[Asterisk-Users] RE: Help with H.323

[Michael Manousos]

Dan wrote:
> Hi,
> 
> I have channel oh323 and there is no place in oh323.conf to enter something
> like you say (type=user in h323.conf called [100] with host=<valid ip
> address>
> 
> More, I have no gatekeeper for H.323, just the gateway included in the H.323
> channel.

Just add in your extensions.conf the following:

exten => 100,1,Dial(OH323/<ip_of_PC_with_Netmeeting>)

and make sure that you have
gatekeeper=DISABLE

in oh323.conf

> 
> I think that I need another solution.
> 
> Thank you,
> Dan

Michael.




> 
> ----- Original Message -----
> From: "Wade Weppler" <weppler at wwworks-inc.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, May 23, 2003 5:17 PM
> Subject: [Asterisk-Users] RE: Help with H.323
> 
> 
> 
>>>I can call the SIP phone from NM, but not reverse.
>>>How can I register a H.323 client without using a gatekeeper, in order
> 
> to
> 
>>>be
>>>able to define an extension for him?
>>
>>I spoke with Jeremy about this, and here's what I found by testing:
>>
>>I've setup a type=user in h323.conf called [100] with host=<valid ip
>>address>
>>
>>If I use:
>>
>>exten => 100,1,Dial,H323/100
>>
>>The call doesn't go through.
>>
>>If I use:
>>
>>exten => 100,1,Dial,H323/100@<valid ip address>
>>
>>It works fine.  According to Jeremy, chan_h323 first tries to resolve an
>>address through the gatekeeper.  If a gatekeeper isn't found (or used), it
>>can't resolve the name (100) to an IP address.  I suggested looking at the
>>IP in h323.conf if a gatekeeper isn't used.
>>
>>I'm not sure if Jeremy will take this any further or not, especially after
>>the last couple days of flame war.
>>
>>-wade
>>
>>
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>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
> 
> 
> 
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