Sound quality problem

Hello,

I have small office where I have linux Fedora server running Asterisk (2.4 Ghz procesor,
512MB RAM).My server is connected with PSTN via ISDN.I also have 4 operators who are connected to server in LAN.I registered phone number in PSTN from which I recieve calls.Costumers call and server give calls to avaliable operator.All operators in LAN have Xten XLite phone.Everything works perfect in LAN.

Now my business grows and I need more operators. I want to include operators who will work from home.For begining that will be 1 operator.So I get ADSL 768/192 Kbit/s in my office.My operator has 256/128 Kbit/s cable internet.

Problem is that on particular days costumers can not hear well my operator who works from home.That happends usually in the morning hours when I recieve the highest number of calls.In the afternoon when I recieve less calls everything is always fine.
I tried to use XLite , Eyebeam (with codec g729) and result is the same.There is no other program which is using bandwith at my operator’s computer.

What is the problem ? Is it server isue ? (maybe I need more memory or better CPU ?) or maybe connection speed issue ?Can you also tell me how meny operators I can connect to my present system ? How meny of them can work from home ?

Thank you

The server dosent seem like to be the issue. It seems more to be your internet. One of the things that can cause this is QOS. Make sure you have good QOS.

i would just check that g729 is the codec being used for these calls.

Yes when operator from home recieves calls via Eyebeam codec is g729.That is out of question.Would g711 be fine with that speed ? Sound quality is great usually in the afternoon (when I have less calls) but in the morning when I have more calls caller from the other side can not hear my operator who is working from home while in LAN everything works fine.

I forgot to tell you that my server is connected to PSTN via ISDN (30 lines maximum).
So teoreticly 30 persons (channels) can call me at one momemnt and 4 operators can speak with 4 of them at the same time while rest of 26 callers got music on hold until one operator become avaliable.[color=red]Maybe 26 Musics of hold and 4 speaking is more than my server can handle ?[/color] One more thing all operators can transfer calls to eachother on caller request.[color=red]I noticed that problem occurs when Operater from home is talking and other operator from LAN transfers call to her.[/color]

My configuration is following : all incoming calls from PSTN comes in one big callqueue whre are all agents , then first avaliable operator who is also in his own que (que1) gets the call and he can talk or transfer call to other operator.[color=red]Is it possible that this transfer use some bandwith from operater who is working from home and that for that reason sound quality become bad due to insuficient connection speed for more than 1 call ?[/color]

Sounds like your internet connection where the server is located.