Hi Dave,
Regards,
Assuming you do the record-routing for the initial INVITE, can you post
a SIP capture showing the call flow from the OpenSIPS perspective
(containing both legs).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 13.01.2016 22:54, Dave Lechasseur wrote:
Hi Eric,
Thank you for the quick answer.
With record_route it go thought the proxy but every in-session
messages aren’t sent to the PBX or to the phone from the PBX.
Just to make sure I’ve tried adding : xlog("DEBUG : METHOD $rm”);
right after : route {
and I don’t pick any log from it.
Thank you,
Dave L.
From: <[email protected]
<mailto:[email protected]>> on behalf of Eric Tamme
<[email protected] <mailto:[email protected]>>
Reply-To: OpenSIPS users mailling list <[email protected]
<mailto:[email protected]>>
Date: Wednesday, January 13, 2016 at 3:44 PM
To: OpenSIPS users mailling list <[email protected]
<mailto:[email protected]>>
Subject: Re: [OpenSIPS-Users] T_Relay Re-Invite
You need to record route the initial request so that your proxy stays
in the signalling path. This doesnt have anything to do with the tm
module, but rather building up the routeset and loose routing.
-Eric
On 01/13/2016 01:42 PM, Dave Lechasseur wrote:
Hi everyone,
I have a problem with t_relay.
When the session is established (last 200 OK) the phone receive a
in-dialog invite from the PBX directly and since there every packet
don’t go thought OpenSIPS but goes directly to the PBX meaning that I
have no way to do anything on the active channel and I don’t receive
the BYE message on OpenSIPS, it go directly between the phone and the
PBX.
Is there a way to not have this ?
Thank you for your help !
Dave L.
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