Problem with Call Quality "PLEASE HELP"

Well i am at a stumbling block, we have implemented asterisk and aheeva in a call center enviorment down in Panama City, Panama. We are expieriencing some of the following:

  1. Very bad voice quality, with static on the line, calls sometimes sounds like it is muted in spots.
  2. Not getting “ring back” from AIX softphones for every call. Some calls get ringback to know if the phone is being dialed but most do not.
  3. Both called parties and calling parties have to say “hello” several times before the call can actually be heard.
  4. just overall, a bad expierience with VOIP and asterisk in general.

Now, i understand that there are many many many installations of asterisk in production enviorments however, i cant imagine anyone being able to run a successful call center with this type of voice quality.

Our system is as follows:

Asterisk, is connected to a standard 2mb iinternet connection and the VOIP traffic is being terminated in miami. We have a cosco 2801 series router and a firewall on our network. Pretty standard equipment.

Now, all that being said i have aheeva and my local internet service provider pointing fingers at one another saying its “their” fault… What we have done so far for troubleshooting is put a standard analog VOIP gateway on the internet connection and used a standard “analog” phone to place calls. The call quality for this call is better, however it isnt nearly as good as standard TDM circuits. Not sure, what to expect as far as “true call quality”

If anyone can lead me in the correct direction, i would really appriciate it.


Here are a couple of things that you should think about.

1.) The call quality will depend on the codec that you use. If you are not already, try using the ulaw codec, it has the highest call quality.

2.) You need to monitor your bandwidth very closely, trying to send more calls than your bandwidth will allow will cause your call quality to decrease significantly. On a 2Mb connection you will be able to do a maximum of 25 simultaneous calls using the ulaw codec. 2000 kb / 80 kb = 25. Other codecs will give you different mileage. Try using a voip bandwidth calculator to fit the codec to your bandwidth / call capacity needs.

3.) Send calls over the public Internet at your own risk. If you are making calls over the public Internet, your traffic will likely cross multiple ISP’s networks (not just your ISP) and have to compete for the bandwidth on each of those network. So having a 2 Mb connection to your ISP doesn’t mean that you will get 2 Mb to your end destination. I would suggest running traceroutes between your network and your SIP provider’s network to see how many hops away it is and how much time it takes to get there. If there are too many hops or the latency is too high, then you may need to think about getting a private or point-to-point connection to your SIP provider.

4.) Lastly I would check the call quality on your internal network. Make a few calls between extensions directly connected to your Asterisk PBX. Perhaps there are some things in your configuration that you can tweak to get a higher call quality.

This should get you started, but is not an exhaustive list of all the possibilities.


with aheeva it uses asterisk, and a custom softphone called “Starphone” from my understanding this softphone uses ulaw and then gets converted by asterisk to G.729 when going to our termination point.

We are using a standard internet 2mb connection however, we are having this trouble with only one or 2 calls being placed at a time. Now, that being said i did get a bandwidth utilization reports from our provider and we are not ever going higher than 10% utilization on this link.

this is driving me crazy having two or three vendors pointing their fingers at one another.

one other thing we did do, is our ISP providor brought in an Avaya gateway plugged it directly into our internet connection and plugged two analog phones into the Avaya, and boom… the call quality was toll quality on 50 test calls that we made. This being said, i think it has to have something to do with the way asterisk is configured. However, i am not an expert on VOIP and asterisk.

Again, any help would be appriciated.

Thank You,



Download sinapps tester it will set a call up between sites and you will get a visual idea of whats going on. It may be a lot of jitter or latency on the pipe


From what you have described, I agree that it is probably something with your Asterisk configuration.

First I would recommend that you try making some calls between extensions (softphone) on your network. My guess (from what your are telling me) is that these calls will sound fine (you are using the ulaw codec).

If your internal calls sound fine, then it is probably the codec you are using between you and your VoIP provider. You said that you are using G.729 between your Asterisk machine and your VoIP provider. In my opinion the G.729 has poor sound quality. It is a low bit rate codec, and in general, the higher the bit rate, the better the quality. Not everyone will agree with me on this point, So I would recommend trying a few different codecs, if you can, to see if that makes a difference and to see which one you think sounds acceptable.

These are the general rules I use when choosing a codec. If that codec is available for use.
1.) High Bandwidth Available like internal network, use ulaw or alaw
2.) Medium Bandwidth Available like t-1, use g726
3.) Low Bandwidth Available like dsl or cable connection, use gsm


Have you tried to place the “qualify=yes” into the iax.conf section of this extensions?

After that do:

You can see there how much lag there is between your connections… and then we can start looking for the source of this lags.

Im my opinion if you are doing G729 you should be fine…

hope this helps


About a month ago we tried implementing VOIP from Panama over the internet to the US, and we were never able to obtain a decent quality of call. Since then, we converted the asterisk box to TDM service with standard E1 circuits. However, now it seems as thought we are having the same type of issues with the E1 service as we had with VOIP.

To me this doesnt seem even close to right, we should get outstanding voice quality with standard E1 circuits. Can someone please assist me in where to look for correcting our quality of voice calls.


Is Aheeva not assisting with this issue? As surely they have several production installations and may provide an optimal configuration for Asterisk for your environment.

Aheeva, is floundering around. Not much help so far at all… Now anyone that is familiar with Aheeva and its configuration here ia a brief rundown:

  1. Their softphone in manual dial works fine…
  2. The Likksys SIP phone works fine as well…
  3. their dialer within aheeva is where the voice quality goes south…

Any sugustions as to where to look as far as E1 configuration?


Just a few observations. What sort of LAN is all of this on? Are you using network switches that support QoS? If not then other network traffic may cause problems with voice traffic. If having problems even locally or using T1/E1 circuits then this might be a cause.

Is the 2Mb internet line shared with any other traffic? If so then prioritising the voice traffic using the CISCO router may help a bit. If you have a bad internet connection then you will never get good quality. When you said that it is 2Mb is that SDSL or a leased line?

You could try as this will give you a MOS reading for your connection. You can test with G.711 and G.729.

I’m not exactly sure here, but having dealt with asterisk auto dialers quite extensively, what is the loading like on the machine? are you getting a high system load when starting to run a campaign? if so, is this load related to the database running on the same box or just to asterisk (top should show you this)

Just a thought

Where does your box come in and what type of hardware do you have and how did u install asterisk on the box (someones ISO or did u do a install)

What Version OS is the asterisk box running ?

He stated Earlier its an Aheeva

they have thier own thing wrapped completely around asterisk

Curious, what version of Asterisk are you running?

I saw that but I went to the Aheeva site and did not see much info there

That is why I asked was it a ISO install (complete system like TB or did He / They install Asterisk and then the Aheeva suite.

I see issue with call quality all the time when folks use a TrixBox install of asterisk, some boxes want the asterisk / zaptel to be built from source.

As we all know in general Asterisk does not have the issues he is stating.