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

[mustardman29]

I might as well jump in.  I am not clear on what the problem is but whether
it's a problem on something that needs to be done frequently or infrequently
or perhaps can be avoided with little effort, it's still a problem.  

Your argument is more like the classic "it's not a bug, it's a feature". 

> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk at benshaw.com] 
> Sent: Wednesday, March 22, 2006 9:46 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.
> 
>