[Asterisk-Dev] Dev Conf 2pm CST

[creslin at digium.com]

On Thu, Feb 10, 2005 at 02:22:04PM -0600, Rich Adamson wrote:
> > Please join #asterisk-dev and get your notes in line to start the dev 
> > conf.
> > 
> > IAX2/guest at 66.250.68.194/996
> > 
> > You will enter muted... you can mute and unmute yourself via *1
> > 
> > I ask that if you join please mute yourself if you put us on hold or 
> > have alot of background noise.
> 
> I keep getting 'Unable to negotiate codec' error while using gsm.
> Does this require another codec?

bandwidth = high
allow = all

Matthew Fredrickson