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

[Sathya Weerasooriya]

I am sorry I mean dtmfmode=info

> -----Original Message-----
> From: Sathya Weerasooriya [mailto:sathyaw at sbcglobal.net]
> Sent: Wednesday, November 19, 2003 10:34 AM
> To: Eric Wieling; Asterisk-Users at Lists. Digium. Com
> Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
> 
> 
> 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)