[asterisk-users] mysql sip realtime
harry R
rhm.noa101 at gmail.com
Thu Aug 20 08:36:26 CDT 2009
>
> All generic parameters are still taken from sip.conf and you must set
> rtcachefriends=yes
>
> If you change anything in your mysql sip table you do not need to reload
> the modue, what you need to do is
> sip prune realtime <peername>
> from the CLI
>
> As stated previously, you should never have to reload the sip module
> once realtime is working properly
>
I try CLI command sip prune realtime <peer name> and my peer infos was
perfectly updated when I do sip show <peer name> but have you any idea of
how I can do that automatically ?
I read chapter below on
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip.
1) Do anyone knows what exactly what delay is ?
2) It seems that you need to reload module in some cases or maybe I
misunderstand what he want to say ?
"Realtime Caching... As of CVS-HEAD 3/16/05, if you enable RealTime caching
in your sip.conf, Voicemail MWI works and so does 'sip show peers'. To do
so, add "rtcachefriends=yes" to the general section of your sip.conf file.
As the name implies, this caches the "RealTime" information from the
database. As a result, there is a delay in updating some (if not all) fields
in the SIP entry when you update the database. For instance, if you create
an entry with a context = "context1" and Asterisk loads it from the database
(perhaps the phone registered or tried to make a call), Asterisk holds on to
that information as far as I can tell, indefinitely until a sip reload
occurs.
This also means that you will have to do a sip reload to clear out any
entries. Removing them from the database does not seem to work. You can
still add new entries though without reloading Asterisk."
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