[OpenSIPS-Users] T_Relay Re-Invite

[Dave Lechasseur]

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