[asterisk-users] h323-sip: one way connection
Asghar Mohammad
asghar144 at gmail.com
Fri Apr 26 11:38:38 CDT 2013
try
UserByAlias=yes in general and type=user in user context.
On Fri, Apr 26, 2013 at 9:48 AM, s m <sam.gh1986 at gmail.com> wrote:
> oh yes, i'm using h323 not openh323
>
>
> On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>
>> nuFone h323 or openh323?
>>
>>
>> On Thu, Apr 25, 2013 at 9:33 PM, s m <sam.gh1986 at 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 <asghar144 at gmail.com>wrote:
>>>
>>>> what flavor of h323 you are using?
>>>>
>>>>
>>>> On Wed, Apr 24, 2013 at 8:50 AM, s m <sam.gh1986 at 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 <asghar144 at gmail.com>wrote:
>>>>>
>>>>>> try type=peer instead of friend.
>>>>>>
>>>>>>
>>>>>> On Tue, Apr 23, 2013 at 10:04 AM, s m <sam.gh1986 at 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 <
>>>>>>> asghar144 at gmail.com> wrote:
>>>>>>>
>>>>>>>> please post cli output for both calls.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Apr 22, 2013 at 11:32 AM, s m <sam.gh1986 at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130426/10236296/attachment.htm>
More information about the asterisk-users
mailing list