i know what is the exactly problem. i enable debug for h323 and it says: "could not find user by name 200 or address 192.168.0.146"
when i change "peer-146" to "200" every thing is ok and i can call from two side. but it is not good for me because 200 is the name of extension and when i config asterisk systems, i don't know the name of extensions, therefore i should use addresses not name of extensions. do you know how i should define address of the other end in h323.conf file? i define the address by "host=192.168.0.146" but asterisk can not find it? why? On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad <[email protected]>wrote: > please post cli output for both calls. > > > On Mon, Apr 22, 2013 at 11:32 AM, s m <[email protected]> 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 >
-- _____________________________________________________________________ -- 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
