[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
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>>>>>>>> 
>>>>>>>> 
>>>> 
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>>>>>> 
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>>>>>> Anthony Minessale II
>>>>>> 
>>>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
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> 
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