[asterisk-users] Exit Dial Application

Atis Lezdins atis at iq-labs.net
Tue Apr 14 13:08:29 CDT 2009


CLI> core show application Dial

    d    - Allow the calling user to dial a 1 digit extension while waiting for
           a call to be answered. Exit to that extension if it exists in the
           current context, or the context defined in the EXITCONTEXT variable,
           if it exists.

Regards,
Atis

On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
<fuch_lists at kurtkrenn.com> wrote:
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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
>
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> IP-Communications
>
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-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835



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