[Freeswitch-users] Attended transfer to a conference room
Stanislav Sinyagin
ssinyagin at yahoo.com
Fri Sep 7 02:48:38 MSD 2012
for outgoing calls, this could be a different Lua script, or the same script that handles both modes:
so, the conference moderator calls the destination number with some unique special prefix, this call gets into the script, then script dials out and connects B leg with the moderator, and after that waits for moderator's input to bridge the called person into a conference.
also if I'm not mistaken, I've seen somewhere an extension to the strandard freeswitch conference which allows to dialout and bring new people into the room.
in any case, you can easily callout and bring people into the room from FreeSWITCH command line. This can also scripted in some way.
----- Original Message -----
> From: Emrah <lists at kavun.ch>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Sent: Friday, September 7, 2012 12:39 AM
> Subject: Re: [Freeswitch-users] Attended transfer to a conference room
>
>T hanks Stanislav, your idea is good, but works only for incoming calls. I know
> we could trick the idea around to have it work for outbound calls as well, but
> it sounds to me as being too much of a workaround for something so trivial.
>
> I'll play around some more and loop back my calls if I have to.
>
> Thanks for this and have a good night,
> Emrah
> On Sep 6, 2012, at 6:19 PM, Stanislav Sinyagin <ssinyagin at yahoo.com>
> wrote:
>
>> 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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>>>>>>>
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>>>>>>
>>>>>>
>>>
> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Official FreeSWITCH Sites
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>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>>
>>>>> 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
>>>>
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>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
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>>>>
>>>> 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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>>
>>
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>>
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>
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>
>
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>
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