Hello Metik, 2009/12/6 Metik <freeswitch-users-list@metik.com>
You previously stated that your Cisco gateway has some "bug" that prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on the voip dial-peer that the call is using?
It is a PSTN dialpeer here, and it cannot be defined on it...
Unless you have configured the Cisco to support assymetric SDP or are using a non-default "rtp payload-type nte" setting that does not agree to well with FS's (default) "rfc2833-pt" setting, you should not have to use (SIP) INFO unless you want to.
I would recommend doing the following to ensure you are hitting the correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):
command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)
Unfortunately this does not work on PSTN dial peers.
Also, you can sift through "show sip-ua calls" for the call and ensure that the value of "Negotiated Dtmf-relay" is "rtp-nte".
This indeed shows that it has negotiated rtp-nte. Even when I do debug for CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them while it accepts them via INFO. As I said: I guess this is a bug. Since the gateway is on a remote site I hesitate on upgrading it until I hae the chance to go there. Thanks, __Yehavi: