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[Freeswitch-users] att_xfer+loopback

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anthony.minessale at g...
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PostPosted: Wed Nov 12, 2008 12:12 pm    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

so you are answering the other call
then pressing 0 to conference them together
and then hanging up?


2008/11/12 x y <fs_ask_sy@citromail.hu (fs_ask_sy@citromail.hu)>
Quote:
I've sent both of them.




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Anthony Minessale II

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fs_ask_sy at citromail.hu
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PostPosted: Thu Nov 13, 2008 3:44 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

The dtmf 0 works fine with sofia/ and user/, but with loopback/ I dont even have time to make a conference like transfer, because the caller A hangs up as soon as C asnwers the call of B, instead of being hold (in a A-call->B-att_xfer->C situation), and there will be nobody to transfer to. I'm trying to do that, the two calls is bridged together when B hangs up, or A gets B back when C hangs up, depends on the situation.
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anthony.minessale at g...
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PostPosted: Thu Nov 13, 2008 9:29 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

loopback may not be supported in every situation.
You do know transfer etc is built into sip already right?
you just press hold on your sip phone, call someone else and press transfer.


On Thu, Nov 13, 2008 at 2:36 AM, x y <fs_ask_sy@citromail.hu (fs_ask_sy@citromail.hu)> wrote:
Quote:
The dtmf 0 works fine with sofia/ and user/, but with loopback/ I dont even have time to make a conference like transfer, because the caller A hangs up as soon as C asnwers the call of B, instead of being hold (in a A-call->B-att_xfer->C situation), and there will be nobody to transfer to. I'm trying to do that, the two calls is bridged together when B hangs up, or A gets B back when C hangs up, depends on the situation.



Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
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http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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fs_ask_sy at citromail.hu
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PostPosted: Thu Nov 13, 2008 11:55 am    Post subject: [Freeswitch-users] att_xfer+loopback Reply with quote

Thx for the advices.
Unfortunately, using transfer with hold wont work for me. For example, with a sip phone, wich dont have built in support for transfer.
The whole situaion is that I'm using softphone, wich has a client freeswitch running in the background, and there is another freeswitch on the server, witch is responsible for the main part. I would like to process the att_xfer procedure in the client freeswitch, then just send the numbers to the server's dialplan, and let to the server's freeswitch decide what to do with it.
I tried the regular transfer, and it worked fine. With att_xfer, the problem is that the client's freeswitch doesnt know where will be the call transfered, it just knows the dialnumber of it. Thats why I have tried to use loopback with att_xfer.
Is there another way to that?
Would it worth to create a similar att_xfer application, wich accepts dialnumbers like transfer? I checked the source both of transfer and att_xfer, hoping they are similar, but they werent for the first look, and I dont really know how much work would it cost.
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