[OpenSIPS-Users] T_Relay Re-Invite

[Bogdan-Andrei Iancu]

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: <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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>
>
>
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