Hello,
I'm new to asterisk and I'm really amaced of this software.
I've installed asterisk to take calls via SIP and Capi, all calls get
forwarded to a specific phone. Now I've to transfer some calls to other
phones. That workes already, but only this way, that I can transfer the
call without contacting the person on the new phone, for example to tell
him about the person that waits on the other line. So what I like to do
is the following:
1. A call comes in, either via SIP or Capi.
2. I take the call.
3. When I need to forward the call to an internal or an external number,
I'd like to park the call from outside and call the new phone. I'd
would be cool to play some music for the person who has to wait.
4. A person on the new phne hopefuly answers my call, I tell something
about the caller from outside, and after i hang up the waiting caller
from outside gets transfered automaticly to the new phone.
Is this possible with asterisk? If yes, how do I need to configure it
via the extensions.conf? Or is it not a problem of asterisk but a
problem of my IP phones (we use some Grandstreams and a Snom).
Here is my actual extensions.conf. Is this file OK in general or do I
have to change something (remember that I'm still a newbie :)). Can I
make some thing better? How can I setup the transfer stuff? Can anyone
give me an example or point me in the right direction please?
-----
[general]
static=yes
writeprotect=no
;[globals]
;CONSOLE => Console/dsp
;IAXINFO => guest
;TRUNK => Zap/g2
;TRUNKMSD => 1
[local]
exten => 300,1,SetLanguage(de)
exten => 300,2,VoicemailMain()
exten => 300,3,Hangup()
exten => 301,1,Ringing()
exten => 301,2,Dial(SIP/301,30,t)
exten => 301,3,Congestion()
exten => 301,4,Busy()
exten => 301,5,Hangup()
exten => 302,1,Ringing()
exten => 302,2,Dial(SIP/302,30,t)
exten => 302,3,Congestion()
exten => 302,4,Busy()
exten => 302,5,Hangup()
exten => 303,1,Ringing()
exten => 303,2,Dial(SIP/303,30,t)
exten => 303,3,Congestion()
exten => 303,4,Busy()
exten => 303,5,Hangup()
[sip-in]
exten => anruf,1,Setlanguage(de)
exten => anruf,2,Ringing()
exten => anruf,3,Dial(SIP/302,30,t)
exten => anruf,4,Congestion()
;exten => anruf,3,Voicemail(302)
exten => anruf,5,Busy()
exten => anruf,6,Hangup()
exten => xyz,1,Goto(anruf,1)
[capi-in]
exten => isdn,1,Setlanguage(de)
exten => isdn,2,Ringing()
exten => isdn,3,Dial(SIP/302,30,t)
;exten => isdn,4,Voicemail(302)
exten => zyx,1,Goto(isdn,1)
[dialout]
include => local
exten => _0.,1,SetCallerID(zyx)
exten => _0.,2,Dial(CAPI/contr1/${EXTEN:1})
exten => _0.,3,Congestion()
exten => _0.,4,Busy()
exten => _0.,5,Hangup()
exten => _1.,1,SetCallerPres(some testing stuff)
exten => _1.,2,Dial(CAPI/contr1/${EXTEN:1})
exten => _1.,3,Congestion()
exten => _1.,4,Busy()
exten => _1.,5,Hangup()
exten => _2.,1,SetCallerID(zyx)
exten => _2.,2,Dial(SIP/${EXTEN:1}@sip-in,60,tTr)
exten => _2.,3,Congestion()
exten => _2.,4,Busy()
exten => _2.,5,Hangup()
-----
Best regards,
Schoeppi
--
Christian Schoepplein <chris at schoeppi.net>
Manage your communication: http://www.otrs.com
Linux for the blind: http://www.blinux.suse.de
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