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

Mon May 26 12:01:25 CDT 2008

On Mon, May 26, 2008 at 7:01 PM, Mindaugas Kezys <mkezys at gmail.com> wrote:
> 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?

Yes, manager action "Command: " does the trick. Works reliably
(supposing you prune only when changing something).

If you are willing to follow this up with feedback - i suggest that
you open a bug.

Greetings from Latvia :)
Atis

>
> 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.
>> >>
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>>
>>
>> --
>> 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
>>
>> _______________________________________________
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>
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-- 
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