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.
I have the same on calls originating from a sip phone and going into a ZAP channel. Andre ----- Oorspronkelijk Bericht ----- ONDERWERP: --- Disclaimer This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. Finally, the recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050613/312a89f8/attachment.htm