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

[Asghar Mohammad]

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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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>                http://www.asterisk.org/hello
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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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