[Asterisk-Users] g723 to g723 SIP call - warning message

[Sathya Weerasooriya]

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 <eric at fnords.org>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Reply-To: asterisk-users at lists.digium.com

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)