[asterisk-users] How to do not use Asterisk internal DB for SIP register?

Mindaugas Kezys mkezys at gmail.com
Mon May 26 11:01:26 CDT 2008


I have fullcontact field in DB - it's empty. It's only filled when
rtcacfriends = yes. Same on 3 servers we tested.

Thank you for good idea about sip prune.

In system with several Asterisk servers it should be done over AMI I guess,
or is here better way to do this?

Best wishes from Lithuania!

Regards,
Mindaugas Kezys
http://www.kolmisoft.com


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Atis Lezdins
> Sent: Monday, May 26, 2008 4:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to do not use Asterisk internal DB
> for SIP register?
> 
> On Mon, May 26, 2008 at 4:02 PM, Mindaugas Kezys <mkezys at gmail.com>
> wrote:
> > That's what I'm talking about.
> >
> > Asterisk 1.4.18.1 and 1.4.20.1 are tested on 3 different servers from
> clean
> > install (from sources). (1.4.19 does not work with SIP Realtime at
> all)
> >
> > Realtime cashing is OFF (sip.conf rtcachefriends = no). But I still
> can see
> > (after device registration):
> >
> > SIP/Registry/106      :
> > 193.138.yyy.xxx:62501:1800:106:sip:106 at 193.138.yyy.xxx:5060
> >
> > with "database show" command
> >
> > "sip show peers" shows nothing
> >
> > and
> >
> > "fullcontact" in DB is empty.
> >
> > (BTW - I cleaned DB with "database deltree SIP/Registry" before
> > registering.)
> >
> > What's happening? New bug?
> 
> I checked this in sources, and seems that having "fullcontact" in
> realtime table should do the trick and write to realtime engine
> instead of Berkeley. However my production server also have
> "SIP/Registry" entries filled in, but I have fullcontact in RT
> populated too.I have "rtcachefriends" enabled, as realtime SIP peers
> aren't really identical to static without cache - no call limit, state
> in queues, and lot of other troubles.
> 
> Btw, rtcachefriends isn't that bad, you just have to issue "sip prune
> realtime peer XXX" after each update in database.
> 
> Regards,
> Atis
> 
> 
> 
> >
> > Regards,
> > Mindaugas Kezys
> > http://www.kolmisoft.com
> >
> >
> >> -----Original Message-----
> >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
> users-
> >> bounces at lists.digium.com] On Behalf Of Grey Man
> >> Sent: Monday, May 26, 2008 3:28 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] How to do not use Asterisk internal DB
> >> for SIP register?
> >>
> >> On Mon, May 26, 2008 at 12:55 PM, Mindaugas Kezys <mkezys at gmail.com>
> >> wrote:
> >> >
> >> > To illustrate my question take small client with 1 Asterisk
> server.
> >> >
> >> > No OpenSER, no other SIP proxy, just plain Asterisk which uses
> >> Realtime and
> >> > MySQL.
> >> >
> >> > That's very common setup and you tell: "you should be fully
> reliant
> >> on the
> >> > external db and not use the Asterisk internal db at all"
> >>
> >> No in that case I would not say use an external SIP Registrar, it's
> >> not worth the effort for a small set up and Asterisk should be able
> to
> >> cope. If you turn offall the realtime caching settings that will
> stop
> >> Asterisk using the internal db and result in it relying exclusively
> on
> >> the external realtime one.
> >>
> >> > So I want to know, how to do that? How to turn off internal
> Asterisk
> >> DB, and
> >> > how to tell Asterisk to update "fullcontact" field in DB when user
> >> > registers?
> >>
> >> You shouldn't have to do anything except configure realtime and turn
> >> off the realtime caching settings in sip.conf.
> >>
> >> Regards,
> >>
> >> Greyman.
> >>
> >> _______________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -
> -
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> --
> Atis Lezdins,
> VoIP Project Manager / Developer,
> atis at iq-labs.net
> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Cell Phone: +1 800 7300689
> Work phone: +1 800 7502835
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list