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

[s m]

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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>>
>>
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>
>
> --
> _____________________________________________________________________
> -- 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
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