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

[Nick Khamis]

On Tue, Apr 9, 2013 at 2:31 PM, Joshua Colp <jcolp at digium.com> wrote:

> Nick Khamis wrote:
>
>> Is our asterisk server not relaying the RR along with the INVITE? If so,
>> can we configure the PBX to do so using one of it's variables? * Mailing
>> list CC'ed in this email...
>>
>
> Asterisk is not a SIP proxy, it does not forward or relay INVITEs. It is a
> back to back user agent. Each leg is individual.
>
> Cheers,
>
> --
> 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>



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.


Thanks in Advance,

Nick.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130409/fbda22b4/attachment.htm>