flavor? i do not understand what you mean. please explain more. thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad <asghar...@gmail.com>wrote: > what flavor of h323 you are using? > > > On Wed, Apr 24, 2013 at 8:50 AM, s m <sam.gh1...@gmail.com> 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 <asghar...@gmail.com>wrote: >> >>> try type=peer instead of friend. >>> >>> >>> On Tue, Apr 23, 2013 at 10:04 AM, s m <sam.gh1...@gmail.com> 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 >>>> <asghar...@gmail.com>wrote: >>>> >>>>> 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 >>>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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