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<font face="monospace">Hi Simon,<br>
<br>
If you have OpenSIPS and a simple SIP proxy (not a B2B), you must
not reply to the REFER, but to let it be routed as an in-dialog
request to the other end-point, as that end-point will actually do
the transfer (starting the new call).<br>
<br>
Regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 23rd May - 3rd June 2022
<a class="moz-txt-link-freetext" href="https://opensips.org/training/OpenSIPS_eBootcamp_2022/">https://opensips.org/training/OpenSIPS_eBootcamp_2022/</a></pre>
<div class="moz-cite-prefix">On 3/2/22 12:02 PM, Simon Gajski wrote:<br>
</div>
<blockquote type="cite"
cite="mid:f0fb2d68-6a0e-d3a2-b940-55d84e01c92b@softnet.si">
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<p>Hi<br>
<br>
no, we don't use B2B on OpenSIPS side. Is this the correctway
to do it?<br>
The thing is that I don't know what would be the best way to
implement this with use of RTPengine.<br>
I found very little info available online.<br>
<br>
Call forwards that I manually set in DB (cfu, cfnr, cfb.....like
we were doing on last bootcamp) are working fine.<br>
Only issue is with answered call and then attempting to transfer
it.<br>
</p>
<p>BR<br>
Simon<br>
<br>
<br>
<br>
</p>
<div class="moz-cite-prefix">Bogdan-Andrei Iancu je 01.03.2022 ob
15:30 napisal:<br>
</div>
<blockquote type="cite"
cite="mid:21daf27a-bfec-04d5-184c-55f1deeef1b2@opensips.org">
<meta http-equiv="Content-Type" content="text/html;
charset=UTF-8">
<font face="monospace">Hi Simon,<br>
<br>
Do you use B2B on the OpenSIPS side ? Which entity is actually
performing the transfer ?<br>
<br>
Regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com" moz-do-not-send="true">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp
<a class="moz-txt-link-freetext" href="https://www.opensips.org/Training/Bootcamp" moz-do-not-send="true">https://www.opensips.org/Training/Bootcamp</a></pre>
<div class="moz-cite-prefix">On 2/24/22 1:54 PM, Simon Gajski
via Users wrote:<br>
</div>
<blockquote type="cite"
cite="mid:55ddf70b-10a8-6c1f-6d04-f81f3bce311a@softnet.si">
<meta http-equiv="content-type" content="text/html;
charset=UTF-8">
<br>
<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>
</div>
<br>
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