[Asterisk-Users] Problem with DTMF Relay and Oh323

[Federico Alves]

When the inbound leg of the all is SIP and the outbound leg is Oh323
(Voip-to-Voip only here), the DTMF relay (either RFC2833 or SIP Info), fails
to go through, while it works perfectly when both legs of the call are SIP.
Is this a shortcoming of the Asterisk core or the Oh323 channel? Is this
solvable at all with some configuration change or a simple rewriting of the
Oh323 channel driver? Second question: how can I force the Oh323 to propose
only one codec to the outbound H323 endpoint, and do not negotiate? The
choice of codec is a business decision: if the gateway is located in my own
subnet I don't need compression, but if not I need to use only G29, etc.