IVR system voice quality is very poor

Dear Gurus,
I am new user in freePBX,i have installed asterisk to connect my three office in one city.
with total 40 extension.

The first thing when i press *97 or i used other codes of IVR ,the voice quality is very poor ,even i can’t recognize.
please tell me where is the option to change IVR system Voice quality and secondly where i have to change the Voice call Quality.

Thanks… and waiting…

You have a issue in your setup someplace else then if you can’t understand the voicemail audio the audio quality is not a direct thing you can set.

Way to many things can effect the quality of the audio. Choice of audio codec, manufacture and quality of phones, network traffic, using analog phones with a cheep quality zaptel card, overall bandwidth usage between phone and server, bandwidth limits in and out between each of the locations, you have it running in a VM, quality of the server(s) by this I mean is it properly powered (not using a 100 Mhz 486), and not over loaded, and decent quality motherboard where it is not forcing all devices to share the same IRQ, just to name a few of the many reasons.

So can you start by providing details of your setup so we can start to eliminate some of the potential issues. Is this a distro build, if so who’s and what version, if hand built, what OS and who’s directions did you use, this on a real dedicated system or VM, phones make and model, codec’s used, etc. The more you provide the better there is a chance that we can determine what the real issue it and help you solve it.

I’m experiencing what I believe to be the same issue as outlined in the original post. I am running asterisk 1.4 with freepbx 2.5 and all modules updated. This is all running on Ubuntu 8.04, a 3 year old dell (plenty of hard drive space, ram, good processor, etc.). At this time, I only have 2 softphones (xlite).

I’m not really experiencing call quality issues. The voicemail prompts sound very garbled, staticy, and distorted. If I go into my mailbox options to record my name or my greeting, playing the recording back (within the voicemail system) sounds fantastic. But if I set a greeting as active and then call the voicemail from another extension, the greeting, along with the prompt, sounds distorted.

Any thoughts as to why this might be? Thanks for your help

My first guess would be that you don’t have a timing source. Meetme, iax trunking, MoH, voicemail, and streaming audio in general require a timing device. Look here for more: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy


There are many things that can effect audio quality. The first being the performance of the network. This includes the local system performance when using software based phones. If the local system is busy doing other things and you are using it with a Codec that does compression and have say a web browser downloading a huge file or sitting on a page with 4 flash files running yea it can get garbled real easily.

The next thing is interrupt sharing on the box. If the Nic and something else like the hard drive are sharing the same IRQ then it can easily effect the quality even if the load on the box is almost non-existent.

Google asterisk audio problem and you will get hundreds of pages with good tips for trouble shooting these things.