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."
[EMAIL PROTECTED] (0 0) http://www.alphaque.com/
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