I've found that I can only get ${DIALSTATUS} to work reliably if I have a
qualify= directive in sip.conf against the user. If you don't put the qualify=
directive in there, Asterisk will just ring and ring until the ring timeout
occurs. If you do put it in there, and Asterisk has had a chance to send
OPTIONS to the peer/user then it will report the result of the Dial (ie no
connect, 404 Not Found) in ${DIALSTATUS}.
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wed 12/7/2005 7:40 AM
To: [email protected]
Cc:
Subject: Re: [Asterisk-Users] Busy recognition
try with ${DIALSTATUS} ->
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
you could start an agi(which receives the dialstatus-variable) and let
this agi write to your webapp-db...
regarding recall you should have a look at callfiles ->
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
hth
christian
p.s.: this is not a
On Wed, 7 Dec 2005 13:41:09 +0100
Ronald <[EMAIL PROTECTED]> wrote:
> Hi
>
> I'm not sure if I can ask it here or on Dev. But the following
>
> I'm trying to make a webinterface for calling/hanging up/transfer.
> Thats all going very well.
>
> But I'm wondering about the following. I'm trying to catch a busy
> call from the CLI or the manager.
> If I call someone and it takes to long I see busy passing by
> And I want to pick that up and put that on a webpage. and give the
> option to redial/call another or hangup
>
> Can anyone help me with that
>
> Thanks
>
> Ronald
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