[asterisk-dev] Found a bug

[Atis]

On 9/15/07, Gregory Nietsky <gregnietsky at gmail.com> wrote:
>
> On 9/15/07, Nicholas Blasgen <nicholas at blasgen.com> wrote:
> > Atis,
> >
> > If you're using RT you can turn on RTCacheFriends and just issue a
> "asterisk
> > -rx sip reload" or it's something like that.  The other option is to use
> the
> > Asterisk Manager Interface (AMI) to issue a CLI "COMMAND" to reload SIP
> (or
> > whatever channel driver).  I think there is also another way to tell
> > RealTime to clear it's cache but I can't think of it off the top of my
> head.
> >  Oh, and by turning on RTCacheFriends you'll have access to "qualify=yes".
>
> Yes, i have read that, but it seems to be very realtime-unfriendly. I
> definitely won't use it on my production (i have a VPN there), but for
> development it seems to be too much effort. Btw, do you have some more
> info on RTCacheFriends - as i suppose i have to do it too often - i
> got a lot of users, that are changing often.. Would turning it on lead
> to that all users will be cached by asterisk and only reload will read
> them again?
>
> Regards,
> Atis
>
> Ive found a sip prune realtime peer XXXX useful when a peer is
> updated/changed via the web gui it fires off a prune and sip show peer XXXX
> load to get the new info active.

Yes, but that's not the realtime-way, i'm used to. Wouldn't it be
better for asterisk to cache only keep-alives and status (and maybe
even update availability status in db, the same way as it does with
regseconds), but re-read peer upon call - the same way as it's without
RTCacheFriends.

Btw, this is developers ML (however topic is quite user-oriented).
What do you guys think, of changing this behavior?

Regards,
Atis


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