hello all i,m newbie in asterisk and now want to sip and h323 connection. this is my scenario: phone(ext100)--->freepbx---sip--->system1---H323--->system2--->freepbx--->phone(ext200)
when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says "service unavailable". i debug asterisk in my system 2 and see below message: "Dropping call because extensions '200', 's' and 'i' doesn't exists in context [from-trunk]" i googled about this message and found that file extensions_mor_h323.conf should be included into /etc/asterisk/extensions_mor.conf. but i don't have any extensions_mor.conf file at all!!! is extensions_mor.conf really necessary to fix my problem?if yes, how i have connection in one way without this file? if no, how i can fix this problem? 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