[SR-Users] Σχετ: Problem with Realtime Kamailio-Asterisk integration

[aaaa aaaa]

Barry,
It solved with your guidance.
Thanx!


________________________________
 Απο: Barry Flanagan <barry at flanagan.ie>
Προς: aaaa aaaa <indefix1 at yahoo.gr>; Kamailio (SER) - Users Mailing List <sr-users at lists.sip-router.org> 
Στάλθηκε: 10:35 μ.μ. Κυριακή, 31 Μαρτίου 2013
Θέμα: Re: [SR-Users] Problem with Realtime Kamailio-Asterisk integration
 

On 30 March 2013 21:32, aaaa aaaa <indefix1 at yahoo.gr> wrote:

Hello,
>I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error:
> 
>[Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'.


Looks like you have an issue with your database table. The context being returned from the database appears to be "DEFAULT NULL" which is definitely not what you want. I don't know how you ended up with that value in your db by following the instructions.

Quick fix might be to change the value of the field sipusers.context for all entries to something  - say "mycontext" and then in your asterisk extensions.conf make sure that you are defining your extensions in this context, like so:

 [mycontext]
; our phones use 3 digit extensions, starting with 1
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup

(You will need to reload/restart Asterisk after making these changes) 

The issue you are having is an Asterisk one, not Kamailio and would be better asked in the Asterisk list if you dont get it working fr m the above..  
 

Hope this helps.

-Barry
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