On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
**
Hi Nick,
The BYE is not properly formed and rejected by script - in the 200 OK of
the INVITE, you can see that your opensips is doing Record-Routing, but the
BYE does not contain the corresponding Route hdr, so
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.
On Tue, Apr 9, 2013 at 2:31 PM, Joshua Colp jc...@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,
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
On Tue, Apr 9, 2013 at 3:04 PM, Joshua Colp jc...@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
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
On Tue, Apr 9, 2013 at 3:22 PM, Joshua Colp jc...@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)
On Tuesday, April 09, 2013 1:31 PM, Nick Khamis wrote:
As I see asterisk rewrites the callid unexpectedly when initiating the
INVITE with the SIP trunk (trace packet 4).
[...]
Asterisk has mapped the call with the two different ids together.
Nick,
As Joshua has already tried to explain to
