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cursor at telecomabmex...
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 Posted: Tue Apr 05, 2016 3:12 pm Post subject: [asterisk-users] Best timing source? |
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I am currently having a voice quality problem with one of our
Asterisk servers. We have checked the network and we have found no
problems that could cause the voice to sound cracked and with small
interruptions. I am looking at the timing source for Asterisk and it is
currently using timerfd even though we have an E1 card installed. Is
timerfd better than dahdi? Any recommendations to test if timing may be
a problem for voice quality and DTMF?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
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jcolp at digium.com
Guest
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| Posted: Tue Apr 05, 2016 3:17 pm Post subject: [asterisk-users] Best timing source? |
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Carlos Chavez wrote:
| Quote: | I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF?
|
What is the scenario and the channels involved? Timing is only used for
things such as playback, music on hold, and ConfBridge. If it's strictly
a two party call then Asterisk forwards media as received.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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cursor at telecomabmex...
Guest
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| Posted: Tue Apr 05, 2016 4:23 pm Post subject: [asterisk-users] Best timing source? |
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On 4/5/16 3:17 PM, Joshua Colp wrote:
| Quote: | Carlos Chavez wrote:
| Quote: | I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF?
|
What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received.
|
The problem appears on all calls, no matter the source or
destination. There are desk phones, softphones and a couple SIP trunks
to another office. They all experience the problem. Calls between
extensions, from or to the E1, from or to trunks. The only scenario
left to try is connecting calls only via the E1 so we completely
eliminate the network side of things and se if we get the same
behaviour. During calls you can hear some background noice and
interruptions in the voice. DTMF fails when we try to dial to external
IVR.
I do not really believe that the fault is in the Asterisk server
but I have to eliminate all posibilities on my side before I can lay
blame on the network infrastructure. I was also just wondering if DAHDI
would not be a better timing source for Asterisk since it is hardware based?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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ngom at numericap.com
Guest
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| Posted: Tue Apr 05, 2016 4:28 pm Post subject: [asterisk-users] Best timing source? |
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Hello,
I am doing a configuration for connecting my server asterisk to a SIP provider. I ask if somebody can give me a basic code or a link to begin well;
Thanks !!!! | Quote: |
Le 5 avril 2016 à 23:23, Carlos Chavez <cursor@telecomabmex.com> a écrit :
On 4/5/16 3:17 PM, Joshua Colp wrote: | Quote: |
Carlos Chavez wrote: |
| Quote: | | Quote: | I am currently having a voice quality problem with one of our Asterisk>> servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF? |
What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received. |
The problem appears on all calls, no matter the source or destination. There are desk phones, softphones and a couple SIP trunks
to another office. They all experience the problem. Calls between
extensions, from or to the E1, from or to trunks. The only scenario
left to try is connecting calls only via the E1 so we completely
eliminate the network side of things and se if we get the same
behaviour. During calls you can hear some background noice and
interruptions in the voice. DTMF fails when we try to dial to external
IVR.
I do not really believe that the fault is in the Asterisk server
but I have to eliminate all posibilities on my side before I can lay
blame on the network infrastructure. I was also just wondering if DAHDI
would not be a better timing source for Asterisk since it is hardware based?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
Mamadou NGOM
Ingénieur Télécommunications & Réseaux
Mobile: 06 72 45 23 03
Skype: Mamadou Numericap
NumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015.
siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: finance@numericap.com ([email]mail%3Afinance@numericap.com[/email])
Centre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 |
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jcolp at digium.com
Guest
|
| Posted: Tue Apr 05, 2016 4:36 pm Post subject: [asterisk-users] Best timing source? |
|
|
Carlos Chavez wrote:
| Quote: | On 4/5/16 3:17 PM, Joshua Colp wrote:
| Quote: | Carlos Chavez wrote:
| Quote: | I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF?
|
What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received.
|
The problem appears on all calls, no matter the source or destination.
There are desk phones, softphones and a couple SIP trunks to another
office. They all experience the problem. Calls between extensions, from
or to the E1, from or to trunks. The only scenario left to try is
connecting calls only via the E1 so we completely eliminate the network
side of things and se if we get the same behaviour. During calls you can
hear some background noice and interruptions in the voice. DTMF fails
when we try to dial to external IVR.
I do not really believe that the fault is in the Asterisk server but I
have to eliminate all posibilities on my side before I can lay blame on
the network infrastructure. I was also just wondering if DAHDI would not
be a better timing source for Asterisk since it is hardware based?
|
Either timerfd or dahdi should perform the same. I wouldn't expect dahdi
to improve things unless something was really wrong on the system itself.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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asterisk.org at sedwar...
Guest
|
| Posted: Tue Apr 05, 2016 4:37 pm Post subject: [asterisk-users] Best timing source? |
|
|
On Tue, 5 Apr 2016, Mamadou NGOM wrote:
| Quote: | I am doing a configuration for connecting my server asterisk to a SIP
provider. I ask if somebody can give me a basic code or a link to begin
well;
|
1) You should start a fresh thread rather than hijacking an existing
unrelated thread.
2) You should show us what you have done so far.
Most SIP providers have sample snippets for your sip.conf and
extensions.conf files.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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asterisk.org at sedwar...
Guest
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asterisk_list at earth...
Guest
|
| Posted: Wed Apr 06, 2016 2:46 am Post subject: [asterisk-users] Best timing source? |
|
|
On Tuesday 05 Apr 2016, Mamadou NGOM wrote:
| Quote: | Hello,
I am doing a configuration for connecting my server asterisk to a SIP
provider. I ask if somebody can give me a basic code or a link to begin
well; Thanks !!!!
|
Rule One: Start your own topics -- don't jump in on someone else's, unless
it's actually relevant.
Rule Two: Type your reply *after* the thing you are replying to.
Your SIP trunk provider will know how to connect an Asterisk box to them.
That, after all, is what most people are going to be connecting.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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cursor at telecomabmex...
Guest
|
| Posted: Wed Apr 06, 2016 1:02 pm Post subject: [asterisk-users] Best timing source? |
|
|
On 4/5/16 3:17 PM, Joshua Colp wrote:
| Quote: | Carlos Chavez wrote:
| Quote: | I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF?
|
What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received.
|
I am starting to think that the problem may be in the server hardware
itself. We are using a Dell R220 with 8gb of ram and 2 hard disks in a
Raid 1 configuration (Linux Raid). We are using CentOS 7. We had to
remove the raid card from the server to install an E1 card (the raid
card was Windows only so no loss there really). Internally everything
sounds good (from E1 to a conference or music) but once you hit a
network interface we start getting pops and drops.
Anyone with this server and Asterisk ever had issues like these?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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duncan at e-simple.co.nz
Guest
|
| Posted: Wed Apr 06, 2016 2:40 pm Post subject: [asterisk-users] Best timing source? |
|
|
| Quote: | On 7/04/2016, at 6:01 AM, Carlos Chavez <cursor@telecomabmex.com> wrote:
| Quote: | On 4/5/16 3:17 PM, Joshua Colp wrote:
Carlos Chavez wrote:
| Quote: | I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF?
|
What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received.
|
I am starting to think that the problem may be in the server hardware itself. We are using a Dell R220 with 8gb of ram and 2 hard disks in a Raid 1 configuration (Linux Raid). We are using CentOS 7. We had to remove the raid card from the server to install an E1 card (the raid card was Windows only so no loss there really). Internally everything sounds good (from E1 to a conference or music) but once you hit a network interface we start getting pops and drops.
Anyone with this server and Asterisk ever had issues like these?
|
Just checking you have your E1 timing set to slave off the upstream. If not you are going to have E1 sync errors which will give you the voice problems you describe
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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cursor at telecomabmex...
Guest
|
| Posted: Wed Apr 06, 2016 4:01 pm Post subject: [asterisk-users] Best timing source? |
|
|
On 4/6/16 2:39 PM, Duncan Turnbull wrote:
| Quote: |
| Quote: | On 7/04/2016, at 6:01 AM, Carlos Chavez <cursor@telecomabmex.com> wrote:
| Quote: | On 4/5/16 3:17 PM, Joshua Colp wrote:
Carlos Chavez wrote:
| Quote: | I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking at the timing source for Asterisk and it is currently using
timerfd even though we have an E1 card installed. Is timerfd better than
dahdi? Any recommendations to test if timing may be a problem for voice
quality and DTMF?
|
What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received.
|
I am starting to think that the problem may be in the server hardware itself. We are using a Dell R220 with 8gb of ram and 2 hard disks in a Raid 1 configuration (Linux Raid). We are using CentOS 7. We had to remove the raid card from the server to install an E1 card (the raid card was Windows only so no loss there really). Internally everything sounds good (from E1 to a conference or music) but once you hit a network interface we start getting pops and drops.
Anyone with this server and Asterisk ever had issues like these?
|
Just checking you have your E1 timing set to slave off the upstream. If not you are going to have E1 sync errors which will give you the voice problems you describe
|
Dahdi_test gives me a 99.97% average. The problem is present on
all calls (except calling into the E1 to a conference or to MoH). I am
preparing a new server to see if it is a hardware issue.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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duncan at e-simple.co.nz
Guest
|
| Posted: Wed Apr 06, 2016 4:21 pm Post subject: [asterisk-users] Best timing source? |
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|
On 07/04/16 09:00, Carlos Chavez wrote:
| Quote: | On 4/6/16 2:39 PM, Duncan Turnbull wrote:
| Quote: |
| Quote: |    I am starting to think that the problem may be in the server hardware itself. We are using a Dell R220 with 8gb of ram and 2 hard disks in a Raid 1 configuration (Linux Raid). We are using CentOS 7. We had to remove the raid card from the server to install an E1 card (the raid card was Windows only so no loss there really). Internally everything sounds good (from E1 to a conference or music) but once you hit a network interface we start getting pops and drops.
   Anyone with this server and Asterisk ever had issues like these?
|
Just checking you have your E1 timing set to slave off the upstream. If not you are going to have E1 sync errors which will give you the voice problems you describe
|
   Dahdi_test gives me a 99.97% average. The problem is present on all calls (except calling into the E1 to a conference or to MoH). I am preparing a new server to see if it is a hardware issue.
|
This is the bit I mean, but if you have calls going over the E1 that are okay then its probably not this.
http://www.voip-info.org/wiki/view/Asterisk+PRI
| Quote: | # span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow]
#
# All T1/E1 spans generate a clock signal on their transmit side. The
# <timing source> parameter determines whether the clock signal from the far
# end of the T1/E1 is used as the master source of clock timing. If it is, our
# own clock will synchronise to it. T1/E1's connected directly or indirectly to
# a PSTN provider (telco) should generally be the first choice to sync to. The
# PSTN will never be a slave to you. You must be a slave to it.
#
# Chose 1 to make the equipment at the far end of the E1/T1 link the preferred
# source of the master clock. Chose 2 to make it the second choice for the master
# clock, if the first choice port fails (the far end dies, a cable breaks, or
# whatever). Chose 3 to make a port the third choice, and so on. If you have, say,
# 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each
# port should be different.
#
# If you choose 0, the port will never be used as a source of timing. This is
# appropriate when you know the far end should always be a slave to you. If the
# port is connected to a channel bank, for example, you should always be its
# master. Any number of ports can be marked as 0.
#
# Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed
# faxes, unreliable modem operation, and is a general all round bad thing.
|
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