[Freeswitch-users] Attended transfer to a conference room
Emrah
lists at kavun.ch
Fri Sep 7 08:45:42 MSD 2012
I now realize that even blind transfers fail. There must be something wrong in my config. Will post the logs tomorrow.
On Sep 6, 2012, at 6:48 PM, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> 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
>>>>>>>> http://www.FreeSWITCH.org
>>>>>>>> http://www.ClueCon.com
>>>>>>>> http://www.OSTAG.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>
>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
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>>>>>>>>
>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>>>>>>>
>>>>>>>>
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>>>>>>>
>>>>>>>
>>>>>>>
>>>>
>> _________________________________________________________________________
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>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
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>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>>> 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/
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>>>>>
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>>>>> MSN:anthony_minessale at hotmail.com
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>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
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>>>>> 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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>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
>
>
>
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