[Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting

[Dinesh Nair]

On 04/05/06 13:52 Dinesh Nair said the following:
> 
> 
> On 04/05/06 13:17 Avi Miller said the following:
> 
>> I had a similar problem connecting Asterisk to an Avaya IP403 via 
>> OOH323: In the end, I removed all the disallow=all and allow=<codec> 
>> lines in Asterisk. This seems to have allowed the two systems to 
>> overcome the codec negotiation problems they were having and proceed 
>> with actual audio transfer. :)
> 
> 
> we'll try with this, but further testing reveals that the H.323 
> negotiation over port 1720 happens fine, with H.245 then being done over 
> another TCP port tuple. we didnt see the RTP port session being 
> created/negotiated. i'm assuming from the asterisk-ooh323 docs that it 
> uses asterisk's builtin RTP mechanism, and this should be over UDP. 
> there were no UDP packets being exchanged at all.
> 
> we will try your suggestion however.

more tests reveal that with ohphone, calls from SIP->ohphone work fine with 
audio passed both ways. however when ohphone calls a SIP device, the call 
is hungup when the SIP device answers. obviously, SIP-IAX and SIP-SIP calls 
work fine, so there's nothing wrong with the SIP device per se.

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)    http://www.alphaque.com/
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