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

[Nick Khamis]

On Tue, Apr 9, 2013 at 3:04 PM, Joshua Colp <jcolp at digium.com> wrote:

> Nick Khamis wrote:
>
>>
>> Hey Joshua,
>>
>> It was a poor choice of words on my part. What I meant to say was
>> whether the problem was due to our asterisk configuration re-writing
>> the RR when initiating the INVITE to our SIP trunk provider. Not sure if
>> you had looked at the SIP trace included in the original email? If not
>> I can resend it.
>>
>
> I saw, but my response stands. Asterisk does not rewrite anything. The
> outgoing leg to your SIP trunk is completely separate, it is not a
> forwarded/modified INVITE. With the information you have available I don't
> think Asterisk is the problem here. The traces also illustrate this, the
> BYE in the trace is from a completely different call than the other
> messages. (You can see by looking at the Call-ID).
>
>
> --
> 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 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 at 192.168.2.11.


Asterisk (Internal) <-> SIP Trunk (External)

Call-ID: 5a5fb47111cadd6146746c4446a1790c at 70.10.163.44:5060.


SIP Trunk (External) "BYE" <-> OpenSIPS (Internal)


Call-ID: 705605f129adbf5a38b5a0ff72de8f39 at 70.10.163.44:5060.


The call id was changed twice.... Could this be a two part problem?

N.
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