[Asterisk-Dev] asterisk 'stable'?

[Woody Anderson]

Terry -

It sounds like you are a happy camper with RT sip. I assume from your
post you are using real realtime (not loading sip configuration from
the database). I took the latter approach thinking real realtime in
sip is not ready for primetime. I see the issues with voicemail MWI
and 'sip show peers' have  been fixed, but there seems to be other
issues with caching and consistency. With static realtime, I just do a
sip reload after updating the database and thus get near-realtime. But
then I don't have thousands of users like you. Have you had any of
these (or other) issues with sip rt?

I hear you about the realtime extensions. Not being able to create a
context in realtime is a big shortcoming in the design, and greatly
limits its usefulness. I wasn't aware of the redundant queries either.
Is anyone working on or looking at 'fixing' these issues with rt
extensions?

- Woody

Terry Wilson wrote:
Odd, I've been using realtime for thousands of users with an average
of many 
> calls per second using realtime sip. Now, I don't use realtime extensions, 
> which may be what you are talking about because I can't add contexts easily

> in realtime. Having to have a switch statement in a static file (which
means 
> a reload on adding a context) makes it non-realtime for our purposes. (And,

> last time I checked it still did the same query at least 3x in a row each 
> time as a hold-over from the IAX switch, gah!)
> 
> On 9/13/05, alex at pilosoft.com <alex at pilosoft.com> wrote:
> > 
> > On Tue, 13 Sep 2005, Sherwood McGowan wrote:
> > 
stuff deleted...