Matt, can I assume from your silence that you concurr with my thinking that
realtime is in fact broken, or is it that I am using it incorrectly?
----- Original Message -----
From: "Rod Bacon" <rod.bacon@empoweredcomms.com.au>
To: "Matthew Boehm" <mboehm@cytelcom.com>; "Asterisk Users
Mailing List -
Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Sent: Wednesday, April 13, 2005 9:06 AM
Subject: [Asterisk-Users] Realtime Friends
> Matthew, I got the updates to start working again by ensuring that
> rtcachefriends=yes. I don't see why this should make a difference, but
> it does. My understanding was that this parameter only controlled the
> seeding of the in-memory friends list from the realtime db for purposes
> of MWI and KeepAlive.
>
> I have, however, one remaining issue that I need to resolve.
>
> Essentially, I am testing two Asterisk servers (Server1 ans Server2),
> configured to talk to a common database. I am trying to have calls
> placed on ANY server routed to SIP UAs registered on ANY OTHER server.
>
> Specifically;
>
> UA1 registers to Server1. DB is updated correctly. UA2 registers to
> Server2. DB is updated correctly. I can query the db (using REALTIME
> LOAD) from either server and see the correct SIP info for either UA.
>
> The central dialplan simply routes calls to SIP/UA1 or SIP/UA2.
>
> The problem is that Server1 ONLY knows about UA1 and Server2, UA2. The
> logic seems to be that the lookup in the extensions table (realtime
> dialplan) happens, then tries to route the call to a SIP registrant that
> is not in the local (in-memory) friends table.
>
> I thought the Server would then go back to the friends realtime table to
> get the registration info? Is this NOT how it is supposed to work?
>
> Should rtcachefriends force the server to update it's friends list on
> server startup, then at predetermined (configurable?) intervals?
>
>
>
> Matthew Boehm wrote:
>> (I removed the [] header cause that is what i base my email filter on.)
>>
>> Rod Bacon wrote:
>>
>>
>>>I think there's a more sinister bug in play somewhere. The
phones are
>>>on the same LAN. It was working when I only had a single asterisk
>>>server using the database, and seemed to stop when I added a second
>>>server. I know this doesn't make any sense...
>>
>>
>> OK. Lemme picture this. You had originally 1 asterisk server and 1
>> database server. This worked fine with RealTime. Then you added a
second
>> asterisk server to connect to this same database server and now the
phone
>> won't register with either asterisk server?
>>
>>
>>>The SIP registration MUST be ok, because the in-memory database on
the
>>>server that accepts the registration shows the correct
information...
>>>the problem is that it doesn't write it to the database.
>>
>>
>> Oh. Weird. But if you turn off the 2nd asterisk server, everything
is
>> fine?
>>
>>
>>>I think the bug must lie in the update code. When the registration
is
>>>accepted, the update command is sending nulls to the database for
some
>>>reason.
>>
>>
>> Yes, this is wierd cause I can't duplicate this. You don't
have
>> entries
>> in BOTH sip.conf AND ARA do you? You said the phone does indeed
register,
>> it
>> just doesn't update the database using RealTime?
>>
>> Is there any way you can send a full debug output starting slighty
>> before the phone tries to register? have you done a packet sniff to see
>> if
>> asterisk is indeed sending back a 200 OK to the register request?
>>
>> -Matthew
>>
>
> --
> =========================================> Rod Bacon - VOIP Systems
Engineer
> Empowered Communications
> Ground Floor, 102 York St. South Melbourne
> Victoria, Australia. 3205
> Phone: +613 99401600 Fax: +613 99401650
> =========================================
--------------------------------------------------------------------------------
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-user