[asterisk-users] SIP Qualify

[Watkins, Bradley]

Only the Asterisk box that a phone is registered on WILL send the sip
notify messages.  The others will have no idea where to send them, and
will not do so.
 
- Brad

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Tuesday, August 15, 2006 12:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SIP Qualify

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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