call quality issues

We signed up with a couple weeks ago and have been having intermittent audio issues. Sometimes we’ll only have problems once a day, other days it’s pretty difficult to use. What we’re experiencing are glitches in the audio, usually where the caller will drop out for 3-5 seconds. I’ve also had a few issues with incoming calls ringing busy or the caller getting a “call could not be completed” message. Usually after calling right back the call will connect.

We’re using their Gold plan, but most of the time we’ve only got one call so we shouldn’t be overloading anything. Before we used, we were using FreePBX on a small server in our office and had no issues at all. The extension that I’m using was a remote extension even when we had our own server. The only configuration I changed was the server IP in the phone. still thinks that this is an issue on our end. They’ve asked for a couple trace routes and asked me to check my internet speed and both are acceptable. They’ve asked me to turn off SIP ALG, but my router is running DD-WRT and I don’t believe that’s an option.

I’m at a loss as to what I should check next. Any help would be appreciated.

This all sounds like a SIP issue with your carrier. You would only get a busy message if your carrier is not able to deliver the call. I would start with your SIP trunk provider. Do you have issues with ext to ext calls?

If you go to Reports, Asterisk Info and click on SIP Info what are some of the qualify times for your phones?

When I check Reports > Asterisk Info > SIP Info, the status for my main extension is OK (39 ms). We only have two extensions, and this is the one I use 90% of the time.

I thought of that, but we never had issues before we switched. We have accounts with two providers: and IIS Voip. We were always using IIS as our primary provider, but a few days ago I switched to to see if it made a difference on outbound calls. I’ve noticed the same issues with calls as I did with IIS Voip. Just to be sure though, I’ll call IIS to see if they have any advice.

In response to your question about extension to extension calls, we don’t really make any internal calls. Twice I’ve had issues with call quality when logging in to voice mail, but it’s not the same problem as on external calls. The issue I had with voice mail sounded like the server was overloaded. It would play two seconds of the password prompt, pause, play the next two seconds, pause, etc. The issue I have with outside calls is the callers audio will drop out for 3-5 seconds, and that usually only happens once or twice on a single call.

I’m still having this issue and the more I test, the more it seems (to me at least) to be a server issue. I’ve contacted our main SIP provider and they said they only handle the call setup and tear down. Audio isn’t passing through their servers, so they don’t see where this could be an issue on their end.

I tried defaulting my router and reconfiguring to see if there were any issues there, but no luck.

Finally, I tried making calls through DISA access using a landline and am still hearing the audio issues. That should completely eliminate my network as the cause.

Is there anything I can try to pinpoint the issue or other server configurations I might try? I also went through the old configuration of our on premise FreePBX server to compare and all the settings seem to be the same.

I really want to be able to make this work, but I’m completely out of my depth on what should be done to pinpoint and fix the issue.

If you only have two extensions and one VSP, have you thought of a PAP2 for $25 and use a VSP that provides voicemail?