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

Barry Flanagan barry at flanagan.ie
Sun Mar 31 21:35:41 CEST 2013


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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20130331/42a1c6a5/attachment.htm>


More information about the sr-users mailing list