[Freeswitch-users] Is it possible to use att_xfer on channels already bridged via loopback?

[Anthony Minessale]

Guess who wrote that code in res_features ;)


On Mon, Jan 24, 2011 at 12:51 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Yes, I'm using SIP as my endpoints.
>
> There are number of reasons why I want to use in-call triggered transfers.
>
> 1. We are migrating several large ISP PBX to Freeswitch from Asterisk.
> In-call transfers in Asterisk via res_features are mature and stable,
> and many users use it - if we remove this feature, we will loose our
> users.
>
> 2. Not all SIP endpoints support call transfer, and despite majority
> of devices support this, we want fine control over who is allowed to
> do transfer in which situation - there are many paths of call in our
> PBXs.
>
> 3. We also need to provide call transfer functionality to external
> PSTN users, which are dialed via SIP proxy and then SIP-E1 gateway
> like audiocodes or cisco. There is no simple universal solution to
> trigger such transfers from PSTN side besides in-call transfers with
> bind_meta_app.
>
> I agree, in ideal world we would live without server-side attended
> transfers, but for now att_xfer is an ultimate feature of real PBX,
> even old Avaya and Samsung boxes support this :)
>
>
> 2011/1/24 João Mesquita <jmesquita at freeswitch.org>:
>> I am just now discussing this with another developer and the question that
>> is never answered is:
>> Why are you trying to use att_xfer if it is your endpoint's duty to make the
>> transfer? Are you using SIP?
>> Regards,
>> João Mesquita
>>
>>
>> On Mon, Jan 24, 2011 at 1:01 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>>
>>> Is att_xfer or mod_loopback is broken in FS-current?
>>> I use FreeSWITCH Version 1.0.head (git-7eceff4 2011-01-16 22-33-50 +0000)
>>> Seems there were no updates of att_xfer or mod_loopback since that.
>>>
>>> I use loopback channel as destination when doing att_xfer to re-enter
>>> dialplan.
>>> With loopback_bowout=false and loopback_bowout_on_execute=false this
>>> works. But when any of connected parties tries to do att_xfer again,
>>> all channels get hangup on transferer hangup.
>>>
>>> Scenario:
>>>
>>> A calls B, B answers
>>> A launches att_xfer via *7, B listens to MOH
>>> A dials C and we do att_xfer to loopback/C
>>> C answers, A hangs up to complete transfer
>>> C and B are now bridged via loopback, `show channels` shows 4 channels
>>> include 2 loopback legs.
>>>
>>> Now, C also tries to do in-call transfer with *7.
>>> C launches att_xfer via *7, B listens to MOH
>>> C dials D and do att_xfer to loopback/D
>>> D answers, C hangs up to complete transfer
>>> B and D are hung up instead of be bridged together.
>>>
>>> There are also issues with MOH wile running att_xfer, but they are not
>>> so important as att_xfer behavior itself.
>>>
>>>
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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