[asterisk-users] Problems With Playback of Audio On SIP Only System

[Brian Alexander]

I have been installing Asterisk as a SIP only system (no Digium Hardware)
for demonstration purposes. SIP users can connect to menus and voicemail
fine but the audio quality is terrible. The stock voicemail problems are bad
but basically understandable - voice menus recorded through the
asterisk-gui-2.0 are difficult to even understand.

The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have
configured the phone for ulaw to be it primary codec and set disallow all
and allow ulaw in the users.conf.

When that did not work I guessed that something was wrong with dahdi_dummy
but dahdi_test is showing results around 99.987%.

Here are the details of what software I have been using:
asterisk-1.4 (r168975)
dahdi-linux-complete 2.1.0 (r 5662)
asterisk-gui-2.0 (r4446)

The linux kernel is 2.6.24.6 built with 1000 Hz timer.

Thank you for your help, I am a stumped.

-Brian
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