Hi all. First off my setup is a dedicated server with a 2.2 GHz Quad core Xeon with 2 GB of RAM. I do have a TDM800P card install providing timing. Asterisk is 1.2.22. Only 15 SIP users on the system at the moment.
The issues is when I’m in a IVR menu, voice mail menu, listening to music on hold, or any action that plays audio off the server I get a quick breakup of audio about every 20 seconds. This does occur over a SIP connection over the LAN which is mostly 1 GB/s except to the phone. I can even make an incoming call through an analog line on the ZAP card and the menus garble up a little bit. Now this isn’t unusable but it doesn’t sound smooth like one would expect. I also noticed it only seems to be the built in audio files like MOH or menus. If I make a custom recording for the IVR off my Cisco phone everything is clear. Very strange.
Anyone have any tips on tracking down what is causing this? FYI the CPU max load average has never gone past 0.02. Peeked at 0.07 once. I’m kinda at a loss…