[asterisk-users] SIP Qualify

[Douglas Garstang]

Yes, it might be a problem in our situation. We have three Asterisk boxes in a 'cluster'. The sip.conf is identical on all three. In that case, all three of the Asterisk boxes in our cluster are going to send sip options messages to the phones, which is silly. 
 
Only the Asterisk box that a phone is registered on needs to send the sip notify messages. The rest are a waste. I'm not sure how we'd work around this.
 
We may just have to make do with the caller of an unavailable phone getting ringback until the dial timeout occurs.
 
Doug.

	-----Original Message----- 
	From: Alexander Lopez [mailto:Alex.Lopez at OpSys.com] 
	Sent: Mon 8/14/2006 9:46 PM 
	To: Asterisk Users Mailing List - Non-Commercial Discussion 
	Cc: 
	Subject: RE: [asterisk-users] SIP Qualify
	
	

	Qualify does what the name implies "qualifies the connection' It pools
	every 60s but it calculates he time it took for the packet to reach the
	end device. If the endpoint has a latentcy > than the qualify parameter,
	* considers the endpoint unreachable.  This does not however address the
	point you made in another post about RINGING before the INVITE. It is
	still possible to have a phone go dead in the 60sec between qualify
	re-checks.
	
	There are several post in history about qualify and it sending LARGE
	amounts of traffic to endpoints. I think it was John Todd that was the
	OP on the subject IIRC.
	
	
	
	SNIP
	
	
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