[Asterisk-Users] Sip -> H323 using oh323 and G729

[David Allen]

Hi All,

	I believe I have fixed the problem now (had to do with frame sizes),
however what I need to be able to do is send through g729, g711a and g711u
through on the setup message (in case the remote endpoint is not G729) along
with RFC2833 signalling, however the only thing that is being sent is g729
and no RFC2833 message is being sent.

In the sip.conf, I have set up the order as follows:

disallow=all
allow=g729
allow=alaw
allow=ulaw

and I have dtmfmode=rfc2833 set in the config for the SIP user as well as on
the phone.

I have the oh323.conf setup as
userInputMode=RFC2833

and the order of the codecs is:
codec=G729
frames=2
codec=G711A
frames=20
codec=G711U
frames=20

Is there anything extra in the config files to get this going? or does oh323
not support this and the RFC2833 messages?

Thanks in advance.

David

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Michael
Manousos
Sent: Thursday, 22 July 2004 8:11 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Sip -> H323 using oh323 and G729



There is nothing wrong with asterisk-oh323, the call is rejected from
the remote endpoint. Try to turn on only G.729 and retry.

And yes, you don't need g729 licenses to do g729 passthrough.

Michael.