On 12/28/08, jonathan augenstine <[email protected]> wrote: > I am trying to resolve an issue and I believe it is my configuration. The > scenario is that I have a SIP detected on the server. The dial plan then > makes a local connection to another part of the dial plan. The new dial > plan extension then places another SIP call out to a SIP phone. When the > call is accepted there is streamed from the calling SIP phone. When the > audio is complete a DTMF is transmitted to Asterisk. The DTMF is detected > by Asterisk but it does not get passed through to the other SIP phone. I > would like the DTMF to pass-through to the other SIP phone. Is this a > configuration issue? Or do I need to handle this on the dial plan level? > > Jonathan
Asterisk version? What are both dtmfmodes set to for each SIP endpoint? Are the calls natively bridged or bridged through Asterisk? MATT--- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
