Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk in the rtp captures. Asterisk seems to detect those and give SIP INFO to the SIP client. However it fails to detect some of the digits (which is random) hence the correct sequence of digits is not received at the SIP client.
I have tried setting relaxdtmf=yes in sip.conf but that does not seem to help. Can anyone help me out here? Regards, Mayur
_______________________________________________ -- 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
