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:
<users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>> on
behalf of Eric Tamme <e...@uphreak.com<mailto:e...@uphreak.com>>
Reply-To: OpenSIPS users mailling list
<users@lists.opensips.org<mailto:users@lists.opensips.org>>
Date: Wednesday, January 13, 2016 at 3:44 PM
To: OpenSIPS users mailling list
<users@lists.opensips.org<mailto:users@lists.opensips.org>>
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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