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  • Enabling 16kHz audio for all FreeSWITCH services in sipXecs 4.4
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To enable 16kHz audio for all FreeSWITCH services in sipXecs 4.4, edit the following files:

/etc/sipxpbx/freeswitch/local_stream.conf.xml.vm

Change all instances of:

<param name="rate" value="8000"/>

to:

<param name="rate" value="16000"/>

/etc/sipxpbx/freeswitch/sofia.conf.xml.vm

change:

<param name="inbound-codec-negotiation" value="scrooge"/>

to:

<param name="inbound-codec-negotiation" value="generous"/>

/etc/sipxpbx/freeswitch/conf/autoload_configs/conference.conf.xml

change all instances of:

<param name="rate" value="8000"/>

to:

<param name="rate" value="16000"/>

Applying settings

Log into sipXconfig then browse to System >> Servers >> {server_name} >> Media Services then click OK. sipXecs will ask you to restart media services. Perform the Media Services restart then try out a conference bridge and you should hear HD audio

Alleviating Polycom G.722 audio garble for voicemail

To alleviate audio garble caused by Polycom phones, set SPEEX as the top codec in System >> Servers >> {server_name} >> Media Services then restart Media Services. The audio quality won't be quite as good as with G.722 for continuous streams but will help alleviate the garbled audio issue reported here: http://track.sipfoundry.org/browse/XTRN-1064

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