you are missing some of the sip and make sure its GIT HEAD (this should be on jira) sofia global siptrace on On Tue, Sep 4, 2012 at 9:35 AM, Emrah <lists@kavun.ch> wrote:
Hi MC,
Thanks a bunch for your reply, sorry for the delay. Here are my logs: http://pastebin.freeswitch.org/19831
Any idea would be greatly appreciated.
Best, Emrah
On Aug 28, 2012, at 10:56 AM, Michael Collins <msc@freeswitch.org> wrote:
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@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@freeswitch.org http://www.freeswitchsolutions.com
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