[asterisk-users] Change codec when dial from SIP to DAHDI

[d tbsky]

2014-09-26 3:34 GMT+08:00 Eric Wieling <EWieling at nyigc.com>:
> You will find not transcoding much less useful that one might imagine.
>
hi:
    can you give some more hint about the topic?

in my testing, if the sip phone use G.722 and the sip trunk use G.711,
I can hear the quality is not as good as both side use G.711.

    but you maybe right when both legs use G.711 but transcoding in
the middle. the quality seems not so bad but I have test it very
deeper..

    thanks a lot for your help!!

Regards,
tbskyd