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.
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