<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
All generic parameters are still taken from sip.conf and you must set<br>
rtcachefriends=yes<br>
<br>
If you change anything in your mysql sip table you do not need to reload<br>
the modue, what you need to do is<br>
sip prune realtime <peername><br>
from the CLI<br>
<br>
As stated previously, you should never have to reload the sip module<br>
once realtime is working properly
<br></blockquote></div><br>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 ?<br>
<br>I read chapter below on <a href="http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip">http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip</a>. <br>1) Do anyone knows what exactly what delay is ?<br>2) It seems that you need to reload module in some cases or maybe I misunderstand what he want to say ?<br>
<br><h2>"Realtime Caching...
</h2>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.
<br>
<br>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. <br>
<br>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."<br><br><br>