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[asterisk-users] Realtime integration: Unregistered clients showing as registered?


 
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ohjelmistoarkkitehti a...
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PostPosted: Thu Apr 24, 2014 3:27 am    Post subject: [asterisk-users] Realtime integration: Unregistered clients Reply with quote

Hello all,


I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation.


My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. 


In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the user data. Field regseconds has a value and fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as they are on the same machine).


I have a very simple dialplan:


[general]


[default]
exten => _XXX,1,NoOp(general : Dialed ${EXTEN})
 same => n,Dial(SIP/${EXTEN},3600,rt)
 same => n,Hangup





Here's more on my problem and background to it, guys on the Kamailio list helped out but looks like I need to check my Asterisk configuration. 
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html


My goal is to have all clients in the asterisk database, asterisk (one at this point, several later) handling the calls and Kamailio as proxy. In Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one domain 'testers.com'. 


I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on the same rental virtual server. Clients are in my home network behind nat.
In MySQL I have database asterisk with table sippeers, where I have clients added like this: 
INSERT INTO sippeers (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type) VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com','660','friend');


In this message there are some outputs and a sip trace of a register:
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html


What I don't know is how to configure sip.conf, so far I've just been making guesses based on online examples and documentation. 
My current sip.conf looks like this:


[general]
bindport = 5070
bindaddr = 127.0.0.1
tcpbindaddr = 127.0.0.1:5070
tcpenable = no
limitonpeers = yes
;rtcachefriends = yes
tos_sip=cs3
tos_audio=ef
realm = testers.com


I've tried defining realm and domain values, but I lack proper understanding of those. Can you guys help me out? Are there any other configurations I need to check? 


Respectfully,
Olli
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