VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 7:38 am Post subject: [asterisk-users] Call quality |
|
|
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.
--
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com |
|
Back to top |
|
|
davies147 at gmail.com Guest
|
Posted: Tue Jul 01, 2008 7:54 am Post subject: [asterisk-users] Call quality |
|
|
2008/7/1 Loic Didelot <ldidelot at mixvoip.com>:
Quote: | 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 |
|
Back to top |
|
|
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 7:57 am Post subject: [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.
root at ppsite1:~# mpstat 1
Linux 2.6.22-14-server (ppsite1) 07/01/2008
02:54:40 PM CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s
02:54:41 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4210.00
02:54:42 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4207.00
02:54:43 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4208.00
02:54:44 PM all 0.00 0.00 0.00 0.00 45.00 0.00 0.00 55.00 4127.00
02:54:45 PM all 0.00 0.00 0.00 0.00 97.00 0.00 0.00 3.00 4148.00
02:54:46 PM all 0.00 0.00 0.00 0.00 93.00 4.00 0.00 3.00 4195.00
02:54:47 PM all 0.00 0.00 0.00 0.00 92.00 6.00 0.00 2.00 4175.00
02:54:48 PM all 0.00 0.00 0.00 1.00 91.00 2.00 0.00 6.00 4154.00
02:54:49 PM all 0.00 0.00 0.00 0.00 100.00 0.00 0.00 0.00 4069.00
02:54:50 PM all 0.00 0.00 0.00 0.00 23.00 0.00 0.00 77.00 4125.00
02:54:51 PM all 2.00 0.00 0.00 0.00 19.00 0.00 0.00 79.00 4123.00
02:54:52 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4236.00
02:54:53 PM all 0.00 0.00 6.00 2.00 0.00 3.00 0.00 89.00 4302.00
02:54:54 PM all 0.00 0.00 5.00 0.00 0.00 3.00 0.00 92.00 4267.00
02:54:55 PM all 0.00 0.00 20.00 0.00 0.00 1.00 0.00 79.00 4328.00
02:54:56 PM all 0.00 0.00 0.00 0.00 9.00 0.00 0.00 91.00 4352.00
02:54:57 PM all 0.00 0.00 0.00 49.00 46.00 0.00 0.00 5.00 4376.00
02:54:58 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4350.00
02:54:59 PM all 11.00 0.00 2.00 36.00 0.00 0.00 0.00 51.00 4237.00
02:55:00 PM all 0.00 0.00 0.00 100.00 0.00 0.00 0.00 0.00 4221.00
02:55:01 PM all 1.00 0.00 1.00 62.00 36.00 0.00 0.00 0.00 4318.00
02:55:02 PM all 0.00 0.00 0.00 2.00 98.00 0.00 0.00 0.00 4219.00
02:55:03 PM all 0.00 0.00 0.00 0.00 100.00 0.00 0.00 0.00 4342.00
02:55:04 PM all 0.00 0.00 0.00 2.00 98.00 0.00 0.00 0.00 4236.00
02:55:05 PM all 14.00 0.00 4.00 20.00 62.00 0.00 0.00 0.00 4229.00
02:55:06 PM all 39.00 0.00 3.00 38.00 18.00 1.00 0.00 1.00 4346.00
02:55:07 PM all 8.00 0.00 8.00 79.00 3.00 1.00 0.00 1.00 4240.00
02:55:08 PM all 1.00 0.00 0.00 98.00 0.00 0.00 0.00 1.00 4217.00
02:55:09 PM all 0.00 0.00 1.00 6.00 0.00 0.00 0.00 93.00 4167.00
02:55:10 PM all 0.00 0.00 0.00 0.00 25.00 0.00 0.00 75.00 4132.00
02:55:11 PM all 0.00 0.00 0.00 0.00 75.00 0.00 0.00 25.00 4117.00
02:55:12 PM all 0.00 0.00 0.00 0.00 53.00 0.00 0.00 47.00 4130.00
02:55:13 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4103.00
02:55:14 PM all 0.00 0.00 0.00 50.00 0.00 0.00 0.00 50.00 4124.00
02:55:15 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4216.00
02:55:16 PM all 1.00 0.00 0.00 0.00 32.00 0.00 0.00 67.00 4214.00
02:55:17 PM all 0.00 0.00 0.00 0.00 98.00 0.00 0.00 2.00 4209.00
02:55:18 PM all 0.00 0.00 0.00 0.00 94.00 0.00 0.00 6.00 4220.00
02:55:19 PM all 0.00 0.00 0.00 0.00 58.00 0.00 0.00 42.00 4216.00
02:55:20 PM all 1.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 4204.00
02:55:21 PM all 1.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 4210.00
02:55:22 PM all 1.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 4234.00
02:55:23 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4202.00
02:55:24 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4109.00
02:55:25 PM all 1.00 0.00 1.00 1.00 53.00 1.00 0.00 43.00 4179.00
02:55:26 PM all 0.00 0.00 0.00 0.00 35.00 0.00 0.00 65.00 4213.00
02:55:27 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4204.00
02:55:28 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4169.00
02:55:29 PM all 0.00 0.00 3.96 0.00 37.62 0.99 0.00 57.43 4149.50
02:55:30 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4208.00
02:55:31 PM all 0.00 0.00 0.00 16.00 3.00 0.00 0.00 81.00 4117.00
02:55:31 PM CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s
02:55:32 PM all 0.00 0.00 4.00 0.00 4.00 0.00 0.00 92.00 4185.00
02:55:33 PM all 18.00 0.00 10.00 0.00 0.00 0.00 0.00 72.00 4241.00
02:55:34 PM all 8.00 0.00 3.00 29.00 0.00 0.00 0.00 60.00 4279.00
02:55:35 PM all 0.00 0.00 0.00 100.00 0.00 0.00 0.00 0.00 4274.00
02:55:36 PM all 0.00 0.00 3.00 79.00 16.00 2.00 0.00 0.00 4235.00
02:55:37 PM all 0.00 0.00 0.00 11.00 89.00 0.00 0.00 0.00 4237.00
02:55:38 PM all 0.00 0.00 0.00 20.00 78.00 2.00 0.00 0.00 4165.00
02:55:39 PM all 0.00 0.00 0.00 7.00 93.00 0.00 0.00 0.00 4103.00
02:55:40 PM all 1.00 0.00 2.00 6.00 69.00 0.00 0.00 22.00 4147.00
02:55:41 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4023.00
02:55:42 PM all 0.00 0.00 3.00 0.00 12.00 0.00 0.00 85.00 4086.00
02:55:43 PM all 0.00 0.00 0.00 0.00 14.00 0.00 0.00 86.00
On Tue, 2008-07-01 at 14:38 +0200, Loic Didelot wrote:
Quote: | 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.
--
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com |
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Tue Jul 01, 2008 7:58 am Post subject: [asterisk-users] Call quality |
|
|
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <ldidelot at mixvoip.com> wrote:
Quote: | 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.
--
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
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 |
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Tue Jul 01, 2008 8:02 am Post subject: [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 <ldidelot at mixvoip.com> wrote:
Quote: | 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.
root at ppsite1:~# mpstat 1
Linux 2.6.22-14-server (ppsite1) 07/01/2008
02:54:40 PM CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s
02:54:41 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4210.00
02:54:42 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4207.00
02:54:43 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4208.00
02:54:44 PM all 0.00 0.00 0.00 0.00 45.00 0.00 0.00 55.00 4127.00
02:54:45 PM all 0.00 0.00 0.00 0.00 97.00 0.00 0.00 3.00 4148.00
02:54:46 PM all 0.00 0.00 0.00 0.00 93.00 4.00 0.00 3.00 4195.00
02:54:47 PM all 0.00 0.00 0.00 0.00 92.00 6.00 0.00 2.00 4175.00
02:54:48 PM all 0.00 0.00 0.00 1.00 91.00 2.00 0.00 6.00 4154.00
02:54:49 PM all 0.00 0.00 0.00 0.00 100.00 0.00 0.00 0.00 4069.00
02:54:50 PM all 0.00 0.00 0.00 0.00 23.00 0.00 0.00 77.00 4125.00
02:54:51 PM all 2.00 0.00 0.00 0.00 19.00 0.00 0.00 79.00 4123.00
02:54:52 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4236.00
02:54:53 PM all 0.00 0.00 6.00 2.00 0.00 3.00 0.00 89.00 4302.00
02:54:54 PM all 0.00 0.00 5.00 0.00 0.00 3.00 0.00 92.00 4267.00
02:54:55 PM all 0.00 0.00 20.00 0.00 0.00 1.00 0.00 79.00 4328.00
02:54:56 PM all 0.00 0.00 0.00 0.00 9.00 0.00 0.00 91.00 4352.00
02:54:57 PM all 0.00 0.00 0.00 49.00 46.00 0.00 0.00 5.00 4376.00
02:54:58 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4350.00
02:54:59 PM all 11.00 0.00 2.00 36.00 0.00 0.00 0.00 51.00 4237.00
02:55:00 PM all 0.00 0.00 0.00 100.00 0.00 0.00 0.00 0.00 4221.00
02:55:01 PM all 1.00 0.00 1.00 62.00 36.00 0.00 0.00 0.00 4318.00
02:55:02 PM all 0.00 0.00 0.00 2.00 98.00 0.00 0.00 0.00 4219.00
02:55:03 PM all 0.00 0.00 0.00 0.00 100.00 0.00 0.00 0.00 4342.00
02:55:04 PM all 0.00 0.00 0.00 2.00 98.00 0.00 0.00 0.00 4236.00
02:55:05 PM all 14.00 0.00 4.00 20.00 62.00 0.00 0.00 0.00 4229.00
02:55:06 PM all 39.00 0.00 3.00 38.00 18.00 1.00 0.00 1.00 4346.00
02:55:07 PM all 8.00 0.00 8.00 79.00 3.00 1.00 0.00 1.00 4240.00
02:55:08 PM all 1.00 0.00 0.00 98.00 0.00 0.00 0.00 1.00 4217.00
02:55:09 PM all 0.00 0.00 1.00 6.00 0.00 0.00 0.00 93.00 4167.00
02:55:10 PM all 0.00 0.00 0.00 0.00 25.00 0.00 0.00 75.00 4132.00
02:55:11 PM all 0.00 0.00 0.00 0.00 75.00 0.00 0.00 25.00 4117.00
02:55:12 PM all 0.00 0.00 0.00 0.00 53.00 0.00 0.00 47.00 4130.00
02:55:13 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4103.00
02:55:14 PM all 0.00 0.00 0.00 50.00 0.00 0.00 0.00 50.00 4124.00
02:55:15 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4216.00
02:55:16 PM all 1.00 0.00 0.00 0.00 32.00 0.00 0.00 67.00 4214.00
02:55:17 PM all 0.00 0.00 0.00 0.00 98.00 0.00 0.00 2.00 4209.00
02:55:18 PM all 0.00 0.00 0.00 0.00 94.00 0.00 0.00 6.00 4220.00
02:55:19 PM all 0.00 0.00 0.00 0.00 58.00 0.00 0.00 42.00 4216.00
02:55:20 PM all 1.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 4204.00
02:55:21 PM all 1.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 4210.00
02:55:22 PM all 1.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 4234.00
02:55:23 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4202.00
02:55:24 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4109.00
02:55:25 PM all 1.00 0.00 1.00 1.00 53.00 1.00 0.00 43.00 4179.00
02:55:26 PM all 0.00 0.00 0.00 0.00 35.00 0.00 0.00 65.00 4213.00
02:55:27 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4204.00
02:55:28 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4169.00
02:55:29 PM all 0.00 0.00 3.96 0.00 37.62 0.99 0.00 57.43 4149.50
02:55:30 PM all 0.00 0.00 1.00 0.00 0.00 0.00 0.00 99.00 4208.00
02:55:31 PM all 0.00 0.00 0.00 16.00 3.00 0.00 0.00 81.00 4117.00
02:55:31 PM CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s
02:55:32 PM all 0.00 0.00 4.00 0.00 4.00 0.00 0.00 92.00 4185.00
02:55:33 PM all 18.00 0.00 10.00 0.00 0.00 0.00 0.00 72.00 4241.00
02:55:34 PM all 8.00 0.00 3.00 29.00 0.00 0.00 0.00 60.00 4279.00
02:55:35 PM all 0.00 0.00 0.00 100.00 0.00 0.00 0.00 0.00 4274.00
02:55:36 PM all 0.00 0.00 3.00 79.00 16.00 2.00 0.00 0.00 4235.00
02:55:37 PM all 0.00 0.00 0.00 11.00 89.00 0.00 0.00 0.00 4237.00
02:55:38 PM all 0.00 0.00 0.00 20.00 78.00 2.00 0.00 0.00 4165.00
02:55:39 PM all 0.00 0.00 0.00 7.00 93.00 0.00 0.00 0.00 4103.00
02:55:40 PM all 1.00 0.00 2.00 6.00 69.00 0.00 0.00 22.00 4147.00
02:55:41 PM all 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 4023.00
02:55:42 PM all 0.00 0.00 3.00 0.00 12.00 0.00 0.00 85.00 4086.00
02:55:43 PM all 0.00 0.00 0.00 0.00 14.00 0.00 0.00 86.00
On Tue, 2008-07-01 at 14:38 +0200, Loic Didelot wrote:
Quote: | 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.
--
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
|
Back to top |
|
|
dbackeberg at gmail.com Guest
|
Posted: Tue Jul 01, 2008 8:10 am Post subject: [asterisk-users] Call quality |
|
|
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <ldidelot at mixvoip.com> wrote:
Quote: | 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. |
|
Back to top |
|
|
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 8:11 am Post subject: [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:
Quote: | On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <ldidelot at mixvoip.com> wrote:
Quote: | 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.
--
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
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
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com |
|
Back to top |
|
|
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 8:14 am Post subject: [asterisk-users] Call quality |
|
|
Hello,
I forgot to include CPU information
root at ppsite1:/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:
Quote: | 2008/7/1 Loic Didelot <ldidelot at mixvoip.com>:
Quote: | 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
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com |
|
Back to top |
|
|
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 8:15 am Post subject: [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:
Quote: | On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <ldidelot at mixvoip.com> wrote:
Quote: | 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.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com |
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Tue Jul 01, 2008 9:39 am Post subject: [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 <ldidelot at mixvoip.com> wrote:
Quote: | 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:
Quote: | On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <ldidelot at mixvoip.com> wrote:
Quote: | 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.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
|
Back to top |
|
|
tzafrir.cohen at xorco... Guest
|
Posted: Tue Jul 01, 2008 10:03 am Post subject: [asterisk-users] Call quality |
|
|
On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
Quote: | 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?
Quote: |
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:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
|
Back to top |
|
|
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 10:17 am Post subject: [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:
Quote: | On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
Quote: | 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?
Quote: |
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.
|
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com |
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Tue Jul 01, 2008 10:37 am Post subject: [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 <ldidelot at mixvoip.com> wrote:
Quote: | 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:
Quote: | On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
Quote: | 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?
Quote: |
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.
|
| --
Lo?c DIDELOT
MIXvoip S.a.
ldidelot at mixvoip.com
http://www.mixvoip.com
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
|
Back to top |
|
|
tzafrir.cohen at xorco... Guest
|
Posted: Tue Jul 01, 2008 10:50 am Post subject: [asterisk-users] Call quality |
|
|
On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
Quote: | 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?
Quote: | 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:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
|
Back to top |
|
|
ldidelot at mixvoip.com Guest
|
Posted: Tue Jul 01, 2008 1:52 pm Post subject: [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:
Quote: | On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
Quote: | 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?
Quote: | 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).
|
|
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|