Go ahead and clean up the logs and put them on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a>. <br>-MC<br><br><div class="gmail_quote">On Mon, Aug 27, 2012 at 10:03 PM, Emrah <span dir="ltr"><<a href="mailto:lists@kavun.ch" target="_blank">lists@kavun.ch</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
I am experiencing a strange issue with SIP based attended transfers.<br>
<br>
If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up.<br>
There is no much activity on the SIP side of things, it seems to be very much related to FS.<br>
<br>
Some info:<br>
I call out through a provider configured on the external profile, from a phone registered on the internal profile.<br>
It is not a codec conflict.<br>
Both lines are answered when I actually finalize the transfer.<br>
I tried with multiple phones and softphones.<br>
<br>
I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.<br>
<br>
Best,<br>
Emrah<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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