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

[Avi Miller]

Dinesh Nair wrote:
> 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. 

This was sort of my problem too. I have two Asterisk servers, with an 
IAX2 trunk between them:

Phone -> Asterisk 1 <- IAX -> Asterisk 2 <- H323 -> Avaya IP403 -> Phone

If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya, 
it worked fine. If I dialled from a phone on the Avaya, the SIP phone 
would ring, but the call would drop as soon as it was answered because 
of codec negotiation failure.

After removing the various disallow= and allow= lines, the codec 
negotation is now successful in both directions.

cYa,
Avi

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