We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband and make sure you're using a ulaw connection. If you use a lossy codec, it will scramble the DTMF tones.
Your config would change like so, [sipphone] type=peer host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=inband canreinvite=no disallow=all allow=ulaw -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jason DiCioccio Sent: Monday, August 08, 2005 10:27 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] DTMF issues with SIPPhone? Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=rfc2833 canreinvite=no Any ideas? Am I doing something wrong? Thanks! -JD- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
