[OpenSIPS-Users] Attended call transfer in opensips with use of RTPengine

[Bogdan-Andrei Iancu]

Hi Simon,

Do you use B2B on the OpenSIPS side ? Which entity is actually 
performing the transfer ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp
   https://www.opensips.org/Training/Bootcamp

On 2/24/22 1:54 PM, Simon Gajski via Users wrote:
>
> Hi
>
>
> I am using opensips 3.2 with rtpengine on same server and trying to 
> achieve attended call transfer.
>
> In theory, I'm trying to do:
> 1. A calls B...and B answers
> 2. B puts A on hold (MOH is played from RTPengine)
> 3. B calls C...and C answers
>
> Now the funny part:
> B tries to transfer A to C and sends REFER to opensips
> In opensips I responds with 202 Accepted and B gets disconnected.
>
> However A and C don't get connected together
> A still receives MOH and C has no voice
>
> We have another installation of opensips where REFER handles 
> Freeswitch, and there such type of transfer is working fine.
>
> Can someone help me how to handle such call behaviour in opensips with 
> RTPengine?
>
>
> relevant part of code:
>
> route[handle_sequential]{
> ...
> if(is_method("REFER")) {
>         xlog("[IN_DIALOG] [$rm] Transfer from $fu to $tu");
>         send_reply(202, "Accepted");
>
>         #what next?
>
>         exit;
>     }
> ...
> }
>
>
> Thank you!
>
> Simon
>
>
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