[Asterisk-Users] Asterisk to H.323 without gatekeeper

[bam]

I'm missing something here. I've put the following in extensions.conf and a 
few variations thereof. I've taken the sample configs and added to them, so 
when I dial 2200 from netmeeting * answers and runs me through the demo 
announcements.

The pots extensions 2200 and 2107 (TDM400) work fine calling each other and 
cause netmeeting to ring when I dial 3100, but the audio is one way 
pots->netmeeting when I answer in netmeeting.

If it is an RTFM situation please give me a URL, pretty postcard to anyone 
than can help me.

extensions.conf

[incoming-h323]

exten => 3001,1,Dial,OH323/192.153.153.64
exten => 3001,2,Busy
exten => 3001,102,Busy


[default]

include => incoming-h323
include => demo

exten => 2107,1,Dial(Zap/32,20)
exten => 2107,2,Voicemail(u2107)
exten => 2107,102,Voicemail(b2107)


exten => 2200,1,Dial(Zap/33,20)
exten => 2200,2,Voicemail(u2200)
exten => 2200,102,Voicemail(b2200)





At 13:42 19/12/03, you wrote:
>bam wrote:
>
>>I've read through the archives and have picked up that * does not need a 
>>gatekeeper to talk directly with an H323 handset to send and receive calls.
>>
>>I'm trying to go PSTN----*-----H323 and all the examples that I can find 
>>use a gatekeeper. Are there any examples or hints for doing it without 
>>the gatekeeper?
>>
>>many thanks in advance
>>
>>Brian
>
>
>[your_context]
>
>exten => _9XXXXXX,1,Dial,H323/78632${EXTEN:1}@192.168.100.2|30
>exten => _9XXXXXX,2,Busy
>exten => _9XXXXXX,102,Busy
>
>>
>>
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>--
>----
>---------
>Best Regards,
>     Pavel Litvinenko.
>     ICQ: 16224754
>     Ph: (8632) 923962, 923640
>     sip:litw at iptel.org
>
>
>
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