try UserByAlias=yes in general and type=user in user context.
On Fri, Apr 26, 2013 at 9:48 AM, s m <sam.gh1...@gmail.com> wrote: > oh yes, i'm using h323 not openh323 > > > On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad <asghar...@gmail.com>wrote: > >> 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 >> > > > -- > _____________________________________________________________________ > -- 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