Well I am getting the phones to ring but have no voice. When someone dials an IP number does this circumvent the * server? I was trying to make a capture of the call with ethereal but saw no traffic at the server for the call. Unfortunatly I have no way to set the dtmfmode on the phone side so I am stuck with inband. Is there something I am missing that is causing the lack of voice on the line.
Brian Date: Mon, 24 May 2004 16:20:36 -0500 From: Eric Wieling <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Making a SIP call Reply-To: [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: > I am still having this problem of only capturing part of the IP address, I > am currently checking into a possible hardware/software issue on the > client side but was wondering if there are any setting I need to set on > the asterisk server to allow an peer to peer call. I have set > dtmfmode=inband. Is there anything else I need to set? dtmfmode=inband only works with the ulaw and alaw codecs. If you use any other codec you MUST use rfc2833 or info DTMF modes (set on the phone AND on Asterisk) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
