[Asterisk-Users] Busy recognition

[Douglas Garstang]

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: a0305292 at unet.univie.ac.at [mailto:a0305292 at unet.univie.ac.at] 
	Sent: Wed 12/7/2005 7:40 AM 
	To: asterisk-users at lists.digium.com 
	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 <asterisk108 at gmail.com> 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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