Manuel Wenger wrote:> We are planning to deploy a pretty large asterisk server with many SIP
extensions (might be up to 10000 in the future), and I have a few questions:
> 1) is this possible, or are we running into some kind of limitation in the
software that I wasn't aware of and that I didn't find by browsing
through the archives and through Wiki? No, we don't need any G729-G711
transformations, it would only be acting as a SIP proxy (even if asterisk
isn't a proxy).
/Should/ be psosible with canreinvite=yes & no use of T,t in the dial
commands, so that Asterisk can stay out of the media path except when
absolutely necessary.
> 2) is there a way to store extensions.conf and/or sip.conf in some kind of
database, maybe MySQL? This would make life easier if someone wanted to change
his SIP password. Or how would you otherwise solve this problem?
http://voip-info.org/wiki-Asterisk+configuration+from+database
Option 1 is being enhanced through the development of ast_data.
I currently use Option 2
> 3) is there a quick way of reloading only a part of
sip.conf/extensions.conf, for example if only a user password changed, or an
extension's behaviour (eg. routing to voicemail instead of a SIP user)?
sip reload
extensions reload
That's as granular as it gets.
Should be harmless to keep doing this, though.
> Maybe I'm looking at the wrong software here and SER would be better
for what I want to do... I know asterisk is supposed to be a PBX replacement,
but the functions and flexibility it has really tells me "stick with
asterisk". Or am I way off with these assumptions?
Possibly - depends whether you're after a SIP proxy or a PBX ;)
F