I’m a newbie to Linux/Asterisk/FreePBX but I’m catching on quick. I’ve spent the last week trying to get a system up and running the way I want.
The only thing keeping me from rolling this out to production is audio issues on anything that I have recorded myself using the System Recordings Module from a connected phone or any of my voice mail messages that I have recorded. The audio seems to skip or stutter randomly every 5-10 seconds.
If I download the actual recorded .wav file to my computer and play it, it sounds crystal clear, so this appears to be a playback issue within Asterisk. The problem does not seem to present itself when playing any of the built-in Asterisk files (including Music on Hold). Also, this doesn’t appear to be an issue with my phones as the same problem presents itself whether I connect via a SIP extension or from an outside line via the Trunk.
CPU Usage stays below 1% while playing back the audio, so it doesn’t appear to be a performance issue.
The likely culprit, from what I have read so far, seems to be a timing issue. This is a VOIP only setup with no FXO/FXS card. I am using dahdi_dummy which appears to be up and running, but performing below par. dahdi_test gives Max 99.995% Min 99.683% Avg 99.883.
My asterisk server (CentOS 5.4) is running on a virtual machine under Windows Server 2008 Hyper-V R2 and has the latest (version 2.0) Integration Services for Linux installed. Unfortunately, I believe the virtual environment is what is causing my timing problems. The integration services offer support for accelerated disk and ethernet access via VMBUS, but I don’t believe there is any hardware timing or synchronization support.
I’m hoping someone on the forum can give me some direction on how to either improve my timing issues, or get around them. Since the built-in recordings and music on hold files are in ulaw format, and my SIP trunk and phones use ulaw, and those files seem to work, I would like to try using ulaw as the format for my own custom recording to see if that solves the problem. Unfortunately, I haven’t been able to find a way to change the default recording format for the System Recordings module or for voicemail. I did find the Call Recording Format setting within the General Settings Module, but this doesn’t seem to have any effect on the System Recordings or voicemail.
I would try recording the files locally in the ulaw format and then uploading them, but I’m not sure where I would even start to get a file recorded or converted into ulaw format.
Any help or suggestions on getting System Recordings to record in ulaw or in getting better timing results would be much appreciated.
Here are all of the relevant bits about my system setup that I can think of. Let me know if I am missing something relevant:
Kernel 2.6.18-164.11.1.el5 x86_64 SMP
I installed using the AsteriskNow 1.5 (x86_64) iso, then updated the kernel so that I could properly install the Microsoft Integration Services. After installing the Integration Services I did a yum update to update all modules to the latest versions. Then I had to manually fix a bunch of folder permissions and ownership settings that got overwritten to get everything working. Then I upgraded to FreePBX 184.108.40.206 using the GUI based upgrade module. Then I upgraded to Asterisk 220.127.116.11.1 from Asterisk 18.104.22.168 (I believe), however, the audio problems were present before the upgrade to Asterisk 1.6.