[asterisk-users] mysql sip realtime

[Ishfaq Malik]

harry R wrote:
>
>     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 ?
How are you updating your sip table? Are you doing it manually or have 
you built an interface for it? If you have built an interface you can do 
the sip prune realtime by using the AMI 
(http://www.voip-info.org/wiki/view/Asterisk+manager+API), that's what 
we do here. If you're updating the table manually then what differerence 
does it really make to do the realtime pruning 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 ?
The only time you'll have to do a sip reload is if your changing any 
global setting. On the whole you do not want to be doing sip reloads as 
it's clear your sip realtime cache
>
>
>     "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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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

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