choppy sound when checking voice mail

Yes, that’s exactly the case, this is a fully VOIP system.

Sorry for not providing that information originally.

works for me...

Yes, but it does not work on all systems properly.

I was refered by another member to this thread, so I will attempt to describe out my issues which although similar, I do not believe they have to do explicity with the Zaptel based timing. (ztdummy) I have upgraded to Kernel 2.6.22.9 and using the svn checkout Zaptel 1.2 branch, which includes support for kernel based high resolution timers. I achieve an average zttest timing of over 99.996%, with a worst case of 99.98%, The issue is not neccessarily chop, or out of sync type, but instead litterally a fade out of the music on hold, the best way I have of describing it is a cordless phone as you walk away from the basestation (900MHz style cordless phone) it slow build up with white noise, and music can still be heard in sync in the white noise, but it reduces to almost no music output, and then comes in and out. This was the same function for the first machine which was a default trixbox install, and for the additional 2 machines, which were more custom ( mainly due to hardware). The original post is below:

I have installed Trixbox on 3 seperate machines, each with different architectures, all systems work perfectly with crisp sound for all functionality other than MOH. Current usage on Trixbox 2.2.4. System Architectures:

1) Intel 1 x P4 2.6GHz / 1GB RAM / 1 80GB IDE 7200 HDD

2) AMD 1 x Dual Opteron 1220 / 2GB RAM / 3WARE 9500-4LP RAID 5 4x250GB HDD

3) AMD 2 x Quad Opteron 2350 / 4GB RAM / 3WARE SE9650-8LPML RAID 6 6x250GB HDD

Ulizing IAX2 for inter-server trunking, and SIP for connection with Phones & FXO/FXS gateways.
Approximately 80 Phones distributed through the systems, + 16FXS (for communicating with old PBX) +16FXO (for remaining POTS that have not switched to SIP Trunks as of yet)

Now the Problem:
When making or receiving calls via the Sip trunks, or from the FXO/FXS clairity is perfect with no artifacts in the speech, when utilizing any built in sounds such as the queue functions with the time annoucements quality is perfect. However, when playing MOH through the queue, I am not sure how to properly describe it, but the symptom is as follows. Initial Music comes through fine, after approx 3 second delay it begins to degrade, as time goes on it almost sounds like static you would hear when an old cordless phone gets to far away from the base station, and then it will come back a little, and then fade some more (This is with only 1 inbound call routing to the Queue), no other active connections on either system tested. I have attempted this from the PSTN->SIP and from the VOIP provider->SIP, I have also tested from SIP Phone ->SIP. The only thing I am seeing in common is SIP. Now I know that asterisk did away with external mpg123, and switched to a "native MOH", but I seem to recall not having this issue with the older MPG123 on custom asterisk install of quite some time ago. Now I know the first question asked will be I don't have a proper timing sources such as an XP100, this may be true, however I have downloaded the current branch 1.2 zaptel driver, to use support for HPET timers in kernel, as I have also installed Kernel 2.6.22.9 in vain attempts following other posts about "choppy" sound as a method of correcting. when I issue the zttest, I get a best value of 100%, and average value of 99.995754% and a worst of 99.987793, so I am under the assumption from previous posts that "zaptel timing" is not the issue here. At first I thought it was a volume issue when I attempted to load up servers by making multiple calls from phones (different channels) to the Queue, but I realized the problems persist even with only 1 inbound call. I have see people come up with cures using Asterisk 1.4.xx, however I have never seen a full out address of Asterisk 1.2.xx.
What blows my mind is the fact that all sounds internal work perfectly but not the MOH. As an additional measure to try to see codecs might have been an issue, I went ahead and converted all the default MOH sounds to their respective codecs, ulaw, gsm, g729. I also altered codec settings for communication in asterisk to the trunks, and to the phones to see if it was a translation error which is is not. Still same result, Asterisk internal sounds such as the "You are next in line" perfect quality. Phone to Phone, including through the IAX trunk perfect, but MOH fades out, and or choppy.

If anyone has any suggestions I would love to hear them,

As it is , if this pertains only to Asterisk 1.2 variants., I have no problem going for an upgrade via source to 1.4, as well as supporting packages. I am aware that this breaks update capabilities, but since I am a Slackware guy by default source is easier to deal with then just waiting for a yum update.

If I have missed any critical specs, please let me know. As it stands these servers are not currently in production use do to this issue, One will never be in production use, and is simply used as a test dummy server, before applying soft updates to others. The end result of these servers will be more in automated proccessing applications through the PHPAGI extensions, but when someone needs to get a real person and is waiting in a queue it would be nice for MOH not to fall off.

I used asterisk 1.4 with freepbx on top, instead of an all-in-one trixbox (not for reasons critical of trixbox...) What, if any, plans are there to migrate trixbox to 1.4?

I am having the same problem on a new system I am trying to put together as a replacement. My calls on PSTN are perfect, extension to extension are perfect. VOIP calls inbound and outbound have loss of audio(Choppy). I am using a Supermicro 5015MF+ with a PDSMi motherboard, Xeon 3040 dual core 1.8g, 2g ram and a Sangoma A200 card. Is the Sangoma card not providing the timing source or do I really need a card to use the zaptel drivers.

I'm also having the choppy sound issue - but only on the system recordings (mailbox - conferences, etc)

the original test box I had was a very old Dell P2 233 with 384MB memory, this worked fine - but calls were being dropped and CPU usage was too high, that's ok, it was only a first pass test box, however - all the system recordings worked without problems

Next I took my old machine from home as a demo box, this is an AMD Athlon64 3000+ with 1GB memory, with an MSI K8T Neo2 v2. - This machine was very good at it, everything worked, system recordings were clear.

now it was time to build the server. we chose to use an Intel D945GCPE i945GC Socket 775 MOBO, with a Intel E2160 processor & 1GB Kingston Memory. - this machine although the calls are clear, the system recordings are choppy. what I did notice though, which may be of some interest to some of you, is that if I had the web GUI up, and I ordered it to do something while the system recording was being spoken - it went completely clear (as in sound worked without chopping) as soon as the page has finished loading, it starts chopping again - I do find this bizarre.

I've tried removing the WAV files - this just gave me nothing no sound atall - and the Mailbox number ( *98 ) just ended the call.

is there a way I can force it to use the GSM files for the system recordings?

Jason.