[SR-Users] Realtime integration: Unregistered clients showing as registered?

[Olli Heiskanen]

Hello,

Gracias Pedro, kiitos Mikko.

It's good to know I have configured Kamailio correctly. I added the type
into my table but so far no luck having asterisk see the clients
registered, at least on cli. I do see that asterisk adds registration data
into the table. I'll work on this for a bit and ask in the asterisk list on
more tricks on asterisk side. I'll post back here if I find out what the
problem was, in case someone is having similar issues.

Thanks again,
Olli



2014-04-22 21:06 GMT+03:00 Pedro Niño <nino.pedro at gmail.com>:

> Don't forget to include peer type (friend), and The callbacknumber In The
> table.
>
> It happened to me and asterisk/kamailio behavior was wayyy to weird  until
> made sure both parameters were there.
>
> -----
>
> In this setup I have SIP peers in an asterisk table added like this:
>
> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
> testers.com');
>
> ------
>  El abr 19, 2014 1:17 PM, "Olli Heiskanen" <ohjelmistoarkkitehti at gmail.com>
> escribió:
>
>>
>> Hello,
>>
>> One of the tests I've been working with is Asterisk realtime integration
>> according to Daniel's guide here:
>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>
>> Weird thing is the client looks registered but I'm not sure if it really
>> is registered. If I'm not mistaken I should see the peers when I issue 'sip
>> show peers' on asterisk cli. Instead I get this:
>>
>> *CLI> sip show peers
>> Name/username      Host      Dyn Forcerport Comedia      ACL Port
>>  Status      Description      Realtime
>> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
>> offline]
>>
>>
>> Also, calling between clients will fail; in Asterisk cli I get:
>> *CLI>
>>   == Using SIP RTP TOS bits 184
>>   == Using SIP RTP CoS mark 5
>>     -- Executing [661 at default:1] NoOp("SIP/660-00000000", "Testing:
>> Dialed 661") in new stack
>>     -- Executing [661 at default:2] Dial("SIP/660-00000000",
>> "SIP/661,3600,rt") in new stack
>>   == Using SIP RTP TOS bits 184
>>   == Using SIP RTP CoS mark 5
>>     -- Called SIP/661
>>   == Everyone is busy/congested at this time (1:0/0/1)
>>     -- Executing [661 at default:3] Hangup("SIP/660-00000000", "") in new
>> stack
>>   == Spawn extension (default, 661, 3) exited non-zero on
>> 'SIP/660-00000000'
>>
>>
>> In this setup I have SIP peers in an asterisk table added like this:
>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
>> testers.com');
>>
>> I have Kamailio and Asterisk on the same machine where Kamailio listens
>> port 5060 and Asterisk listens 5070. Things that differ from the guide are
>> Kamailio and Asterisk versions, which in my case are newer. Also, for
>> another testing case I have MULTIDOMAIN enabled in Kamailio, does this
>> interfere with the realtime integration? I'm using only one domain though.
>>
>> Please let me know if any configs or traces I can provide will help
>> figure out what's going on.
>>
>> cheers,
>> Olli
>>
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>> sr-users at lists.sip-router.org
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>>
>>
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