[Asterisk-Users] Codecs and negotiations

[Eric Wieling]

Mike Meyer wrote:

> Stig,
> 
> 	I have not had that problem. Assuming there are no other CODECs allowed
> under the general and user sections, Your sip.conf looks OK. Can the
> stanaphone be configured itself for the codecs via a on-board web page
> like the GS and snoms? In my case the settings in the sip.conf always
> overrode what was defined on the phone itself though. So I don't really
> know what is going on with yours.
> 

The dial line from extensions.conf would be useful.