subject:"Re\: \[asterisk\-users\] Call quality"

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,  
 but 10 sim calls usually).  I've one user who complains of  
 intermittent bad calls, though I suspect the bad calls are across  
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that  
 Asterisk never uses more than 4-5% cpu, systems idle besides that.  
 Memory seems ok too. Network utilisation is  300kbps.  The voice  
 network (clients + server) sit on their own dedicated 100Mb  
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a  
 bad call, is stuttered speech, from BOTH sides (so local SIP client,  
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then  
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without  
 putting a wireshark PC on the switch, port-mirroring the server and  
 then capturing all of the traffic in a round-robin-type capture and  
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that  
 user and see if that helps.

 Any ideas welcome,


What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

I don't need QoS.

The voice network here is seperated from the PC LAN physically (2
separate switches), by design. So theres no web browsing etc on that 2mb
circuit.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...

I am assuming you've already implemented QOS? That is likely the problem
if the intermittent quality issue is only on calls between internal and
external parties.

If someone tries to access the yahoo homepage while someone else is on
the phone, without QOS, they are really going to be fighting for that
bandwidth.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but 10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,


What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

2009-06-02 Thread David Gibbons

2mb is small potatoes... unless you mean MegaBytes instead of Megabits...

I am assuming you've already implemented QOS? That is likely the problem if the 
intermittent quality issue is only on calls between internal and external 
parties.

If someone tries to access the yahoo homepage while someone else is on the 
phone, without QOS, they are really going to be fighting for that bandwidth.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but 10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,


What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

Scratch that,  my inventory tool says the system has 256Mb not 1Gb.
I wonder if a memory upgrade would help it out...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 02 June 2009 14:59
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,  
 but 10 sim calls usually).  I've one user who complains of  
 intermittent bad calls, though I suspect the bad calls are across  
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that  
 Asterisk never uses more than 4-5% cpu, systems idle besides that.  
 Memory seems ok too. Network utilisation is  300kbps.  The voice  
 network (clients + server) sit on their own dedicated 100Mb  
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a  
 bad call, is stuttered speech, from BOTH sides (so local SIP client,  
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then  
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without  
 putting a wireshark PC on the switch, port-mirroring the server and  
 then capturing all of the traffic in a round-robin-type capture and  
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that  
 user and see if that helps.

 Any ideas welcome,

What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

2009-06-02 Thread Darrick Hartman

Do you have any idea the number of bugs that have been fixed since 
1.4.15?  Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this.

On 06/02/2009 08:58 AM, Adrian Marsh wrote:
 Hi,

 It's a 2mb dedicated leased fibre line, with50% utilisation.
 My first thoughts were the internet link, but that wouldn't explain why
 the client transmit (other channel), which is on the same LAN as the
 server, would have the same problem at the same time.

 Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
 none of the stats show that going on at all, although my CPU stats are
 5min samples - so that might hide a 60s of intense CPU activity.

 It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
 Only runs Asterisk.

 Adrian


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
 Howes
 Sent: 02 June 2009 14:23
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Call quality - how to debug


 On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,

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Re: [asterisk-users] Call quality - how to debug

Yeah, I know,  but when I last tried an upgrade to 1.4.18 it broke the
whole IAX connectivity and I was forced to drop back.

I'll go:

1) Memory upgrade first
2) Clone the machine, and upgrade to latest 1.4.x

However - my question would still stand, how exactly would I be able to
debug whats going on in the RTP stream? And why its stuttering
(sometimes halfway through a call).

Any tips or tricks for actually debugging within Asterisk ?

Thanks,

Adrian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick
Hartman
Sent: 02 June 2009 15:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Do you have any idea the number of bugs that have been fixed since 
1.4.15?  Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug
this.

On 06/02/2009 08:58 AM, Adrian Marsh wrote:
 Hi,

 It's a 2mb dedicated leased fibre line, with50% utilisation.
 My first thoughts were the internet link, but that wouldn't explain
why
 the client transmit (other channel), which is on the same LAN as the
 server, would have the same problem at the same time.

 Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
 none of the stats show that going on at all, although my CPU stats are
 5min samples - so that might hide a 60s of intense CPU activity.

 It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
 Only runs Asterisk.

 Adrian


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
 Howes
 Sent: 02 June 2009 14:23
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Call quality - how to debug


 On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,

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Re: [asterisk-users] Call quality - how to debug

2009-06-02 Thread Jared Smith

On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote:
 However - my question would still stand, how exactly would I be able to
 debug whats going on in the RTP stream? And why its stuttering
 (sometimes halfway through a call).
 
 Any tips or tricks for actually debugging within Asterisk ?

Wireshark has a lot of RTP tools for looking at the latency and jitter
and dropped packets on the line, which are the most common problems I
find when helping people diagnose poor audio connections.  It won't tell
you what is *causing* the problem, but it will help you know what the
problem actually is.  

From there, you can start to track down the source of the problem one
network segment at a time.  For example... is the poor audio being
caused by network problems between the phone and Asterisk, or between
Asterisk and your upstream provider.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Call quality - how to debug

2009-06-02 Thread Steve Howes


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
 I’m at a loss of how to debug the voice issue further, without  
 putting a wireshark PC on the switch, port-mirroring the server and  
 then capturing all of the traffic in a round-robin-type capture and  
 even then I’m not sure what that will achieve.

Why not just tcpdump on the asterisk box then load it into wireshark?
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Re: [asterisk-users] Call quality - how to debug

2009-06-02 Thread David Gibbons

I think you're overlooking your internet uplink, which is what I'm talking 
about:

snip
Inbound calls are via in IAX trunk from Gradwell.
/snip

You certainly DO need QOS to maintain call quality over the INTERNET link.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh
Sent: Tuesday, June 02, 2009 10:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

I don't need QoS.

The voice network here is seperated from the PC LAN physically (2
separate switches), by design. So theres no web browsing etc on that 2mb
circuit.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...

I am assuming you've already implemented QOS? That is likely the problem
if the intermittent quality issue is only on calls between internal and
external parties.

If someone tries to access the yahoo homepage while someone else is on
the phone, without QOS, they are really going to be fighting for that
bandwidth.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but 10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,


What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

2009-06-02 Thread David Gibbons

Unless I've misunderstood and you're not running ANYTHING but voice over that 
internet uplink?

snip
So theres no web browsing etc on that 2mb circuit.
/snip

In which case, I stand corrected and you don't need QOS.

-Dave


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...

I am assuming you've already implemented QOS? That is likely the problem
if the intermittent quality issue is only on calls between internal and
external parties.

If someone tries to access the yahoo homepage while someone else is on
the phone, without QOS, they are really going to be fighting for that
bandwidth.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but 10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,


What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

Hi Dave,

You're quite right, it's a dedicated down and uplink to my ISP, and
Gradwell also has fibre connection into that ISP (so short hop to them)

The reason I don't think it's the fiber link, is that Asterisk recorded
the conversation as two channels. IN (from Gradwell), and OUT (from the
Cisco phone, that's on the same LAN as the asterisk server).  And I hear
distortion on both sides, at the same time.  As thats what asterisk
hears, and that part of the call is a same-LAN RTP stream, pre-ISP,
then that's why I don't think it's the IAX link.

That said, I've not got complaints from users making internal calls.  So
my thinking was maybe its an IAX/SIP conversion thing

As a test, I've switched my account, and the problem account to inbound
SIP, to see if that makes a difference. That makes it 100% SIP.

Next step, memory upgrade and the A*k upgrade.

Thanks,

Adrian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 16:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

Unless I've misunderstood and you're not running ANYTHING but voice over
that internet uplink?

snip
So theres no web browsing etc on that 2mb circuit.
/snip

In which case, I stand corrected and you don't need QOS.

-Dave


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...

I am assuming you've already implemented QOS? That is likely the problem
if the intermittent quality issue is only on calls between internal and
external parties.

If someone tries to access the yahoo homepage while someone else is on
the phone, without QOS, they are really going to be fighting for that
bandwidth.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Hi,

It's a 2mb dedicated leased fibre line, with 50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

 Hi All,

 I've a 1.4.15 A*k server supporting several users (approx 80 total,
 but 10 sim calls usually).  I've one user who complains of
 intermittent bad calls, though I suspect the bad calls are across
 the board, but intermittent.

 Inbound calls are via in IAX trunk from Gradwell. CPU stats say that
 Asterisk never uses more than 4-5% cpu, systems idle besides that.
 Memory seems ok too. Network utilisation is  300kbps.  The voice
 network (clients + server) sit on their own dedicated 100Mb
 switches.  Stats from the switch say its lightly loaded.

 I've turned on voicefile recording.  What we hear, when there is a
 bad call, is stuttered speech, from BOTH sides (so local SIP client,
 and remote IAX inbound call).
 Debug from asterisk just shows the call inbound, answered and then
 hung up as per normal.

 I'm at a loss of how to debug the voice issue further, without
 putting a wireshark PC on the switch, port-mirroring the server and
 then capturing all of the traffic in a round-robin-type capture and
 even then I'm not sure what that will achieve.

 I'm going to switch from IAX to SIP for the inbound calls for that
 user and see if that helps.

 Any ideas welcome,


What internet connection do you have...
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Re: [asterisk-users] Call quality - how to debug

Hi Steve,

Mainly because, if it were a CPU utilisation issue, then putting an
extra load on the server because of tcpdump isn't going to help.  If I
go that route then I'll port mirror on the switch.

But thanks for the reply,

Adrian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 16:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
 I'm at a loss of how to debug the voice issue further, without  
 putting a wireshark PC on the switch, port-mirroring the server and  
 then capturing all of the traffic in a round-robin-type capture and  
 even then I'm not sure what that will achieve.

Why not just tcpdump on the asterisk box then load it into wireshark?
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Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP

2008-11-03 Thread Bob Pierce


On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote:
 Any ideas why the audio quality would be so markedly different when
 the only thing that seems to be different is where the call is
 originating from (POTS line vs. SIP phone)?

Is it possible that calls from your POTS line are going across the VPN
as uLaw while the calls from the sip phones are using a compressed
codec?

Bob

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Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP

2008-11-03 Thread Lincoln King-Cliby

Bob,

It's conceivable, but how would I verify this and how would I change it if that 
was the problem?

The site that those calls terminate at is using an Asterisk Appliance so most 
of the config is done for you but it is possible to tweak the underlying 
configuration files (and I also have SSH access so I can do asterisk -v -r) 
-- If I know what I need to tweak.

Thanks,

Lincoln

--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
Crestron Authorized Independent Programmer

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce
Sent: Monday, November 03, 2008 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP


On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote:
 Any ideas why the audio quality would be so markedly different when
 the only thing that seems to be different is where the call is
 originating from (POTS line vs. SIP phone)?

Is it possible that calls from your POTS line are going across the VPN
as uLaw while the calls from the sip phones are using a compressed
codec?

Bob

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Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP

2008-11-03 Thread Lincoln King-Cliby

Just as an interesting follow-up/additional information, if I place a call to 
Site 2 on a POTS line, someone at Site 2 answers the call (using one of the 
Cisco phones) and then transfers it to me across the VPN the call sounds fine.

So I think Bob's question was on the right track with it being a CODEC issue, 
but I'm not sure how I need to deal with that for the ZAP channel type.

Thanks again,

Lincoln

--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
Crestron Authorized Independent Programmer

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln 
King-Cliby
Sent: Monday, November 03, 2008 1:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP

Bob,

It's conceivable, but how would I verify this and how would I change it if that 
was the problem?

The site that those calls terminate at is using an Asterisk Appliance so most 
of the config is done for you but it is possible to tweak the underlying 
configuration files (and I also have SSH access so I can do asterisk -v -r) 
-- If I know what I need to tweak.

Thanks,

Lincoln

--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
Crestron Authorized Independent Programmer

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce
Sent: Monday, November 03, 2008 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP


On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote:
 Any ideas why the audio quality would be so markedly different when
 the only thing that seems to be different is where the call is
 originating from (POTS line vs. SIP phone)?

Is it possible that calls from your POTS line are going across the VPN
as uLaw while the calls from the sip phones are using a compressed
codec?

Bob

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Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP

2008-11-03 Thread Bob Pierce


On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote:
 It's conceivable, but how would I verify this and how would I change
 it if that was the problem?

There's a few things you can do here.
1) Check the sip.conf on both sides to see what is defined there for the
trunk. Look for some disallow and allow statements. If they are there,
that will tell Asterisk what codecs to use on that trunk.

2) You could also check the codec that is in use during a call by
looking at the sip channel. From the asterisk CLI, start with show
channel SIP/ and tab it out to complete the command showing the trunk
between your two systems. I believe the codecs are listed here as
NativeFormats and ReadFormat. You could check this under both of
your scenarios to see if there is a different codec in use.

3) If you'd like to try and force the use of a compressed codec such as
GSM between your two sites, you would just need to make sure that both
sides had the following lines in the definition for the trunk in
sip.conf and then do a 'reload chan_sip.so from the Asterisk CLI:
disallow=all
allow=gsm

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Re: [asterisk-users] Call quality

2008-07-03 Thread Loic Didelot

Hello,
this is the case. Idle goes to 0% and IRQ goes to 100%.

I have a Junghanns ISDN card (bristuff) card. And I guess it is using
that Echo Canceler.

Best regards,
Loic Didelot.



On Thu, 2008-07-03 at 14:52 +1200, Matt Riddell wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Loic Didelot wrote:
  Hi,
  I am using g711a everywhere.
  
  I checked on a completely idle system (no calls at all) and idle CPU is
  dropping from 100% to 0% more than once per minute.
 
 If you run top, and the idle goes to 0% is it the IRQ that is using the
 other 100%?
 
 If so, what echo canceller are you using?
 
 - --
 Kind Regards,
 
 Matt Riddell
 Director
 ___
 
 http://www.venturevoip.com (Great new VoIP end to end solution)
 http://www.venturevoip.com/news.php (Daily Asterisk News - html)
 http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.7 (MingW32)
 Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
 
 iD8DBQFIbD7dDQNt8rg0Kp4RAql0AJ9hUDFqaNbliJTCLiKvR9BT+rbdNwCgqUjh
 tZxWREnPeYuO5h1PgrXxv30=
 =pPAe
 -END PGP SIGNATURE-
 
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 Register Now: http://www.astricon.net
 
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-- 
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Call quality

2008-07-02 Thread Gavin Henry

2008/7/2 Loic Didelot [EMAIL PROTECTED]:
 Depends on the phone.

 On many devices you can setup buttons to call a url. Thats what I did.

Ah, yes. Would be a good thing to implement here. Then you can do
anything, like a support ticket etc.

Cheers.

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Re: [asterisk-users] Call quality

2008-07-02 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Loic Didelot wrote:
 Hi,
 I am using g711a everywhere.
 
 I checked on a completely idle system (no calls at all) and idle CPU is
 dropping from 100% to 0% more than once per minute.

If you run top, and the idle goes to 0% is it the IRQ that is using the
other 100%?

If so, what echo canceller are you using?

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFIbD7dDQNt8rg0Kp4RAql0AJ9hUDFqaNbliJTCLiKvR9BT+rbdNwCgqUjh
tZxWREnPeYuO5h1PgrXxv30=
=pPAe
-END PGP SIGNATURE-

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Re: [asterisk-users] Call quality

2008-07-01 Thread Steve Davies

2008/7/1 Loic Didelot [EMAIL PROTECTED]:
 Hello,
 one of my customers complained about bad voice quality on several calls,
 so I programmed a button on each phone which users can hit if they have
 audio drops and echo.

 I did this to check if there is a common recurrent problem to a given
 destination or just for one user etc... But till now I could not detect
 a pattern which could explain the problems

 This alert button is pressed between 7%-10% of all calls. The customer
 has 25 phones and around 300 calls per day.

 The SNOM phones are connected to Linksys switches and are totaly split
 from the computers network. The same goes for the asterisk box. No calls
 are routed trough the internet.
 Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier

 The carrier we use is known for his good quality and we never had a
 problem. It is the historic and most expensive carrier in Luxembourg.

 Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
 maximum of 6 concurrent calls.


Which version of asterisk/zaptel, and which echo canceler is running in Zaptel?

Regards,
Steve

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Re: [asterisk-users] Call quality

I tried to get a little into cpu utilization and found the following
results.

Can they help me to come to a conclusion?

Best regards,
Loic Didelot.

[EMAIL PROTECTED]:~# mpstat 1
Linux 2.6.22-14-server (ppsite1)07/01/2008

02:54:40 PM  CPU   %user   %nice%sys %iowait%irq   %soft  %steal   
%idleintr/s
02:54:41 PM  all0.000.001.000.000.000.000.00   
99.00   4210.00
02:54:42 PM  all0.000.000.000.000.000.000.00  
100.00   4207.00
02:54:43 PM  all0.000.000.000.000.000.000.00  
100.00   4208.00
02:54:44 PM  all0.000.000.000.00   45.000.000.00   
55.00   4127.00
02:54:45 PM  all0.000.000.000.00   97.000.000.00
3.00   4148.00
02:54:46 PM  all0.000.000.000.00   93.004.000.00
3.00   4195.00
02:54:47 PM  all0.000.000.000.00   92.006.000.00
2.00   4175.00
02:54:48 PM  all0.000.000.001.00   91.002.000.00
6.00   4154.00
02:54:49 PM  all0.000.000.000.00  100.000.000.00
0.00   4069.00
02:54:50 PM  all0.000.000.000.00   23.000.000.00   
77.00   4125.00
02:54:51 PM  all2.000.000.000.00   19.000.000.00   
79.00   4123.00
02:54:52 PM  all0.000.000.000.000.000.000.00  
100.00   4236.00
02:54:53 PM  all0.000.006.002.000.003.000.00   
89.00   4302.00
02:54:54 PM  all0.000.005.000.000.003.000.00   
92.00   4267.00
02:54:55 PM  all0.000.00   20.000.000.001.000.00   
79.00   4328.00
02:54:56 PM  all0.000.000.000.009.000.000.00   
91.00   4352.00
02:54:57 PM  all0.000.000.00   49.00   46.000.000.00
5.00   4376.00
02:54:58 PM  all0.000.000.000.000.000.000.00  
100.00   4350.00
02:54:59 PM  all   11.000.002.00   36.000.000.000.00   
51.00   4237.00
02:55:00 PM  all0.000.000.00  100.000.000.000.00
0.00   4221.00
02:55:01 PM  all1.000.001.00   62.00   36.000.000.00
0.00   4318.00
02:55:02 PM  all0.000.000.002.00   98.000.000.00
0.00   4219.00
02:55:03 PM  all0.000.000.000.00  100.000.000.00
0.00   4342.00
02:55:04 PM  all0.000.000.002.00   98.000.000.00
0.00   4236.00
02:55:05 PM  all   14.000.004.00   20.00   62.000.000.00
0.00   4229.00
02:55:06 PM  all   39.000.003.00   38.00   18.001.000.00
1.00   4346.00
02:55:07 PM  all8.000.008.00   79.003.001.000.00
1.00   4240.00
02:55:08 PM  all1.000.000.00   98.000.000.000.00
1.00   4217.00
02:55:09 PM  all0.000.001.006.000.000.000.00   
93.00   4167.00
02:55:10 PM  all0.000.000.000.00   25.000.000.00   
75.00   4132.00
02:55:11 PM  all0.000.000.000.00   75.000.000.00   
25.00   4117.00
02:55:12 PM  all0.000.000.000.00   53.000.000.00   
47.00   4130.00
02:55:13 PM  all0.000.000.000.000.000.000.00  
100.00   4103.00
02:55:14 PM  all0.000.000.00   50.000.000.000.00   
50.00   4124.00
02:55:15 PM  all0.000.001.000.000.000.000.00   
99.00   4216.00
02:55:16 PM  all1.000.000.000.00   32.000.000.00   
67.00   4214.00
02:55:17 PM  all0.000.000.000.00   98.000.000.00
2.00   4209.00
02:55:18 PM  all0.000.000.000.00   94.000.000.00
6.00   4220.00
02:55:19 PM  all0.000.000.000.00   58.000.000.00   
42.00   4216.00
02:55:20 PM  all1.000.000.000.000.000.000.00   
99.00   4204.00
02:55:21 PM  all1.000.000.000.000.000.000.00   
99.00   4210.00
02:55:22 PM  all1.000.000.000.000.000.000.00   
99.00   4234.00
02:55:23 PM  all0.000.001.000.000.000.000.00   
99.00   4202.00
02:55:24 PM  all0.000.001.000.000.000.000.00   
99.00   4109.00
02:55:25 PM  all1.000.001.001.00   53.001.000.00   
43.00   4179.00
02:55:26 PM  all0.000.000.000.00   35.000.000.00   
65.00   4213.00
02:55:27 PM  all0.000.001.000.000.000.000.00   
99.00   4204.00
02:55:28 PM  all0.000.001.000.000.000.000.00   
99.00   4169.00
02:55:29 PM  all0.000.003.960.00   37.620.990.00   
57.43   4149.50
02:55:30 PM  all0.000.001.000.000.000.000.00   
99.00   4208.00
02:55:31 PM  all0.000.000.00   16.003.000.000.00

Re: [asterisk-users] Call quality

On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote:
 Hello,
 one of my customers complained about bad voice quality on several calls,
 so I programmed a button on each phone which users can hit if they have
 audio drops and echo.

 I did this to check if there is a common recurrent problem to a given
 destination or just for one user etc... But till now I could not detect
 a pattern which could explain the problems

 This alert button is pressed between 7%-10% of all calls. The customer
 has 25 phones and around 300 calls per day.

 The SNOM phones are connected to Linksys switches and are totaly split
 from the computers network. The same goes for the asterisk box. No calls
 are routed trough the internet.
 Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier

 The carrier we use is known for his good quality and we never had a
 problem. It is the historic and most expensive carrier in Luxembourg.

 Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
 maximum of 6 concurrent calls.

 Maybe someone can help me to track down the problem. What should I
 check, monitor test. Any ideas are welcome.


 Best regards,
 Loic Didelot.

 --
 Loïc DIDELOT
 MIXvoip S.a.
 [EMAIL PROTECTED]
 http://www.mixvoip.com

Is this a new install or a new problem?

If it is a new problem, what has changed?

If it is a new install, I would not rule out the provider, the more
historic may or may not be a good thing.  Describe the audio when it
is poor, popping, clicking, hissing?

Have you tried running a debug on the spans?

Thanks,
Steve T

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Re: [asterisk-users] Call quality

I/O wait is very suspicious.  What is your hardware platform?  Is this
just a plain Jane PBX or are you doing anything unusual?

Thanks,
Steve T

On Tue, Jul 1, 2008 at 8:57 AM, Loic Didelot [EMAIL PROTECTED] wrote:
 I tried to get a little into cpu utilization and found the following
 results.

 Can they help me to come to a conclusion?

 Best regards,
 Loic Didelot.

 [EMAIL PROTECTED]:~# mpstat 1
 Linux 2.6.22-14-server (ppsite1)07/01/2008

 02:54:40 PM  CPU   %user   %nice%sys %iowait%irq   %soft  %steal   
 %idleintr/s
 02:54:41 PM  all0.000.001.000.000.000.000.00   
 99.00   4210.00
 02:54:42 PM  all0.000.000.000.000.000.000.00  
 100.00   4207.00
 02:54:43 PM  all0.000.000.000.000.000.000.00  
 100.00   4208.00
 02:54:44 PM  all0.000.000.000.00   45.000.000.00   
 55.00   4127.00
 02:54:45 PM  all0.000.000.000.00   97.000.000.00
 3.00   4148.00
 02:54:46 PM  all0.000.000.000.00   93.004.000.00
 3.00   4195.00
 02:54:47 PM  all0.000.000.000.00   92.006.000.00
 2.00   4175.00
 02:54:48 PM  all0.000.000.001.00   91.002.000.00
 6.00   4154.00
 02:54:49 PM  all0.000.000.000.00  100.000.000.00
 0.00   4069.00
 02:54:50 PM  all0.000.000.000.00   23.000.000.00   
 77.00   4125.00
 02:54:51 PM  all2.000.000.000.00   19.000.000.00   
 79.00   4123.00
 02:54:52 PM  all0.000.000.000.000.000.000.00  
 100.00   4236.00
 02:54:53 PM  all0.000.006.002.000.003.000.00   
 89.00   4302.00
 02:54:54 PM  all0.000.005.000.000.003.000.00   
 92.00   4267.00
 02:54:55 PM  all0.000.00   20.000.000.001.000.00   
 79.00   4328.00
 02:54:56 PM  all0.000.000.000.009.000.000.00   
 91.00   4352.00
 02:54:57 PM  all0.000.000.00   49.00   46.000.000.00
 5.00   4376.00
 02:54:58 PM  all0.000.000.000.000.000.000.00  
 100.00   4350.00
 02:54:59 PM  all   11.000.002.00   36.000.000.000.00   
 51.00   4237.00
 02:55:00 PM  all0.000.000.00  100.000.000.000.00
 0.00   4221.00
 02:55:01 PM  all1.000.001.00   62.00   36.000.000.00
 0.00   4318.00
 02:55:02 PM  all0.000.000.002.00   98.000.000.00
 0.00   4219.00
 02:55:03 PM  all0.000.000.000.00  100.000.000.00
 0.00   4342.00
 02:55:04 PM  all0.000.000.002.00   98.000.000.00
 0.00   4236.00
 02:55:05 PM  all   14.000.004.00   20.00   62.000.000.00
 0.00   4229.00
 02:55:06 PM  all   39.000.003.00   38.00   18.001.000.00
 1.00   4346.00
 02:55:07 PM  all8.000.008.00   79.003.001.000.00
 1.00   4240.00
 02:55:08 PM  all1.000.000.00   98.000.000.000.00
 1.00   4217.00
 02:55:09 PM  all0.000.001.006.000.000.000.00   
 93.00   4167.00
 02:55:10 PM  all0.000.000.000.00   25.000.000.00   
 75.00   4132.00
 02:55:11 PM  all0.000.000.000.00   75.000.000.00   
 25.00   4117.00
 02:55:12 PM  all0.000.000.000.00   53.000.000.00   
 47.00   4130.00
 02:55:13 PM  all0.000.000.000.000.000.000.00  
 100.00   4103.00
 02:55:14 PM  all0.000.000.00   50.000.000.000.00   
 50.00   4124.00
 02:55:15 PM  all0.000.001.000.000.000.000.00   
 99.00   4216.00
 02:55:16 PM  all1.000.000.000.00   32.000.000.00   
 67.00   4214.00
 02:55:17 PM  all0.000.000.000.00   98.000.000.00
 2.00   4209.00
 02:55:18 PM  all0.000.000.000.00   94.000.000.00
 6.00   4220.00
 02:55:19 PM  all0.000.000.000.00   58.000.000.00   
 42.00   4216.00
 02:55:20 PM  all1.000.000.000.000.000.000.00   
 99.00   4204.00
 02:55:21 PM  all1.000.000.000.000.000.000.00   
 99.00   4210.00
 02:55:22 PM  all1.000.000.000.000.000.000.00   
 99.00   4234.00
 02:55:23 PM  all0.000.001.000.000.000.000.00   
 99.00   4202.00
 02:55:24 PM  all0.000.001.000.000.000.000.00   
 99.00   4109.00
 02:55:25 PM  all1.000.001.001.00   53.001.000.00   
 43.00   4179.00
 02:55:26 PM  all0.000.000.000.00   35.000.000.00   
 65.00   4213.00
 02:55:27 PM  all0.000.001.000.000.000.000.00   
 99.00   4204.00
 02:55:28 PM  all0.00

Re: [asterisk-users] Call quality

Hi,
its a new installation in a new office. Customer moved in, so right
moment to get a new PBX.

The box is running asterisk, nothing else:
 - asterisk
 - postfix just to send out voicemails
 - no realtime
 - som AGIS at call setup and call end
 - Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2
 - zaptel-1.4.10

We use a Junghann BRI card and a XORCOMM Analog Astribank. But only one
modem and 2 fax devices are connected to the astribank.


I did not do a debug on the spans. Anythin special I should look for?

Difficult to describe the audio:
 - basically echo is appearing
 - audio problems are only one way
 - audio has cuts when speaking



Best regards,
Loic Didelot.




On Tue, 2008-07-01 at 08:58 -0400, Steve Totaro wrote:
 On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote:
  Hello,
  one of my customers complained about bad voice quality on several calls,
  so I programmed a button on each phone which users can hit if they have
  audio drops and echo.
 
  I did this to check if there is a common recurrent problem to a given
  destination or just for one user etc... But till now I could not detect
  a pattern which could explain the problems
 
  This alert button is pressed between 7%-10% of all calls. The customer
  has 25 phones and around 300 calls per day.
 
  The SNOM phones are connected to Linksys switches and are totaly split
  from the computers network. The same goes for the asterisk box. No calls
  are routed trough the internet.
  Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier
 
  The carrier we use is known for his good quality and we never had a
  problem. It is the historic and most expensive carrier in Luxembourg.
 
  Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
  maximum of 6 concurrent calls.
 
  Maybe someone can help me to track down the problem. What should I
  check, monitor test. Any ideas are welcome.
 
 
  Best regards,
  Loic Didelot.
 
  --
  Loïc DIDELOT
  MIXvoip S.a.
  [EMAIL PROTECTED]
  http://www.mixvoip.com
 
 Is this a new install or a new problem?
 
 If it is a new problem, what has changed?
 
 If it is a new install, I would not rule out the provider, the more
 historic may or may not be a good thing.  Describe the audio when it
 is poor, popping, clicking, hissing?
 
 Have you tried running a debug on the spans?
 
 Thanks,
 Steve T
 
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 asterisk-users mailing list
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-- 
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Call quality

2008-07-01 Thread David Backeberg

On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote:
 Maybe someone can help me to track down the problem. What should I
 check, monitor test. Any ideas are welcome.

If there are no legal reasons not to, consider recording all calls for
a limited time. It's easier for engineers to debug a voice quality
problem when they have a recording of exactly what it sounds like.
It's possible that different people are complaining about different
perceptions of what they consider a voice quality problem, and that
the problem might not even be on your end of the conversation.

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Re: [asterisk-users] Call quality

Hello,
I forgot to include CPU information

[EMAIL PROTECTED]:/usr/src/bristuff-0.4.0-RC2# cat /proc/cpuinfo 
processor   : 0
vendor_id   : CentaurHauls
cpu family  : 6
model   : 10
model name  : VIA Esther processor 1000MHz
stepping: 9
cpu MHz : 1000.127
cache size  : 128 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 1
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat
clflush acpi mmx fxsr sse sse2 tm nx up pni est tm2 rng rng_en ace
ace_en ace2 ace2_en phe phe_en pmm pmm_en
bogomips: 2002.19
clflush size: 64

The box is running asterisk, nothing else:
 - asterisk
 - postfix just to send out voicemails
 - no realtime
 - som AGIS at call setup and call end
 - Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2
 - zaptel-1.4.10

Best regards,
Loic Didelot.



On Tue, 2008-07-01 at 13:54 +0100, Steve Davies wrote:
 2008/7/1 Loic Didelot [EMAIL PROTECTED]:
  Hello,
  one of my customers complained about bad voice quality on several calls,
  so I programmed a button on each phone which users can hit if they have
  audio drops and echo.
 
  I did this to check if there is a common recurrent problem to a given
  destination or just for one user etc... But till now I could not detect
  a pattern which could explain the problems
 
  This alert button is pressed between 7%-10% of all calls. The customer
  has 25 phones and around 300 calls per day.
 
  The SNOM phones are connected to Linksys switches and are totaly split
  from the computers network. The same goes for the asterisk box. No calls
  are routed trough the internet.
  Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier
 
  The carrier we use is known for his good quality and we never had a
  problem. It is the historic and most expensive carrier in Luxembourg.
 
  Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
  maximum of 6 concurrent calls.
 
 
 Which version of asterisk/zaptel, and which echo canceler is running in 
 Zaptel?
 
 Regards,
 Steve
 
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-- 
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Call quality

I considered that,
but I fear that this would load the machine even more. So I guess I
should take a more powerful box with a good harddrive (at the moment I
have a solid state flash card) and start recording calls.


Best regards,
Loic Didelot.



On Tue, 2008-07-01 at 09:10 -0400, David Backeberg wrote:
 On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote:
  Maybe someone can help me to track down the problem. What should I
  check, monitor test. Any ideas are welcome.
 
 If there are no legal reasons not to, consider recording all calls for
 a limited time. It's easier for engineers to debug a voice quality
 problem when they have a recording of exactly what it sounds like.
 It's possible that different people are complaining about different
 perceptions of what they consider a voice quality problem, and that
 the problem might not even be on your end of the conversation.
 
 ___
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 Register Now: http://www.astricon.net
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Call quality

Recording the calls may or may not reveal an issue.  I have personally
done this exact same method of troubleshooting only to find the
recordings were perfect but not the actual calls.

I think you should try just putting a regular server in place of your
appliance and then test.

I have a feeling the I/O is choking your system, similar to recording
many simultaneous calls, which to me would indicate a flash
bottleneck.  At least put in a real HD and copy over your configs.

Thanks,
Steve T

On Tue, Jul 1, 2008 at 9:15 AM, Loic Didelot [EMAIL PROTECTED] wrote:
 I considered that,
 but I fear that this would load the machine even more. So I guess I
 should take a more powerful box with a good harddrive (at the moment I
 have a solid state flash card) and start recording calls.


 Best regards,
 Loic Didelot.



 On Tue, 2008-07-01 at 09:10 -0400, David Backeberg wrote:
 On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote:
  Maybe someone can help me to track down the problem. What should I
  check, monitor test. Any ideas are welcome.

 If there are no legal reasons not to, consider recording all calls for
 a limited time. It's easier for engineers to debug a voice quality
 problem when they have a recording of exactly what it sounds like.
 It's possible that different people are complaining about different
 perceptions of what they consider a voice quality problem, and that
 the problem might not even be on your end of the conversation.

 ___
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 Register Now: http://www.astricon.net

 asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
 --
 Loïc DIDELOT
 MIXvoip S.a.
 [EMAIL PROTECTED]
 http://www.mixvoip.com


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Re: [asterisk-users] Call quality

2008-07-01 Thread Tzafrir Cohen

On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
 Hello,
 one of my customers complained about bad voice quality on several calls,
 so I programmed a button on each phone which users can hit if they have
 audio drops and echo.
 
 I did this to check if there is a common recurrent problem to a given
 destination or just for one user etc... But till now I could not detect
 a pattern which could explain the problems
 
 This alert button is pressed between 7%-10% of all calls. The customer
 has 25 phones and around 300 calls per day.
 
 The SNOM phones are connected to Linksys switches and are totaly split
 from the computers network. The same goes for the asterisk box. No calls
 are routed trough the internet.
 Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier

Are the problems in SIP-PSTN calls? SIP-SIP calls?
PSTN-Local? (echo test, playback, whatever)

SIP-PSTN or PSTN-SIP (what direction is the call)?

7% is something you have hope of reproducing. Unless you miss the real
factor. Have you managed to reproduce it yourself?

 
 The carrier we use is known for his good quality and we never had a
 problem. It is the historic and most expensive carrier in Luxembourg.
 
 Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
 maximum of 6 concurrent calls.
 
 Maybe someone can help me to track down the problem. What should I
 check, monitor test. Any ideas are welcome.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Call quality

The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
of all alerts are internal calls. I had the chance to notice the problem
once myself but I could never again reproduce.  

Best regards,
Loic Didelot.

On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote:
 On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
  Hello,
  one of my customers complained about bad voice quality on several calls,
  so I programmed a button on each phone which users can hit if they have
  audio drops and echo.
  
  I did this to check if there is a common recurrent problem to a given
  destination or just for one user etc... But till now I could not detect
  a pattern which could explain the problems
  
  This alert button is pressed between 7%-10% of all calls. The customer
  has 25 phones and around 300 calls per day.
  
  The SNOM phones are connected to Linksys switches and are totaly split
  from the computers network. The same goes for the asterisk box. No calls
  are routed trough the internet.
  Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier
 
 Are the problems in SIP-PSTN calls? SIP-SIP calls?
 PSTN-Local? (echo test, playback, whatever)
 
 SIP-PSTN or PSTN-SIP (what direction is the call)?
 
 7% is something you have hope of reproducing. Unless you miss the real
 factor. Have you managed to reproduce it yourself?
 
  
  The carrier we use is known for his good quality and we never had a
  problem. It is the historic and most expensive carrier in Luxembourg.
  
  Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
  maximum of 6 concurrent calls.
  
  Maybe someone can help me to track down the problem. What should I
  check, monitor test. Any ideas are welcome.
 
-- 
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] Call quality

Try IOSTAT
http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat

Maybe you can correlate VM and/or emailing of VM to your IO spikes.

Have you watched top and the Asterisk CLI when someone hits the panic button?

Thanks,
Steve T

On Tue, Jul 1, 2008 at 11:17 AM, Loic Didelot [EMAIL PROTECTED] wrote:
The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
of all alerts are internal calls. I had the chance to notice the problem
once myself but I could never again reproduce.

Best regards,
Loic Didelot.

On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote:
On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
Hello,
one of my customers complained about bad voice quality on several calls,
so I programmed a button on each phone which users can hit if they have
audio drops and echo.

I did this to check if there is a common recurrent problem to a given
destination or just for one user etc... But till now I could not detect
a pattern which could explain the problems

This alert button is pressed between 7%-10% of all calls. The customer
has 25 phones and around 300 calls per day.

The SNOM phones are connected to Linksys switches and are totaly split
from the computers network. The same goes for the asterisk box. No calls
are routed trough the internet.
Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier

Are the problems in SIP-PSTN calls? SIP-SIP calls?
PSTN-Local? (echo test, playback, whatever)

SIP-PSTN or PSTN-SIP (what direction is the call)?

7% is something you have hope of reproducing. Unless you miss the real
factor. Have you managed to reproduce it yourself?

The carrier we use is known for his good quality and we never had a
problem. It is the historic and most expensive carrier in Luxembourg.

Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a
maximum of 6 concurrent calls.

Maybe someone can help me to track down the problem. What should I
check, monitor test. Any ideas are welcome.

--
Loïc DIDELOT
MIXvoip S.a.
[EMAIL PROTECTED]
http://www.mixvoip.com

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Re: [asterisk-users] Call quality

2008-07-01 Thread Tzafrir Cohen

On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
 The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
 of all alerts are internal calls. 

Any chance that omst of the calls are outgoing SIP-PSTN calls?

 I had the chance to notice the problem
 once myself but I could never again reproduce.  

So it doesn't happen with Local-PSTN calls (the type you can easily
test remotely if we assume there's no voip access).

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Call quality

Yes, most calls are SIP-PSTN calls.

Thanks for your help.

I will try a faster box. Are VIA CPUs known to cause problems?

Loic


On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote:
 On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
  The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
  of all alerts are internal calls. 
 
 Any chance that omst of the calls are outgoing SIP-PSTN calls?
 
  I had the chance to notice the problem
  once myself but I could never again reproduce.  
 
 So it doesn't happen with Local-PSTN calls (the type you can easily
 test remotely if we assume there's no voip access).
 


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Re: [asterisk-users] Call quality

I don't think your issue is the VIA CPU but the I/O of your flash
drive.  Voicemail is what I suspect being the I/O bottleneck.

Thanks,
Steve T

On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot [EMAIL PROTECTED] wrote:
 Yes, most calls are SIP-PSTN calls.

 Thanks for your help.

 I will try a faster box. Are VIA CPUs known to cause problems?

 Loic


 On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote:
 On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
  The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
  of all alerts are internal calls.

 Any chance that omst of the calls are outgoing SIP-PSTN calls?

  I had the chance to notice the problem
  once myself but I could never again reproduce.

 So it doesn't happen with Local-PSTN calls (the type you can easily
 test remotely if we assume there's no voip access).



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Re: [asterisk-users] Call quality

Yes,
but they get like 10 voicemails per day. That feature isnt really used
alot.

Loic

On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote:
 I don't think your issue is the VIA CPU but the I/O of your flash
 drive.  Voicemail is what I suspect being the I/O bottleneck.
 
 Thanks,
 Steve T
 
 On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot [EMAIL PROTECTED] wrote:
  Yes, most calls are SIP-PSTN calls.
 
  Thanks for your help.
 
  I will try a faster box. Are VIA CPUs known to cause problems?
 
  Loic
 
 
  On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote:
  On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
   The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
   of all alerts are internal calls.
 
  Any chance that omst of the calls are outgoing SIP-PSTN calls?
 
   I had the chance to notice the problem
   once myself but I could never again reproduce.
 
  So it doesn't happen with Local-PSTN calls (the type you can easily
  test remotely if we assume there's no voip access).
 
 
 
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Re: [asterisk-users] Call quality

Run top along with the tool that indicated the high I/O and see what
is going on.  Are you doing G729 or anything like that?

Thanks,
Steve T

On Tue, Jul 1, 2008 at 3:06 PM, Loic Didelot [EMAIL PROTECTED] wrote:
 Yes,
 but they get like 10 voicemails per day. That feature isnt really used
 alot.

 Loic

 On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote:
 I don't think your issue is the VIA CPU but the I/O of your flash
 drive.  Voicemail is what I suspect being the I/O bottleneck.

 Thanks,
 Steve T

 On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot [EMAIL PROTECTED] wrote:
  Yes, most calls are SIP-PSTN calls.
 
  Thanks for your help.
 
  I will try a faster box. Are VIA CPUs known to cause problems?
 
  Loic
 
 
  On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote:
  On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
   The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
   of all alerts are internal calls.
 
  Any chance that omst of the calls are outgoing SIP-PSTN calls?
 
   I had the chance to notice the problem
   once myself but I could never again reproduce.
 
  So it doesn't happen with Local-PSTN calls (the type you can easily
  test remotely if we assume there's no voip access).
 
 
 
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Re: [asterisk-users] Call quality

2008-07-01 Thread Tzafrir Cohen

On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote:
 Run top along with the tool that indicated the high I/O and see what
 is going on.  Are you doing G729 or anything like that?

vmstat will probably provide more useful data (vmstat 1 etc. for a
continous run).

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Call quality

2008-07-01 Thread Gavin Henry

What did you do to setup a button for alerts?

Thanks.

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Re: [asterisk-users] Call quality

Depends on the phone.

On many devices you can setup buttons to call a url. Thats what I did.


Loic

On Tue, 2008-07-01 at 21:19 +0100, Gavin Henry wrote:
 What did you do to setup a button for alerts?
 
 Thanks.
 
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Re: [asterisk-users] Call quality

Hi,
I am using g711a everywhere.

I checked on a completely idle system (no calls at all) and idle CPU is
dropping from 100% to 0% more than once per minute.

procs ---memory-- ---swap-- -io -system-- cpu
 r  b   swpd   free   buff  cache   si   sobibo   in   cs us sy id wa
 1  0  0 891124   4644  4286800 028 4047 85757  0 97  3  0
 0  0  0 891124   4644  4287600 0 0 4042 68342  0 94  6  0
 0  0  0 891124   4644  4287600 0 0 4042 72429  0 97  3  0
 0  0  0 891124   4644  4287600 0 0 4065 158878  0 100  0  0
 0  0  0 891124   4644  4287600 0 0 4033 59033  0 98  2  0
 0  0  0 891124   4644  4287600 0 0 4012 14464  0 96  4  0
 0  0  0 891124   4652  4286800 076 4013 19727  0 37 62  1
 0  0  0 891124   4652  4287600 0 0 4011 20225  0  4 96  0
 0  0  0 891124   4652  4287600 0 0 4011 23901  0 20 80  0
 0  1  0 891124   4652  4287600 0 4 4025 21165  0 40 55  5
 0  0  0 891124   4660  4287600 032 4028 20190  0  1 95  4
 0  0  0 891124   4660  4287600 0 0 4022 23295  0  0 100  0
 0  0  0 891124   4660  4287600 0 0 4111 20508  0  0 100  0
 0  0  0 891124   4660  4287600 0 0 4102 25239  0 30 70  0
 0  0  0 891124   4660  4287600 0 0 4112 23148  0  0 100  0
 0  0  0 891124   4668  4286800 052 4116 19031  0  0 100  0
 1  0  0 891124   4668  4287600 0 0 4110 21776  0  0 100  0
 0  0  0 891124   4668  4287600 0 0 4150 20332  0  0 100  0
 0  0  0 891124   4668  4287600 0 0 4114 26285  0  0 100  0
 0  0  0 891124   4668  4287600 032 4118 23029  1  0 99  0
 0  0  0 891124   4668  4287600 0 0 4121 23284  0  0 100  0
 0  0  0 891124   4676  4286800 060 4112 25232  0 36 64  0
 0  0  0 891124   4676  4287600 0 0 4134 21583  0 99  1  0
 0  0  0 891124   4676  4287600 0 0 4105 26029  0 100  0  0
 0  0  0 891124   4676  4287600 076 4143 22795  0 25 75  0
 0  0  0 891124   4676  4287600 0 0 4118 21418  0  0 54 46
 0  0  0 891124   4676  4287600 0 0 4108 25499  0  0 100  0
 0  0  0 891124   4684  4286800 052 4081 20778  0  0 100  0
 0  0  0 891124   4684  4287600 0 0 4011 25463  0 13 87  0
 0  0  0 891124   4684  4287600 0 0 4021 23502  0 86 14  0
 0  0  0 891124   4684  4287600 0 0 4015 21693  0  1 99  0


On Tue, 2008-07-01 at 22:28 +0300, Tzafrir Cohen wrote:
 On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote:
  Run top along with the tool that indicated the high I/O and see what
  is going on.  Are you doing G729 or anything like that?
 
 vmstat will probably provide more useful data (vmstat 1 etc. for a
 continous run).
 


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Re: [asterisk-users] Call quality