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

[Robert Jackson]

> -----Original Message-----
> From: Mike Meyer [mailto:mjmeyer at gendesign.com] 
> Sent: Tuesday, September 28, 2004 1:07 PM
> To: Asterisk Users Group
> Subject: [Asterisk-Users] CODECs and sip.conf and voice quality
> 
> 
> Another Caveat:
> Transfer does not work using the # key with the ILBC CODEC on 
> the GS phones. I can transfer only with the transfer button. 
> I have asterisk in the loop doing call supervision since I 
> have the tT option set in the dial command and 
> canreinvite=yes for the SIP phones. Anyone else have this problem?
> 

Shouldn't canreinvite be set to no to keep the phones from 
reinviting? I agree that with the tT flags * should still be
in the path, but I was just curious.