[asterisk-users] h323-sip: one way connection

[Asghar Mohammad]

please post cli output for both calls.


On Mon, Apr 22, 2013 at 11:32 AM, s m <sam.gh1986 at gmail.com> wrote:

> hello everybody
>
> i want to have sip connection between two asterisk systems (145 and
> 146). connection from 145 to 146 is ok but i can not call from 146 to
> 145.
> this is h323.conf file in 145:
> [peer146]
> host=192.168.0.146
> type=friend
> context=from-trunk
>
>
> [to-146]
> type=peer
> host=192.168.0.146
> faststart=yes
> tunneling=no
> progress_audio=yes
> disallow=all
> allow=alaw
> allow=ulaw
>
> this is mu extensions.conf file in 145:
>
> [from-trunk]
> exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1})
> [line-231]
> exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1})
>
> i have this error: dropping call because extensions '100', 's' and 'i'
> doesn't exists in context default".
>
> if i change "peer146" to "general", every thing is ok and i can call
> from two side. my question is: in h323 connection, is it a MUST to
> have "general" context in h323.conf? if not, why i have this error and
> how i can solve it?
> thanks in advance
> sam
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130422/e4e7d68f/attachment.htm>