[Freeswitch-users] Attended transfer to a conference room

Fri Sep 7 02:19:10 MSD 2012

Emrah, I think this particular use case should be quite easy to program in a Lua script.
The incoming call is handed to the script, the script originates a call to the operator, 
bridges them, then waits for operator's input, then hangs up the operator and bridges the A leg into the conference.

on the other side, why would anyone need a human operator in front of a conference bridge? 4 or 5 digits PIN usually works fine.

----- Original Message -----
> From: Emrah <lists at kavun.ch>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Thursday, September 6, 2012 5:57 PM
> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> 
> 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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>>>>> 
>>>>>  --
>>>>>  Michael S Collins
>>>>>  Twitter: @mercutioviz
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>>>> 
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>>> 
>>> 
>>>  --
>>>  Anthony Minessale II
>>> 
>>>  FreeSWITCH http://www.freeswitch.org/
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>>> 
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>> 
>> 
>> 
>>  -- 
>>  Anthony Minessale II
>> 
>>  FreeSWITCH http://www.freeswitch.org/
>>  ClueCon http://www.cluecon.com/
>>  Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>>  AIM: anthm
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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
>> 
>>  _________________________________________________________________________
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>>  http://www.freeswitchsolutions.com
>> 
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> 
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