first thing to do is
switch =>Realtime/@
then closely check which extension ur dialing.....
means length of extension u define in appdata 00XXXXXX
its length is 8 and if u dialed and extension greater then 8 or less then 8
then u cant go through...
for simplcity u check it with
appdata: SIP/_...@pstn GAteway
or
appdata: SIP/_...@pstn GAteway
i am not sure about _. work in this scenario.
u must apply appdata:SIP/${ext...@pstn gatwway
our use any variable which holds ur dialed no .
On Tue, Mar 3, 2009 at 4:09 AM, michel freiha <[email protected]> wrote:
> Hi all,
>
> I'm using asterisk in real time mode...My extensions.conf table contains:
>
> [default]
> switch => Realtime/defa...@extensions
>
> I have added the following to extensions.conf table;
>
> context:micho
> exten: _X.
> priority: 1
> app:Dial
> appdata: SIP/00xxx...@pstn GAteway
>
> Asterisk server is connected succeffully to database...As soon as i make a
> call i got the following error message:
>
> [Mar 3 00:52:02] NOTICE[2898]: chan_sip.c:14386 handle_request_invite:
> Call from 'username' to extension '00XXXX' rejected because extension not
> found.
>
> I'm dialing from an extension registered on asterisk with context micho
>
> Any clue?
>
> Regarda
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Best regards
Yawar Hadi Noshahi
Software Engineer
NGI Islamabad
(+92-0300-5504798)
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
