Troubleshooting Audio or call quality issue

Can you ssh to the server? If not, please do let us know so we can further check.

If you can ssh, first please make sure you are not out of space, then, let's do following to troubleshoot the issue. If you do not know how to ssh to your pbx, please see here

Following is done through Asterisk CLI - see here

1. Do you get dial tones on your phone extension? Can you call between your extensions? If not, first type below to see if connection ok or not

sip show peers

Run mtr when sip show peers showed failed registration (run mtr to your voip provider ip). If extensions are having problems, then we need mtr from your pbx to your sip phone local public IP, and vice versa. You can find your local pc public ip by going to www.whatismyip.com on your website browser. Make sure to run it continueslyrics WHILL you're having issues. It won't help if you run it during no issues. You can compare results from when no issues and when you have ussues.

On your pbx, you can yum install mtr if mtr ip-address not there

On your local pc, you can install winmtr.net

Then assuming your extension 704 not working. The easiest to debug this is to turn on sip debugging in your asterisk CLI (http://www.voip-info.org/wiki/view/Asterisk+CLI). Type

sip set debug peer 704 (something in this effect).

Then, you need to make a call to/from extension 704.

You can trace the sip chat. It could be the phone rejecting the call. It could also the phone is not responding to the PBX at all. For the later, double check your firewall. Some firewall may prematurely close your sip sessions.

2. If you can call between extensions and you get dial tones on your phone, then problem is not between your phone to our server. But if you still do not get incoming calls from outside nor able to make outgoing calls, then problem might be with your sip provider(s). To check, please ssh to your server and open asterisk debugging and type

sip show registry

If it doesn't say OK then there's no connection. same goes with sip show peers above. when no connection, try run mtr from/to your PBX server and your local phone locations. if on windows, you can use winmtr.net. we need to see if there maybe any backbone connection issues in between your place to the PBX server.

Having a audio issue is typically a congestion issue. This is just a guess on our part. If it is possible, you may want to capture packet during your call. It would be useful if you can run capture on both your PBX and your LAN. This is to confirm that you do have congestion. If you do have congestion issue, you will need to solve the point of congestion. That is why you need to do #1.

Lets start with the above, let see if we can learn something. Meanwhile, just to anticipate, it could be the congestion is in your Comcast router. You can help reduce it by the following steps here

Last thing you can do is to watch the log /var/log/asterisk/full as you are making the call. If possible, turn on sip debugging and more verbose and/or debug level. It will tell you your problem.

assuming your extension 704 not working. The easiest to debug this is to turn on sip debugging in your asterisk CLI. Type

sip set debug peer 704 (something in this effect).

Then, you need to make a call to/from extension 704.

You can trace the sip chat. It could be the phone rejecting the call. It could also the phone is not responding to the PBX at all. For the later, double check your firewall. Some firewall may prematurely close your sip sessions.

more info
- http://nerdvittles.com/?p=3026
- http://www.cadvision.com/blanchas/Asterisk/TroubleshootingSIP.html
- http://wiki.sangoma.com/asterisk-full-debug

3 Lastly, if no issues are found above, check your pbx loads and free ram during the incidents. Via ssh, you can type free and uptime or top.
Under top, pay attention to the loads, should be below 1.0 and the %id value, should be above 90 (if below 90 or above 1.0, you may consider adding more CPUs)
Under free, make sure swap usage is at 0 (if not, you may consider adding more ram)

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