I currently have an asterisk server with several HD polycom 550’s and 650’s which are capable of passing HD voice. So far all the research that I’ve done shows them needing the G.722 codec in order to achieve this voice quality (widebound audio).
I believe I have already enabled it in Asterisk, but does anyone have any insight on how to modify the polycom config files (sip.cfg, local-settings.cfg) to enable the phones to pass the codec through asterisk?
Any help would be greatly appreciated.
Thanks,
Matt
So I’ve got the codec working between 2 HD phones, but when I call voicemail or a cell phone the sound is all choppy so it’s almost like asterisk is choosing the wrong codec (g722) for devices that can’t support it…
The only way I can get it to work between polycoms is to put g722 at the top priority of all the other codecs in sip.conf…
Any thoughts?
[general]
;Autocreatepeer=yes
context=sip ;default
maxexpiry=3600
checkmwi=10
vmexten=voicemail
videosupport=yes
rtcachefriends=yes
disallow=all
allow=g722
;allow=alaw
allow=gsm
allow=ulaw
allow=g729
allow=g723.1
;allow=h263
;allow=h263p
;allow=h263+
;allow=h264
;allow=h264+
rtptimeout=60
dtmfmode=auto
nat=yes
canreinvite=no
auth=md5