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

[s m]

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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> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
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