Wow! thanks for all the fast responses. Let me try to summerize the answers
for all the additional questions:
1. David: Yes, QoS is enabled on the firewall. The settins are the
following:
IAX2: Source :192.168.0.xx port 4569, Destination: 0.0.0.0, Port 4569
Protocol: UDP
SIP: Source :192.168.0.xx port 5060, Destination: 0.0.0.0, Port 5060
Protocol: UDP
RTP: Source :192.168.0.xx port 10000-20000, Destination: 0.0.0.0, Port
10000-20000 Protocol: UDP
where xx is the IP of my trixbox.
This seems to me that applies only to outgoing traffic but the device does
not allow me to put in at the same time an incoming rule for the same port
(Hotbrick LB2). I will try to modify the values you mentioned in the
sip.conf and iax.conf.
2. Ovidiu: Buna. The restart fix is only temporary.
3. Syed: I tried to minimize this problem, all ATA's are set to prioritize
the G711. THe below was just an example, you can hear the same losses all
the time during the conversation, so it is not the audio file format.
4. Jim: The QoS is on, the question is if it is set up ptoperly or not.
Please see the configuration at no. 1.
Wife wants to through the system out... I am in trouble... :)
Thanks for your help guys.
Zoltan.
----- Original Message -----
From: "Jim Van Meggelen" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Wednesday, November 01, 2006 11:41 AM
Subject: RE: [on-asterisk] Sound quality question
QoS is not simply a matter of having enough bandwidth. In fact, bandwidth
is
nearly always sufficient these days.
The trick is making sure the voice packets get priority treatment. If you
have not done any QoS work on your connection, every time you check your
email, or download a website with any amount of graphics in it, or
whatever,
those things will have as much right to the pipe as voice.
Jim
-----Original Message-----
From: Pittner, Zoltan [mailto:[EMAIL PROTECTED]
Sent: November 1, 2006 10:45 AM
To: [email protected]
Subject: [on-asterisk] Sound quality question
Hello,
I have my asterisk (trixbox) server set up in my home
environment.The internet connection is through Rogers Cable,
extreme edition, so there is plenty of advertised bandwidth
to both directions. (it is another fact that these advertised
values never live up to their advertised values...)
The system has two incoming lines, one POTS (Bell) through an
SPA 3102, and another directly from asterisk to Atlasvoice.
When the phone rings, the Digital receptionist announces:
"Welcome to our telephone system, please hold the line till I
transfer your call."
Then it rings a predefined ring group. - I know this is long
introduction, but here comes the problem:
When a call comes in on the VOIP (Atlasvoice line) - I mean
if I call my own number from another land line I hear the following:
"Welcom. .. our .elephon. ..stem, pleas. ..ld the line ..ll I
transf.. your call"
Now, I got pissed off for this and I changed the MTU setting
on my firewall, brought it down from 1500 to 1462. For 2 days
it was good, no losses. After 2 days, it is doing that again.
Also I have an extension in Hungary - on a softphone
(IAXComm) - and I almost all the time hear the same thing
doesn't matter what the MTU setting is. That made me think
that this may be coming from somewhere else not the MTU.
Do any of you have any idea what might be causing this?
Thank you,
Zoltan.
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