nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m <sam.gh1...@gmail.com> wrote: > 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 >
-- _____________________________________________________________________ -- 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