please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m <sam.gh1...@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 >
-- _____________________________________________________________________ -- 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