Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your
sip.conf (asterisk) to show the realtime peers
El abr 23, 2014 8:29 AM, "Olli Heiskanen" <ohjelmistoarkkite...@gmail.com>
escribió:
> 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.pe...@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" <
>> ohjelmistoarkkite...@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@default:1] NoOp("SIP/660-00000000", "Testing:
>>> Dialed 661") in new stack
>>> -- Executing [661@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@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
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org
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>>>
>>>
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>
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