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