Hi Andreas,<div><br></div><div>I don't see why you can't do this. In your routing script you can look for the update message (there should be something about this update message that you can detect it), make changes as required and generate a response to any destination you want. This is what the routing script (opensips.cfg) is for.</div>
<div><br></div><div>Regards,</div><div>Ali Pey<br><br><div class="gmail_quote">On Tue, Aug 28, 2012 at 8:06 AM, Andreas Sikkema <span dir="ltr"><<a href="mailto:h323@ramdyne.nl" target="_blank">h323@ramdyne.nl</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
I know what I'm trying to do is very much frowned upon, but I'm<br>
hitting walls everywhere to make this work.<br>
<br>
We have this platform that is using what seems to be RFC3311 UPDATE<br>
messages as part of a call transfer. A calls B, B answers, A calls C<br>
and during the ringing phase A transfers C to B. During the transfer A<br>
sends an UPDATE message to C to signal that it should send RTP to B<br>
instead of A. B is notified using re-INVITE messages. B and C are both<br>
behind an OpenSIPS install built especially for this sceanrio.<br>
<br>
Most free software products have a hard time supporting this. Some<br>
have bugs (see the bugfix for OpenSIPS where we reported the initial<br>
problem) and others just don't support it at all. Since we're running<br>
quite a lot of those products I'd rather have something handle the<br>
UPDATE but don't bother other clients with it.<br>
<br>
So, is there a way for OpenSIPS to receive the UPDATE, handle the SDP<br>
change using either one of the media proxies and respond back to A<br>
with a 200 OK containing the (perhaps) newly updated SDP?<br>
<br>
We're already running an older installation of OpenSIPS that we can't<br>
(yet) upgrade so we're introducing a new one (I'm hoping temporarily).<br>
We found this issue while trying to connect our existing OpenSIPS<br>
based platform to our hosted PBX platform using a SIP connection.<br>
<br>
Thanks!<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
Andreas Sikkema<br>
<br>
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