Very poor voice quality with recorded .wav files

Hello,

I have asterisk-1.6.2.22 on CentOS system. My setup is working properly.
I want to play my own recorded files. Many times successfully I have played many files on many systems.
But on my this system wen I try to play the recorded files I m getting worst quality voice. Even cant hear it properly.

My recording file format : mono, 8000Hz sample rate, PCM

Any guess ?

-Urmi

Without a description of how it sounds wrong, I would have to guess that you have a pure VoIP system but don’t have an internal timing source.

The files should be 16 bit (linear) samples, but I think you would have noticed that. wav must be in lower case, but, if Asterisk accepted it in upper case, you would not get anything recognizable.

Using a virtual machine can also have this result.

Thank you very much for your reply.

I m not using Virtual machine. And I m using 16 bit linear samples, mono, 8000Hz sample rate, PCM recoded .wav file.

Wen I try to play the file its not properly audible. The voice quality is extremely poor.

My .wav files are of proper format. The same are also used in local server without any issue and with very good quality.

what can be the reason of very poor voice or chopped voice ? Is it a network issue ?

My live server is having VPN. I m making the test calls from softphone. One system is having VPN client connected with my live server and I have installed the softphone on it.

[quote]
Live Server ________________ [color=#0000BF]VPN Client[/color] _______ [color=#800080]Softphone installed[/color]
With V.P.N. _________[color=#0000BF]System[/color] [color=#800080]in VPN client System[/color][/quote]

-Thanks.

A network problem wouldn’t affect just custom recorded files.

Is the VPN TCP based. VoIP over TCP is never a good idea, but the network has to be in a bad way for there to be continual problems.

You have to understand that most people don’t have problems, so it is necessary to work out what is special in your case. Without detailed knowledge of your environment, everything here is going to be informed guesses.

Hello,

I have one server on Static IP(public IP) in my office premises. The same IVR system is working very well without any issues.
Its having good voice quality as well. I m using 16 bit linear samples, mono, 8000Hz sample rate, PCM recoded .wav files. Its working fine.

But when I try to play the same IVR on othe server having following scenario :

there are so much voice distortion. The IVR voice is not at all audible. Its choppy and soooo much disturbance is there.

The server info :
Intel Xeon ® E5410, 2.33GHz
2 mbps internet connection.
Not using telephony cards. SIP Trunk is used

what more information is required ?
Note : If I play asterisk sound files like :

[quote]/var/lib/asterisk/sounds/en/demo-congrats
/var/lib/asterisk/sounds/en/queue-thankyou[/quote]
It sounds properly without any issue. It plays with good quality.

please guide for this issue.

-Thanks.

Any help ?