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

[Simon Gajski]

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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