> > 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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