[Freeswitch-users] Attended transfer to a conference room

[Anthony Minessale]

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 at 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 at 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 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
>> _________________________________________________________________________
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>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://www.cluecon.com
>
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-- 
Anthony Minessale II

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