what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m <[email protected]> wrote: > thanks Asghar, > i do it, but no thing happened:( > asterisk do not identify host line as ip address of the other end!!!! > > > On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad <[email protected]>wrote: > >> try type=peer instead of friend. >> >> >> On Tue, Apr 23, 2013 at 10:04 AM, s m <[email protected]> wrote: >> >>> 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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
