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I am very new to VOIP world. Even though i tried to intall asterisk. i have followed scratch install.. Everything is fine except two things, First is, voice breakage. I have tried direct calling from eyebeam with ulaw codec in both client and server side. Some words coming from other end is missing and my voice is clear to them. All together voice is breaking for me. My Server configuration is AMD Athlon 3000+ 1.8 Ghz 64bit. 1GB RAM. I am trying this in Fedora Core 4 64bit. I have tried a couple of installtion, all in vain. My second problem is nothing but, the lead preview is not coming for the agents before call. They are getting the lead info when the call connects.

what could be the reason and solution.

There are lot of reasons why the quality of the call is not good. You can check the latency of your asterisk server going to your voip provider. Also check that you have enough bandwidth and that the hops going to your voip provider isn't that many. Your server might also be overloaded or your voip provider isn't that good.

Your 2nd issue is not an issue at all. If you're on auto-dial, the lead info page will be filled once a call comes in.

I have checked all.. Moreover i have an alternate asterisk server running in the same network using the sam bandwidth and VOIP Gateway. Even i have checked the latency, its 39ms to my gateway, I have a 2Mbps 1:1 connectivity and its only6 hops to my VOIP gateway(I have see that running the command "mtr 202.54.112.194". I am trying to build up an alternate server. I have followed the scratch install on FC4 64bit. Will that be enough, The memory capacity of the server 1 GB.

Machine Configuratuion is

Code: Select all

Processors 1
Model AMD Athlon(tm) 64 Processor 3000+ @  40°C
CPU Speed 1.81 GHz
Cache Size 512.00 KB


Please help me

Seems everything is ok w/ your Asterisk server. I suggest you check the workstation you're running you're softphone. Better yet, try having a voip phone or an ata for testing purposes.

I am using EyeBeam.. but that tooo works fine with the existing server. Will that be a codec issue am using ulaw

does the voice break when you make calls to people on the same network?

No there is no such problemm...

benoyantony wrote:No there is no such problemm...

So the problem is when you make a call via your voip-provider?

if so, do you have annother provider you can try with?

But i have another asterisk server which works fine against the same VOIP gateway and Bandwidth, Only difference is its using G729 codec and am not sure that it has configured according to SCRATCH INSTALL

benoyantony wrote:But i have another asterisk server which works fine against the same VOIP gateway and Bandwidth, Only difference is its using G729 codec and am not sure that it has configured according to SCRATCH INSTALL

Ok, have you checked what codecs your voip-provider supports?