[Asterisk-Users] Fw: anybody has SIP realtime working ?

[Douglas Garstang]

Ok Andrew. Here's one for you... I just changed qualify from yes to no in the database... a 'sip show peers' still showed Asterisk as qualifying the users... I had to do a reload to get to accept the change to the database.


> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk at benshaw.com]
> Sent: Wednesday, March 22, 2006 10:46 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
> 
> 
> On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
> > First thing that comes to mind, what if we decided to 
> change a non user
> > setting in sip.conf?
> 
> You're reaching.  You said you NEED to reload all the time, 
> that this is a 
> MAJOR issue, a deal breaker.  So surely you must have 
> experienced this 
> downtime to be so sensitive to it.  What did you do on your 
> PRODUCTION system 
> that required constant reloads to cause the current behavior 
> to be such a big 
> problem?
> 
> Honestly; if you're changing a non-user setting in sip.conf 
> you're going to do 
> that very, very infrequently, and you'd do it during a low 
> volume time.
> 
> You said this is a major problem.  I'm calling you on it.  
> I'm interested in 
> making Asterisk robust and highly-available too, but I'm not 
> making up 
> scenarios in order to launch complaints and verbal assaults 
> against the 
> project in order to feed my inflated ego and try to get 
> things done "my way."
> 
> If you have a specific problem, let's hear it.
> 
> -A.
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