[asterisk-users] Exit Dial Application

Danny Nicholas danny at debsinc.com
Wed Apr 15 08:09:13 CDT 2009


Here's how core show application dial says you should do it:
Change your dial to 
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)

This will execute the macro, then dial the number.  You will have to take
the hangups out of callback.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Christoph
Fürstaller
Sent: Tuesday, April 14, 2009 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application

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Hi,

Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?

I hope anyone knows if this is possible.

Chris...

Danny Nicholas schrieb:
> I'd change callback to this
> [callback]
> Exten => s,1,Playback(press5msg)
> Exten => s,n,Waitexten(5)
> Exten => s,n,Hangup
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
> 
> This will play a message, wait 5 seconds for user to press 5, then hangup
if
> they don't.
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Christoph
> Fuerstaller
> Sent: Tuesday, April 14, 2009 5:04 AM
> To: Asterisk Users Mailing List
> Subject: [asterisk-users] Exit Dial Application
> 
> Hi,
> 
> I' try to implement an automatic callback mechanism, just for local SIP
> calls.. Callback
> on busy and on no answer. If the other party doen't answer, it should be
> possible to press
> 5 to place an callback.
> 
> Here is my dial:
> exten => _X.,1,Set(EXITCONTEXT=callback)
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
> 
> And here the script for callback.
> [callback]
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
> 
> If I call someone and press 5, nothing happens. What could be a problem?
> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted
correctly,
> I can enter
> the voicmail menue.
> 
> I'm using Asterisk 1.4.21.1.
> 
> Any successions are very appreciated.
> 
> Chris...

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- --
commpany dialog solutions gmbh

Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications

Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512  Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstaller at commpany.at
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