<div dir="ltr">On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><div class="gmail_extra"><div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div bgcolor="#ffffff" text="#000000">
<tt>Hi Nick,<br>
<br>
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.<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre><div><div class="h5">
<br>
On 04/09/2013 08:05 PM, Nick Khamis wrote:
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">Hello Everyone,
<div><br>
</div>
<div>I saw an earlier post about this issue: <a href="http://www.mail-archive.com/users@lists.opensips.org/msg23052.html" target="_blank">http://www.mail-archive.com/users@lists.opensips.org/msg23052.html</a></div>
<div><br>
</div>
<div>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><br>
</div>
<div><a href="http://192.168.2.5" target="_blank">192.168.2.5</a>: OpenSIPS</div>
<div><a href="http://192.168.2.10" target="_blank">192.168.2.10</a>: Asterisk</div>
<div><a href="http://70.10.163.44" target="_blank">70.10.163.44</a>: Public IP<br>
</div>
<div><a href="http://108.59.2.133" target="_blank">108.59.2.133</a>: Service
Provider<br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div>U 2013/04/09 12:17:02.920454 <a href="http://192.168.2.10:5060" target="_blank">192.168.2.10:5060</a>
-> <a href="http://192.168.2.5:5060" target="_blank">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:
<a><sip:192.168.2.5;lr;did=392.62562fb2></a>.</div>
<div>From: "1001" <<a href="mailto:sip%3A1001@server.example.com" target="_blank">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div>
<div>To: <<a href="mailto:sip%3A15178342008@server.example.com" target="_blank">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">192.168.2.5:5060</a> ->
<a href="http://192.168.2.11:5060" target="_blank">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:
<a><sip:192.168.2.5;lr;did=392.62562fb2></a>.</div>
<div>From: "1001" <<a href="mailto:sip%3A1001@server.example.com" target="_blank">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div>
<div>To: <<a href="mailto:sip%3A15178342008@server.example.com" target="_blank">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11" target="_blank">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" target="_blank">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" target="_blank">108.59.2.133:5060</a>
-> <a href="http://192.168.2.5:5060" target="_blank">192.168.2.5:5060</a></div>
<div>BYE <a href="http://sip:1001@70.10.163.44:5060" target="_blank">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" target="_blank">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">192.168.2.5:5060</a> ->
<a href="http://108.59.2.133:5060" target="_blank">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" target="_blank">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com" target="_blank">sip:001110215178342008@sbc.voxbeam.com</a>>;tag=3574513019-870807.</div>
<div>Call-ID: <a href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060" target="_blank">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>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><br>
</div>
<div><br>
</div>
<div>
<div>U 2013/04/09 12:52:52.109611 <a href="http://192.168.2.10:5060" target="_blank">192.168.2.10:5060</a>
-> <a href="http://108.59.2.133:5060" target="_blank">108.59.2.133:5060</a></div>
<div>INVITE <a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com" target="_blank">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" target="_blank">sip:1001@70.10.163.44</a>>;tag=as234a7f7d.</div>
<div>To: <<a href="mailto:sip%3A001110215178342008@sbc.voxbeam.com" target="_blank">sip:001110215178342008@sbc.voxbeam.com</a>>.</div>
<div>Contact: <<a href="http://sip:1001@70.10.163.44:5060" target="_blank">sip:1001@70.10.163.44:5060</a>>.</div>
<div>Call-ID: <a href="http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060" target="_blank">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>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>Is adding topology hiding functionality
a cumbersome task...</div>
<div><br>
</div>
<div>Thanks in Advance,</div>
<div><br>
</div>
<div>N.</div>
<div><br>
</div>
</div>
</div>
</div>
</div></div><pre><fieldset></fieldset>
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</pre>
</blockquote>
</div>
</blockquote></div><br></div><div class="gmail_extra"><br></div><div class="gmail_extra" style>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...</div>
<div class="gmail_extra" style><br></div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>N.</div></div>