[Freeswitch-users] Attended transfer to a conference room

[Emrah]

Hi Anthony,

The reason for the attended transfer to the conference call is to announce "manually" a new participant before releasing the transfer. E.g.: in the scenario where you have an operator transferring calling parties in a conference.
Which of the two alternatives would be the most CPU efficient? A loopback or a SIP call to the same domain? I wish I could give a hand in the code, but my 2 cents would be to optimize this at some point in your roadmap.

Best,
Emrah
On Sep 4, 2012, at 1:17 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Oh,
> 
> And you should try to avoid doing an attended transfer to
> one-legged-calls like conference or ivr, blind transfers work better
> for this because calls to apps are not bridged and the concept of
> transferring becomes confusing.  The other alternative is to bridge to
> the conference by looping the call over loopback or calling to the
> same box on sip so there is a true bridge.  But blind transferring is
> the best solution.
> 
> 
> On Tue, Sep 4, 2012 at 12:15 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> 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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>>>> 
>>>> 
>>>> 
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>>>> 
>>>> 
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
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>>>> 
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com
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>>>> 
>>>> 
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>>> 
>>> _________________________________________________________________________
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>>> 
>>> 
>>> 
>>> 
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>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
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> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
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> 
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