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 ;)
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