<div dir="ltr"><div style><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Apr 9, 2013 at 3:22 PM, Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Nick Khamis wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div class="im">
<br>
Hello Joshua,<br>
<br>
Thanks again for your response. I can understand how * does not rewrite<br>
anything. When you mention the difference in call id, are you referring to:<br>
<br>
UA <-> OpenSIPS <-> Asterisk (Internal)<br>
<br>
Call-ID: <a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11" target="_blank">595ad334-f06e97fa-3bbc8137@<u></u>192.168.2.11</a><br></div>
<mailto:<a href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11" target="_blank">595ad334-f06e97fa-<u></u>3bbc8137@192.168.2.11</a>>.<div class="im"><br>
<br>
<br>
Asterisk (Internal) <-> SIP Trunk (External)<br>
<br>
Call-ID: <a href="http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060" target="_blank">5a5fb47111cadd6146746c4446a179<u></u>0c@70.10.163.44:5060</a><br></div>
<<a href="http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060/" target="_blank">http://<u></u>5a5fb47111cadd6146746c4446a179<u></u>0c@70.10.163.44:5060/</a>>.<div class="im"><br>
<br>
<br>
SIP Trunk (External) "BYE" <-> OpenSIPS (Internal)<br>
<br>
<br>
Call-ID: <a href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060" target="_blank">705605f129adbf5a38b5a0ff72de8f<u></u>39@70.10.163.44:5060</a><br></div>
<<a href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060/" target="_blank">http://<u></u>705605f129adbf5a38b5a0ff72de8f<u></u>39@70.10.163.44:5060/</a>>.<div class="im"><br>
<br>
<br>
The call id was changed twice.... Could this be a two part problem?<br>
</div></blockquote>
<br>
Yes. Until you can isolate it more it's all just a guess but it still doesn't seem like a problem with Asterisk itself.<div class=""><div class="h5"><br>
<br>
-- <br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
--<br>
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Hello Joshua,</div><div style><br></div><div style>Thanks again for your response. I re-ran the test, following a trace on the same call:</div><div style><br></div><div style><br></div><div style>192.168.2.11 - UA</div><div style>
192.168.2.5 - OpenSIPS Server</div><div style>192.168.2.10 - Asterisk Server</div><div style>108.59.2.133 - SIP Trunk<br></div><div style><br></div><div style><br></div><div style><div>U 2013/04/09 15:44:00.549096 <a href="http://192.168.2.11:5060">192.168.2.11:5060</a> -> <a href="http://192.168.2.5:5060">192.168.2.5:5060</a></div>
<div>INVITE sip:15178392000@proxy.example.com:5060;user=phone SIP/2.0.</div><div>Call-ID: <a href="mailto:ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11">ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11</a>.</div></div><div style><br></div>
<div style><br></div><div style><div>U 2013/04/09 15:43:24.325964 <a href="http://192.168.2.5:5060">192.168.2.5:5060</a> -> <a href="http://192.168.2.10:5060">192.168.2.10:5060</a></div><div>INVITE sip:1517839200@asterisk.example.com:5060;user=phone SIP/2.0.</div>
<div>Call-ID: <a href="mailto:ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11">ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11</a>.</div><div><br></div><div><br></div><div>U 2013/04/09 15:43:24.349274 <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/2.0 100 Trying.</div><div>Call-ID: <a href="mailto:ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11">ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11</a>.</div><div><br></div><div><br></div><div style>U 2013/04/09 15:43:24.396204 <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><br>
</div><div>INVITE <a href="mailto:sip%3A001110215178392000@sbc.voxbeam.com">sip:001110215178392000@sbc.voxbeam.com</a> SIP/2.0.</div><div>Call-ID: <a href="http://58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060">58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060</a>.</div>
<div><br></div><div><br></div><div>2013/04/09 15:44:15.086928 <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><br></div><div><div>BYE <a href="http://sip:1001@70.12.168.99:5060">sip:1001@70.12.168.99:5060</a> SIP/2.0.</div>
<div>Call-ID: <a href="http://58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060">58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060</a>.</div><div><br></div><div><br></div><div>U 2013/04/09 15:44:15.087277 <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>Call-ID: <a href="http://58f65c9822f75d5a3da2992c0047c069@70.12.168.99:5060">58f65c9822f75d5a3da2992c0047c069@70.12.168.99:5060</a>.</div></div></div></div><br></div><div class="gmail_extra">
<br></div><div class="gmail_extra" style>As I see asterisk rewrites the callid unexpectedly when initiating the INVITE with the SIP trunk (trace packet 4).</div><div class="gmail_extra" style>In the same trace packet 4, the Record-Route "Record-Route: <sip:192.168.2.5;lr;did=7ea.60b64711>." has also been</div>
<div class="gmail_extra" style>removed.</div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>I am sure this is a configuration issue on our part/end, and was wondering how others with proxy<-->asterisk integrations</div>
<div class="gmail_extra" style>addressed the issue. We can:</div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>1) Rule out the provider as the source of the problem when it comes to the changing of the callid</div>
<div class="gmail_extra" style>2) Relay the non loose route BYE from our proxy to asterisk, which has record of the new callid.</div><div class="gmail_extra" style>Not sure if this is a safe idea, or will even work?</div>
<div class="gmail_extra" style><br></div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>What is interesting to mention is the Session Progress:</div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>
<br></div><div class="gmail_extra" style><div class="gmail_extra">U 2013/04/09 15:43:32.211016 <a href="http://108.59.2.133:5060">108.59.2.133:5060</a> -> <a href="http://192.168.2.10:5060">192.168.2.10:5060</a></div><div class="gmail_extra">
SIP/2.0 183 Session Progress.</div><div class="gmail_extra">Call-ID: <a href="http://58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060">58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060</a>.<br></div><div class="gmail_extra">
<br></div><div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_extra">U 2013/04/09 15:43:32.214127 <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 class="gmail_extra">SIP/2.0 183 Session Progress.</div><div class="gmail_extra">Call-ID: <a href="mailto:ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11">ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11</a>.<br></div><div class="gmail_extra">
<br></div><div class="gmail_extra"><br></div><div class="gmail_extra" style>Asterisk has mapped the call with the two different ids together.</div></div></div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>
<br></div><div class="gmail_extra" style><br></div><div class="gmail_extra" style> Any help is greatly appreciated,</div><div class="gmail_extra" style><br></div><div class="gmail_extra" style>Nick.</div></div>