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Olle E. Johansson wrote:<br>
<blockquote cite="mid42A84446.9080308@edvina.net" type="cite">
<pre wrap="">Christian Cayeux wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
I'm confused in the way Asterisk handles the transfer with SIP.
</pre>
</blockquote>
<pre wrap=""><!---->So am I, and I have been trying to fix it for a month... :-)</pre>
<blockquote type="cite">
<pre wrap="">Just say that A makes call to B, holds B, then makes call to C and make the
transfer.
At the end B is in call with C.
On a SIP point of view, A sends a refer to Asterisk, with a refer-to header
and replaces extension.
On receiving of this refer, Asterisk makes reinvite to B and reinvite to C
with right sdp, so that B speaks to C.
The problem is that on its display, B is still with A, and C is still with A
also.
Is there any way so that * handles refer in a different way?
</pre>
</blockquote>
<pre wrap=""><!---->...working on it, bit by bit. Stay tuned.
</pre>
</blockquote>
I have the strong impression there is a bug in the sip transfer when
the call commes from an queue, when a agent logs in on a sip terminal,
then gets an call from the queue and tries to tranfer that to an other
sip terminal you get oneway audio. with canreinvite=NO it works ok.<br>
<br>
Fredrik<br>
<blockquote cite="mid42A84446.9080308@edvina.net" type="cite">
<pre wrap="">
/Olle
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