[Freeswitch-users] Attended transfer to a conference room

[Michael Collins]

Go ahead and clean up the logs and put them on pastebin.freeswitch.org.
-MC

On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists at kavun.ch> wrote:

> Hi all,
>
> I am experiencing a strange issue with SIP based attended transfers.
>
> 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.
> There is no much activity on the SIP side of things, it seems to be very
> much related to FS.
>
> Some info:
> I call out through a provider configured on the external profile, from a
> phone registered on the internal profile.
> It is not a codec conflict.
> Both lines are answered when I actually finalize the transfer.
> I tried with multiple phones and softphones.
>
> I can clean up my logs and post them here, but if you guys have some info
> already it would be much appreciated.
>
> Best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120828/6ff13f6c/attachment.html