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

[Asghar Mohammad]

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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>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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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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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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