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Olle E. Johansson wrote:
<blockquote cite="mid42A92AAB.20901@edvina.net" type="cite">
<pre wrap="">Fredrik Chabot wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Olle E. Johansson wrote:
</pre>
<blockquote 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>
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<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>
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<pre wrap="">...working on it, bit by bit. Stay tuned.
</pre>
</blockquote>
<pre wrap="">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.
</pre>
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<pre wrap=""><!---->As always, I would like to see a SIP debug with debug=4 and verbose=4
</pre>
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Ok, full became quite large so i send it directly to you and not on the
list hope thats ok,<br>
<br>
<blockquote cite="mid42A92AAB.20901@edvina.net" type="cite">
<pre wrap="">
/O :-)
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