becaus your call rules are are mismatched...
kishor > First of all explan your dial plan and extensions..... > > i will resolve that... > > Regards, > > Kishor kumar > > > >> Hello list, >> >> yesterday I finished work having my whole dialplan available... >> >> Today I want to make a call from one local phone to another and I get >> this >> : >> >> [Aug 28 10:48:57] NOTICE[1895]: chan_sip.c:15144 handle_request_invite: >> Call from 'test2' to extension '60' rejected because extension not >> found. >> >> >> Although I have this context : >> >> [from-TEST] >> >> where all my local extensions are defined... >> >> >> Yesterday all went fine, today it no longer works. >> >> >> With the command "dialplan show [tab]", I also see only a small part of >> all my defined contexts... >> >> Reloading, restarting... it all does not help... >> >> When I look at my file extensions.conf, it has not changed !! Asterisk >> just only loads 20% of the total dialplan... >> >> >> Using asterisk 1.4.30. Don't know which nightmare Asterisk had last >> night, but it's all messed up this morning ! >> >> >> Anyone has had the same experience yet ?! Any solution ?! >> >> >> Kind regards, >> >> Jonas. >> -- >> _____________________________________________________________________ >> -- 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
