.</div><div>Server: Asterisk PBX UNKNOWN__and_probably_unsupported.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.</div>
<div>Supported: replaces, timer.</div><div>Contact: <<a href="http://sip:15178342008@192.168.2.10:5060">sip:15178342008@192.168.2.10:5060</a>>.</div><div>Content-Type: application/sdp.</div><div>Content-Length: 329.</div>
<div>.</div><div>v=0.</div><div>o=root 1860889533 1860889534 IN IP4 192.168.2.10.</div><div>s=Asterisk PBX UNKNOWN__and_probably_unsupported.</div><div>c=IN IP4 192.168.2.5.</div><div>t=0 0.</div><div>m=audio 31148 RTP/AVP 18 101.</div>
<div>a=rtpmap:18 G729/8000.</div><div>a=fmtp:18 annexb=no.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=silenceSupp:off - - - -.</div><div>a=ptime:20.</div><div>a=sendrecv.</div><div>
a=nortpproxy:yes.</div><div><br></div><div><br></div><div><br></div><div>U 2013/04/09 12:17:06.988918 <a href="http://108.59.2.133:5060">108.59.2.133:5060</a> -> <a href="http://192.168.2.5:5060">192.168.2.5:5060</a></div>
<div>BYE <a href="http://sip:1001@70.10.163.44:5060">sip:1001@70.10.163.44:5060</a> SIP/2.0.</div><div>Max-Forwards: 64.</div><div>To: "1001" <<a href="mailto:sip%3A1001@70.10.163.44">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>>;tag=3574513019-870807.</div><div>Reason: Q.850;cause=16;text="".</div><div>Call-ID: <a href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060">705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060</a>.</div>
<div>CSeq: 2 BYE.</div><div>Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.</div><div>Via: SIP/2.0/UDP 108.59.2.133;branch=z9hG4bK2deb.8bfd0b06.0.</div><div>Contact: <<a href="mailto:sip%3Acallee@108.59.2.133">sip:callee@108.59.2.133</a>;did=e9e.a6618961>.</div>
<div>Allow-Events: as-feature-event.</div><div>Allow-Events: call-info.</div><div>Allow-Events: presence.</div><div>Allow-Events: line-seize.</div><div>Allow-Events: dialog.</div><div>Allow-Events: refer.</div><div>Allow-Events: message-summary.</div>
<div>Content-Length: 0.</div><div>.</div><div><br></div><div>Forcing RPORT: <a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a></div><div><br></div><div>U 2013/04/09 12:17:06.989421 <a href="http://192.168.2.5:5060">192.168.2.5:5060</a> -> <a href="http://108.59.2.133:5060">108.59.2.133:5060</a></div>
<div>SIP/2.0 404 Not here.</div><div>To: "1001" <<a href="mailto:sip%3A1001@70.10.163.44">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div><div>From: <<a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>>;tag=3574513019-870807.</div>
<div>Call-ID: <a href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060">705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060</a>.</div><div>CSeq: 2 BYE.</div><div>Via: SIP/2.0/UDP 108.59.2.133;received=108.59.2.133;rport=5060;branch=z9hG4bK2deb.8bfd0b06.0.</div>
<div>Content-Length: 0.</div><div><br></div><div><br></div><div style>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:</div>
<div style><br></div><div style><br></div><div style><div>U 2013/04/09 12:52:52.109611 <a href="http://192.168.2.10:5060">192.168.2.10:5060</a> -> <a href="http://108.59.2.133:5060">108.59.2.133:5060</a></div><div>INVITE <a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a> SIP/2.0.</div>
<div>Via: SIP/2.0/UDP 70.10.163.44:5060;branch=z9hG4bK75a764b9;rport.</div><div>Max-Forwards: 70.</div><div>From: "1001" <<a href="mailto:sip%3A1001@70.10.163.44">sip:1001@70.10.163.44</a>>;tag=as234a7f7d.</div>
<div>To: <<a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>>.</div><div>Contact: <<a href="http://sip:1001@70.10.163.44:5060">sip:1001@70.10.163.44:5060</a>>.</div>
<div>Call-ID: <a href="http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060">5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060</a>.</div><div>CSeq: 102 INVITE.</div><div>User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported.</div>
<div>Date: Tue, 09 Apr 2013 16:52:52 GMT.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.</div><div>Supported: replaces, timer.</div><div>Content-Type: application/sdp.</div><div>
Content-Length: 310.</div><div>.</div><div>v=0.</div><div>o=root 731333659 731333659 IN IP4 70.10.163.44.</div><div>s=Asterisk PBX UNKNOWN__and_probably_unsupported.</div><div>c=IN IP4 70.10.163.44.</div><div>t=0 0.</div>
<div>m=audio 30434 RTP/AVP 18 101.</div><div>a=rtpmap:18 G729/8000.</div><div>a=fmtp:18 annexb=no.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=silenceSupp:off - - - -.</div><div>a=ptime:20.</div>
<div>a=sendrecv.</div><div><br></div><div><br></div><div style>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?</div>
<div style>Is adding topology hiding functionality a cumbersome task...</div><div style><br></div><div style>Thanks in Advance,</div><div style><br></div><div style>N.</div><div><br></div></div></div></div>