[Asterisk-Users] CODECs and sip.conf and voice quality

[Nicolás Gudiño]

Hello,

On Tue, 28 Sep 2004 16:43:09 -0500, Mike Meyer <mjmeyer at gendesign.com> wrote:
> Robert,
[snip]

> In final; I tested ilbc with canreinvite=no just for kicks. Transfer
> still does not work with the #. Had to use the transfer button on the GS
> phone.

Try changing dtmfmode in the phone and in sip.conf. Inband DTMF will
not work using compressed codecs. Try INFO or RFC2833.

If you use tT in Dial, asterisk will stay in the path: canreinvite
will be ignored.

-- 
Nicolás Gudiño
Buenos Aires - Argentina