[asterisk-users] h323-sip: one way connection
asghar144 at gmail.com
Mon Apr 22 10:53:26 CDT 2013
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
> this is h323.conf file in 145:
> this is mu extensions.conf file in 145:
> 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
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