On Tue, Apr 9, 2013 at 3:22 PM, Joshua Colp <[email protected]> wrote:
> Nick Khamis wrote: > >> >> Hello Joshua, >> >> Thanks again for your response. I can understand how * does not rewrite >> anything. When you mention the difference in call id, are you referring >> to: >> >> UA <-> OpenSIPS <-> Asterisk (Internal) >> >> Call-ID: >> 595ad334-f06e97fa-3bbc8137@**192.168.2.11<[email protected]> >> <mailto:595ad334-f06e97fa-**[email protected]<[email protected]> >> >. >> >> >> >> Asterisk (Internal) <-> SIP Trunk (External) >> >> Call-ID: >> 5a5fb47111cadd6146746c4446a179**[email protected]:5060<http://[email protected]:5060> >> <http://**5a5fb47111cadd6146746c4446a179**[email protected]:5060/<http://[email protected]:5060/> >> >. >> >> >> >> SIP Trunk (External) "BYE" <-> OpenSIPS (Internal) >> >> >> Call-ID: >> 705605f129adbf5a38b5a0ff72de8f**[email protected]:5060<http://[email protected]:5060> >> <http://**705605f129adbf5a38b5a0ff72de8f**[email protected]:5060/<http://[email protected]:5060/> >> >. >> >> >> >> The call id was changed twice.... Could this be a two part problem? >> > > 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. > > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> > Hello Joshua, Thanks again for your response. I re-ran the test, following a trace on the same call: 192.168.2.11 - UA 192.168.2.5 - OpenSIPS Server 192.168.2.10 - Asterisk Server 108.59.2.133 - SIP Trunk U 2013/04/09 15:44:00.549096 192.168.2.11:5060 -> 192.168.2.5:5060 INVITE sip:[email protected]:5060;user=phone SIP/2.0. Call-ID: [email protected]. U 2013/04/09 15:43:24.325964 192.168.2.5:5060 -> 192.168.2.10:5060 INVITE sip:[email protected]:5060;user=phone SIP/2.0. Call-ID: [email protected]. U 2013/04/09 15:43:24.349274 192.168.2.10:5060 -> 192.168.2.5:5060 SIP/2.0 100 Trying. Call-ID: [email protected]. U 2013/04/09 15:43:24.396204 192.168.2.10:5060 -> 108.59.2.133:5060 INVITE sip:[email protected] SIP/2.0. Call-ID: [email protected]:5060. 2013/04/09 15:44:15.086928 108.59.2.133:5060 -> 192.168.2.5:5060 BYE sip:[email protected]:5060 SIP/2.0. Call-ID: [email protected]:5060. U 2013/04/09 15:44:15.087277 192.168.2.5:5060 -> 108.59.2.133:5060 SIP/2.0 404 Not here. Call-ID: [email protected]:5060. As I see asterisk rewrites the callid unexpectedly when initiating the INVITE with the SIP trunk (trace packet 4). In the same trace packet 4, the Record-Route "Record-Route: <sip:192.168.2.5;lr;did=7ea.60b64711>." has also been removed. I am sure this is a configuration issue on our part/end, and was wondering how others with proxy<-->asterisk integrations addressed the issue. We can: 1) Rule out the provider as the source of the problem when it comes to the changing of the callid 2) Relay the non loose route BYE from our proxy to asterisk, which has record of the new callid. Not sure if this is a safe idea, or will even work? What is interesting to mention is the Session Progress: U 2013/04/09 15:43:32.211016 108.59.2.133:5060 -> 192.168.2.10:5060 SIP/2.0 183 Session Progress. Call-ID: [email protected]:5060. U 2013/04/09 15:43:32.214127 192.168.2.10:5060 -> 192.168.2.5:5060 SIP/2.0 183 Session Progress. Call-ID: [email protected]. Asterisk has mapped the call with the two different ids together. Any help is greatly appreciated, Nick.
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