Hi, Thanks Jeramy and Eric.
Sorry for my ignorance. I still did not get the point. Do you mean that I have to set each of my context in sip.conf with dtmfmode=inband ? I have the GS phone set as DTMF mode = Via SIP Info. Would that need to be change to something else ? (Send DTMF: in-audio via RTP (RFC2833) via SIP INFO) Cheers Sathya Date: Wed, 19 Nov 2003 06:15:35 -0600 From: Eric Wieling <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message Reply-To: [EMAIL PROTECTED] Jeremy McNamara wrote: > Don't try to do inland DTMF on anything but G.711. > > Jeremy McNamara > Someone really needs to patch Asterisk to print some ugly warning or notice to the Asterisk console when the codec that is being used for a call is not ulaw/alaw and trhe dtmfmode=inband (manyually or automagically set) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
