[Asterisk-Users] Re: Making a SIP call

[bclark at bwkip.com]

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 <eric at fnords.org>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Re: Making a SIP call
Reply-To: asterisk-users at lists.digium.com

bclark at bwkip.com 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)