[Freeswitch-users] Attended transfer to a conference room

Fri Sep 7 02:32:09 MSD 2012

after a second thought, I realized that you probably want to minimize your own DTMF dialing effort after the call is set up. This can be done easier:  you can build additional extensions which only work for your SIP account and connect into the conference rooms without PIN. This can be done (again) with a Lua script, or probably by manipulating the contexts. 

----- Original Message -----
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Sent: Friday, September 7, 2012 12:19 AM
> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
> 
> 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
>>>>>> 
>>  _________________________________________________________________________
>>>>>>   Professional FreeSWITCH Consulting Services:
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>>>>>>   http://www.freeswitchsolutions.com
>>>>>> 
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> Server
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>>>>>> 
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>>>>>> 
>>>>>> 
>>>>>>   --
>>>>>>   Michael S Collins
>>>>>>   Twitter: @mercutioviz
>>>>>>   http://www.FreeSWITCH.org
>>>>>>   http://www.ClueCon.com
>>>>>>   http://www.OSTAG.org
>>>>>> 
>>>>>> 
>>>>>> 
>>  _________________________________________________________________________
>>>>>>   Professional FreeSWITCH Consulting Services:
>>>>>>   consulting at freeswitch.org
>>>>>>   http://www.freeswitchsolutions.com
>>>>>> 
>>>>>>   FreeSWITCH-powered IP PBX: The CudaTel Communication 
> Server
>>>>>>   
>>>>>> 
>>>>>>   Official FreeSWITCH Sites
>>>>>>   http://www.freeswitch.org
>>>>>>   http://wiki.freeswitch.org
>>>>>>   http://www.cluecon.com
>>>>>> 
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>>>>>> 
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>>>>> 
>>>>> 
>>>>> 
>>  _________________________________________________________________________
>>>>>   Professional FreeSWITCH Consulting Services:
>>>>>   consulting at freeswitch.org
>>>>>   http://www.freeswitchsolutions.com
>>>>> 
>>>>>   
>>>>>   
>>>>> 
>>>>>   Official FreeSWITCH Sites
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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
>>>>   MSN:anthony_minessale at hotmail.com
>>>>   GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>   IRC: irc.freenode.net #freeswitch
>>>> 
>>>>   FreeSWITCH Developer Conference
>>>>   sip:888 at conference.freeswitch.org
>>>>   googletalk:conf+888 at conference.freeswitch.org
>>>>   pstn:+19193869900
>>> 
>>> 
>>> 
>>>   -- 
>>>   Anthony Minessale II
>>> 
>>>   FreeSWITCH http://www.freeswitch.org/
>>>   ClueCon http://www.cluecon.com/
>>>   Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>>   AIM: anthm
>>>   MSN:anthony_minessale at hotmail.com
>>>   GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>   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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>>> 
>>>   
>>>   
>>> 
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> 
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