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

s m sam.gh1986 at gmail.com
Wed Apr 24 01:50:42 CDT 2013


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 --
>> 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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>
>
> --
> _____________________________________________________________________
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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