[OpenSIPS-Users] [asterisk-users] 404 When BYE initiated by external callee

[Nick Khamis]

On Tue, Apr 9, 2013 at 3:22 PM, Joshua Colp <jcolp at digium.com> 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<595ad334-f06e97fa-3bbc8137 at 192.168.2.11>
>> <mailto:595ad334-f06e97fa-**3bbc8137 at 192.168.2.11<595ad334-f06e97fa-3bbc8137 at 192.168.2.11>
>> >.
>>
>>
>>
>> Asterisk (Internal) <-> SIP Trunk (External)
>>
>> Call-ID: 5a5fb47111cadd6146746c4446a179**0c at 70.10.163.44:5060<http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060>
>> <http://**5a5fb47111cadd6146746c4446a179**0c@70.10.163.44:5060/<http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060/>
>> >.
>>
>>
>>
>> SIP Trunk (External) "BYE" <-> OpenSIPS (Internal)
>>
>>
>> Call-ID: 705605f129adbf5a38b5a0ff72de8f**39 at 70.10.163.44:5060<http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060>
>> <http://**705605f129adbf5a38b5a0ff72de8f**39@70.10.163.44:5060/<http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060/>
>> >.
>>
>>
>>
>> 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
>
> --
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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:15178392000 at proxy.example.com:5060;user=phone SIP/2.0.
Call-ID: ccc1a3e7-bcfc28f1-ed2257c4 at 192.168.2.11.


U 2013/04/09 15:43:24.325964 192.168.2.5:5060 -> 192.168.2.10:5060
INVITE sip:1517839200 at asterisk.example.com:5060;user=phone SIP/2.0.
Call-ID: ccc1a3e7-bcfc28f1-ed2257c4 at 192.168.2.11.


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: ccc1a3e7-bcfc28f1-ed2257c4 at 192.168.2.11.


U 2013/04/09 15:43:24.396204 192.168.2.10:5060 -> 108.59.2.133:5060
INVITE sip:001110215178392000 at sbc.voxbeam.com SIP/2.0.
Call-ID: 58f65c9822f75d5a3da2992c0047c069 at 70.12.128.44:5060.


2013/04/09 15:44:15.086928 108.59.2.133:5060 -> 192.168.2.5:5060
BYE sip:1001 at 70.12.168.99:5060 SIP/2.0.
Call-ID: 58f65c9822f75d5a3da2992c0047c069 at 70.12.128.44:5060.


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: 58f65c9822f75d5a3da2992c0047c069 at 70.12.168.99:5060.


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: 58f65c9822f75d5a3da2992c0047c069 at 70.12.128.44:5060.


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: ccc1a3e7-bcfc28f1-ed2257c4 at 192.168.2.11.


Asterisk has mapped the call with the two different ids together.



 Any help is greatly appreciated,

Nick.
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