[asterisk-users] sip realtime
Benchev
bbench at mail.bg
Wed Jul 26 06:02:42 MST 2006
On Wednesday 26 July 2006 00:03, marek cervenka wrote:
> i'm reading a lot docs about asterisk realtime
> but i cannot understand how works sip realtime static
>
> i need NAT/qualify for SIP. this is not possible with dynamic realtime
> i want
> - save data to sql
> - asterisk -rx "reload" to read config (sip.conf with sip users) from sql
>
> it is possible?
For sip realtime static you have probably read:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static
However, NAT/qualify for SIP(users) is perfectly possible with
"dynamic" realtime:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
And `sip_buddies` table gives extensive opportunities(including nat and
qualify).
And there is the advantage of using it(realtime), you do not need to reload
when a new user comes. (This is valid for the needed extensions and
voicemail attributes, as well)
Sorry for twisting a bit your question, but basically "realtime static" means
to store a .conf file into a database(in which case you must delete
its equivalent from /etc/asterisk); "realtime" is when you store
users, peers and friends into the database, keeping the skeletons of
sip&iax2.conf files in /etc/astrisk. In that case the "users, peers and
friends" sip or iax2 info is being read "on the fly". The appropriate
extensions though, must be
addressed with "switch => Realtime " statement from extensions.conf.
Since all .conf files exist they have precedence. Also register=> can be done
only from a .conf file.
Hope it helps.
Benchev
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