In this scenario - there is packet loss between the client and your server.
Chances are your server is perfectly fine as it has proxied the calls
perfectly from telco to your server.
The problem is between your server and your client.
If it is **STATIC** then there is a problem with the microphone/handset
of the phones.
Digital Audio does not generate the static type noise randomly heard on
analogue or bad RJ11 type cables attached to the phone set.
Cheers!
--
FOUNDER & SR. TELECOM ANALYST
VOIPERNETICS COMMUNICATIONS
TEL: 647-847-2287 x2016
Chuck Mariotti wrote the following on 3/18/2014 11:34 AM:
We are having issues with call quality at random times during the day.
Client has 3 Yealink phones, OpenVPNed into our datacenter to our Asterisk box.
First time using Yealinks.
Our Asterisk box is using Unlimitel for SIP Trunks.
Most of the time, the client indicates that their customers cannot hear them
and they have to repeat themselves. It usually goes away.
A few times, neither party could hear each other and only a couple of times did
a second phone call need to be made.
The client reported specific calls/times with issues. I reviewed the call
reports form Unlimitel which reported 100% quality.
I turned on call recording and reviewed them... The recordings are perfect
quality... I can hear both parties asking to repeat themselves, etc (so I don't
have a recording of the static/breakup/bad audio)... just the conversation in
perfect quality.
If the recordings are clear, how can the call quality be poor? I am leaning
toward it being the link from their office to the datacenter...
Can anyone explain where the issue might be or how to debug it?
Is there a way to get packet loss/call quality reporting in Asterisk (PIAF)?
Chuck
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