[asterisk-users] h323-sip: one way connection

s m sam.gh1986 at gmail.com
Fri Apr 26 02:48:01 CDT 2013 

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
>>>>>>>>
>>>>>>>> --
>>>>>>>>
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>
>
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