[asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee
symack at gmail.com
Tue Apr 9 14:18:21 CDT 2013
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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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 184.108.40.206:5060.
SIP Trunk (External) "BYE" <-> OpenSIPS (Internal)
Call-ID: 705605f129adbf5a38b5a0ff72de8f39 at 220.127.116.11:5060.
The call id was changed twice.... Could this be a two part problem?
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