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<tt>Nick,<br>
<br>
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).<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 04/09/2013 09:23 PM, Nick Khamis wrote:
<blockquote
cite="mid:CAGWRaZY2Ua9baP7uzS+Z++tnKHGZDZb+-Ujw-6R1SKrcTNMQnQ@mail.gmail.com"
type="cite">
<div dir="ltr">On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu
<span dir="ltr"><<a moz-do-not-send="true"
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: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<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 moz-do-not-send="true" 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
moz-do-not-send="true"
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 moz-do-not-send="true"
href="http://192.168.2.5" target="_blank">192.168.2.5</a>:
OpenSIPS</div>
<div><a moz-do-not-send="true"
href="http://192.168.2.10" target="_blank">192.168.2.10</a>:
Asterisk</div>
<div><a moz-do-not-send="true"
href="http://70.10.163.44" target="_blank">70.10.163.44</a>:
Public IP<br>
</div>
<div><a moz-do-not-send="true"
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
moz-do-not-send="true"
href="http://192.168.2.10:5060"
target="_blank">192.168.2.10:5060</a>
-> <a moz-do-not-send="true"
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 moz-do-not-send="true"><sip:192.168.2.5;lr;did=392.62562fb2></a>.</div>
<div>From: "1001" <<a
moz-do-not-send="true"
href="mailto:sip%3A1001@server.example.com"
target="_blank">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A15178342008@server.example.com"
target="_blank">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a moz-do-not-send="true"
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 moz-do-not-send="true"
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
moz-do-not-send="true"
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
moz-do-not-send="true"
href="http://192.168.2.5:5060"
target="_blank">192.168.2.5:5060</a> ->
<a moz-do-not-send="true"
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 moz-do-not-send="true"><sip:192.168.2.5;lr;did=392.62562fb2></a>.</div>
<div>From: "1001" <<a
moz-do-not-send="true"
href="mailto:sip%3A1001@server.example.com"
target="_blank">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A15178342008@server.example.com"
target="_blank">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a moz-do-not-send="true"
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 moz-do-not-send="true"
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
moz-do-not-send="true"
href="http://108.59.2.133:5060"
target="_blank">108.59.2.133:5060</a>
-> <a moz-do-not-send="true"
href="http://192.168.2.5:5060"
target="_blank">192.168.2.5:5060</a></div>
<div>BYE <a moz-do-not-send="true"
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 moz-do-not-send="true"
href="mailto:sip%3A1001@70.10.163.44"
target="_blank">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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
moz-do-not-send="true"
href="http://192.168.2.5:5060"
target="_blank">192.168.2.5:5060</a> ->
<a moz-do-not-send="true"
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 moz-do-not-send="true"
href="mailto:sip%3A1001@70.10.163.44"
target="_blank">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com"
target="_blank">sip:001110215178342008@sbc.voxbeam.com</a>>;tag=3574513019-870807.</div>
<div>Call-ID: <a moz-do-not-send="true"
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
moz-do-not-send="true"
href="http://192.168.2.10:5060"
target="_blank">192.168.2.10:5060</a>
-> <a moz-do-not-send="true"
href="http://108.59.2.133:5060"
target="_blank">108.59.2.133:5060</a></div>
<div>INVITE <a moz-do-not-send="true"
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
moz-do-not-send="true"
href="mailto:sip%3A1001@70.10.163.44"
target="_blank">sip:1001@70.10.163.44</a>>;tag=as234a7f7d.</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com"
target="_blank">sip:001110215178342008@sbc.voxbeam.com</a>>.</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:1001@70.10.163.44:5060"
target="_blank">sip:1001@70.10.163.44:5060</a>>.</div>
<div>Call-ID: <a moz-do-not-send="true"
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>
_______________________________________________
Users mailing list
<a moz-do-not-send="true" href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a moz-do-not-send="true" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</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>
</blockquote>
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