On Tue, 2010-09-21 at 19:04 -0400, Dan Journo wrote:
> I checked the bug reports and all I could find was similar issues with the
> Asterisk 1.6 which (according to the reports) have been resolved.
> I couldnt find anyone talking about 1.4 so I created a new issue and someone
> closed the case and added this note:-
>
> > This does not appear to be a bug, but rather a support issue. Please use
> > the asterisk-users mailing list for such issues.
> > The problem looks like your device has not re-registered after your 'sip
> > reload' which means it does not exist in memory, and thus causes Asterisk
> > to
> > not know where to send the call. Your device needs to re-register after a
> > 'sip reload' in order for Asterisk to know where to send the call.
>
> I really think that "sip reload" shouldn't purge all the realtime peer
> registrations. It should treat the realtime peers the same way as the
> hardcoded peers. As i've said, the hardcoded peers don't lose registration
> when I issue a SIP RELOAD.
> Asterisk should be flexible enough to allow modification of the sip.conf file
> without losing all the realtime registrations.
>
> Does anyone have a comment on the subject? Am I expecting too much?
> I'm open to feedback.
>
I use realtime on 1.4 and 1.6 servers but always with
rtcachefriends=yes in sip.conf so I can use things like sip show peers.
My experience is that when I issue a sip reload all registrations
disappear but the moment a call comes in for a peer it uses the last IP
where it was registered to send the call. I guess this is the
equivalent do doing "sip show peer XXX load". The peer does not need to
register again to get calls.
I think this works because the internal database has the last IP of the
peer even after a sip reload. If you do a "database show" you will see
something like:
/SIP/Registry/XXXX
192.168.2.215:5060:3600:XXXX:sip:[email protected]:5060;transport=udp
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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