[asterisk-users] 2 PRI Card - Interrupt Problem
Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 ... 37:1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39:1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -- _ -- 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
Re: [asterisk-users] 2 PRI Card - Interrupt Problem
Yep, that's a known problem with older CentOS version. I have some 6.4 customer machines, which I cannot update easily, so I wrote a little patch for irqbalance (only a path problem). I posted some info here: http://lists.digium.com/pipermail/asterisk-dev/2013-December/064182.html Let me know if you want to look at it. jg -- _ -- 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
Re: [asterisk-users] 2 PRI Card - Interrupt Problem
Look for irqbalancer for your distribution: http://www.tutorialspoint.com/unix_commands/irqbalance.htm Am 14.05.2014 09:00, schrieb Chandrakant Solanki: Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 ... 37:1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39:1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -- _ -- 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
Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?
Hello,
After a small break from working on this, I got the idea of tcpdumping the
correct ports. What I see is REGISTER messages from Kamailio port to
Asterisk, which are replied with 401 Unauthorized. Why is this happening?
In my sippeers table the secret field has no value (tried both NULL and
empty string) and the added field sippasswd has the correct password for
the user.
The above might be the cause of my problem, would anyone be able to advice
me to get to correct behaviour? Now Kamailio sees the clients as
registered, which would be wrong if Asterisk doesn't.
cheers,
Olli
2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the user data. Field regseconds has a value and
fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as they
are on the same machine).
I have a very simple dialplan:
[general]
[default]
exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
same = n,Dial(SIP/${EXTEN},3600,rt)
same = n,Hangup
Here's more on my problem and background to it, guys on the Kamailio list
helped out but looks like I need to check my Asterisk configuration.
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html
My goal is to have all clients in the asterisk database, asterisk (one at
this point, several later) handling the calls and Kamailio as proxy. In
Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
domain 'testers.com'.
I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
the same rental virtual server. Clients are in my home network behind nat.
In MySQL I have database asterisk with table sippeers, where I have
clients added like this:
INSERT INTO sippeers
(name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
','660','friend');
In this message there are some outputs and a sip trace of a register:
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html
What I don't know is how to configure sip.conf, so far I've just been
making guesses based on online examples and documentation.
My current sip.conf looks like this:
[general]
bindport = 5070
bindaddr = 127.0.0.1
tcpbindaddr = 127.0.0.1:5070
tcpenable = no
limitonpeers = yes
;rtcachefriends = yes
tos_sip=cs3
tos_audio=ef
realm = testers.com
I've tried defining realm and domain values, but I lack proper
understanding of those. Can you guys help me out? Are there any other
configurations I need to check?
Respectfully,
Olli
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Re: [asterisk-users] 2 PRI Card - Interrupt Problem
Thanks for reply, I am interested to see patch, but I don't find any link for the same. -- Chandrakant Solanki On Wed, May 14, 2014 at 2:09 PM, Thorsten Göllner t...@ovm-group.com wrote: Look for irqbalancer for your distribution: http://www.tutorialspoint.com/unix_commands/irqbalance.htm Am 14.05.2014 09:00, schrieb Chandrakant Solanki: Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 ... 37:1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39:1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Ringing issue
On Tue, 13 May 2014 15:28:26 +0100 Gareth Blades mailinglist+aster...@dns99.co.uk wrote: You would need to provide more information. Mobiles and landlines are not SIP and yet you say calls are coming into your asterisk over SIP. So what or who is doing the translation? My origination provider. While I do have a SIP address, no one is calling it and other than local sets (which don't seem to have this issue) all calls are coming through my single origination provider. This is why I am confused. Virtually all calls are coming from the PSTN through one connection. If all callers had the problem it would almost make more sense. Initial thoughts are that it could be you are sending back SIP/180 with no session progress and indicating ringing but the other end is misconfiguration and not generating its own ring tone. This is possible if you have multiple providers sending you calls or one provider using different kit for different geographic areas. Geographic doesn't seem to be the issue. Most calls are coming from Toronto, Canada where I am. They come from major carriers. Rogers is the largest cell carrier here and that appears to be one place where it fails. I am on Koodo which uses the Telus network, the second largest, and mine works fine. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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
Re: [asterisk-users] new install: no re-invite and unwanted transcoding
I can't help on the first issue, but for the second have you tried doing
Set(${SIP_CODEC}=ulaw) before dialing the trunk? I'm in a similar situation
where we have g722 internally but our trunk provider only offers ulaw so I
see g722-slin-ulaw transcoding. I'm thinking of trying it here (on
1.8.14.1) to troubleshoot our occasional outbound issues.
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Re: [asterisk-users] new install: no re-invite and unwanted transcoding
Inband audio, such as ringing, busy, intercept, silence, etc require
transcoding. We solved the issue on our Asterisk installs by purchasing a
hardware transcoding card (for G729, but the card supports several codecs in
hardware. In my experience transcoding happens, accept it and move forward.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Thomas
Sent: Wednesday, May 14, 2014 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] new install: no re-invite and unwanted transcoding
I can't help on the first issue, but for the second have you tried doing
Set(${SIP_CODEC}=ulaw) before dialing the trunk? I'm in a similar situation
where we have g722 internally but our trunk provider only offers ulaw so I see
g722-slin-ulaw transcoding. I'm thinking of trying it here (on 1.8.14.1) to
troubleshoot our occasional outbound issues.
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Re: [asterisk-users] Ringing issue
On Tue, 13 May 2014 15:28:26 +0100 Gareth Blades mailinglist+aster...@dns99.co.uk wrote: Initial thoughts are that it could be you are sending back SIP/180 with no session progress and indicating ringing but the other end is misconfiguration and not generating its own ring tone. This is possible if you have multiple providers sending you calls or one provider using different kit for different geographic areas. I seem to have solved this, sorta. My Provider, Thinktel in Canada, normally sets PBX plays ringback to false meaning that they generate the ring tone in all cases. By mistake it was set to true on my trunk. They changed that and now the callers are hearing a ring tone. It's still an interesting question I think. What if I wanted to do something with early media? That is not possible with this setup. Anyway, here it is for future searchers. Talk to your origination provider if you have this problem. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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
Re: [asterisk-users] Asterisk 11.9 with webRTC demo integration
On Sat, May 10, 2014 at 2:27 AM, bhavik patel bhavikpatel14...@gmail.com wrote: Hi All, snip For Outbound calls : when i am dialling 8002 - 8001 every time Chrome Browser asking for allow microphone. Is there any way to disable asking permission and allowing it by default ? when i allow microphone then SIpml5 phone showing like Not Allow. That is a question about Chrome, not about Asterisk. A quick Google search pulls up this information: https://support.google.com/chrome/answer/2693767?hl=en If you select Allow on a http URL your preference will not be remembered in future visits. If you select Allow on a https URL, your preference will be remembered in future visits. snip Here is the asterisk logs : http://pastebin.com/JZeDjyay For Incoming calls : When call come to browser,And allow microphone then Call rejected and asterisk showing like Got SIP response 603 Failed to get local SDP in asterisk CLI. But After some google i found new link https://code.google.com/p/sipml5/wiki/Downloads for SIPml-api.js and after replacing that JS File Calls are comming in browser even i am able to answer that calls,Also in browser it says In call but in asterisk CLI it keep showing ringing and other end showing like remote ringing . Not sure what is going on here. You can try following my tutorial for testing with the SIPML5 demo here: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 , It also uses Asterisk 11 and chan_sip which matches what you are doing. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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
Re: [asterisk-users] Asterisk 11.9 with webRTC demo integration
Hello, I'm far from being an expert, but as far as I know when you use https in your website the browser will ask to use the audio devices only once and then remembers your decision. When using http it will ask every time. Sorry I can't be of more help but hope this helps. cheers, Olli 2014-05-10 10:27 GMT+03:00 bhavik patel bhavikpatel14...@gmail.com: Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support. I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I have tried both browser Google Chrome and Firefox with their latest versions. For asterisk, I made some configuration like below. Please check : http://pastebin.com/7KCvtcNf For Outbound calls : when i am dialling 8002 - 8001 every time Chrome Browser asking for allow microphone. Is there any way to disable asking permission and allowing it by default ? when i allow microphone then SIpml5 phone showing like Not Allow. Here is the asterisk logs : http://pastebin.com/JZeDjyay For Incoming calls : When call come to browser,And allow microphone then Call rejected and asterisk showing like Got SIP response 603 Failed to get local SDP in asterisk CLI. But After some google i found new link https://code.google.com/p/sipml5/wiki/Downloads for SIPml-api.js and after replacing that JS File Calls are comming in browser even i am able to answer that calls,Also in browser it says In call but in asterisk CLI it keep showing ringing and other end showing like remote ringing . Here is the asterisk logs : http://pastebin.com/e8Ap3bhq Can anyone please let me know what am i doing wrong? -- Thanks, Bhavik Patel -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Terrible dahdi_test results
I remembered I have an older box with a Wildcard TE12xP that uses the wcte12xp module with a newer 3.9.11 kernel that works perfectly. I setup the problematic machine with the same kernel in the hope that this might be relevant. Unfortunately the same situation persists. I used the /proc/timer_stats to see how the timers were used: Timer Stats Version: v0.2 Sample period: 10.002 s 311, 15081 kworker/u:0 mod_timer (te12xp_timer) With the TE110P I couldn't find any entry It seems that the timing mechanism is different, it doesn't use mod_timer. I'm running out of ideas. Please help. Thanks, Mike On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote: Thanks again Russ, Just a quick reply for now. No virtualization, but yes I am running a tickless kernel: # # Processor type and features # CONFIG_NO_HZ=y Standard for debian kernels. I booted with nohz=off and the behaviour changed. Unfortunately for the worse: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 66.653% 66.683% 66.683% 66.807% 67.705% 66.666% 66.651% 66.679% 67.516% 66.882% 66.649% 66.657% 66.678% 66.668% 66.672% 66.664% 66.675% 66.675% 66.659% 66.692% 66.631% 66.187% 66.650% 66.710% 66.648% 66.633% 66.714% 66.638% 66.688% 66.794% 66.645% 66.696% --- Results after 32 passes --- Best: 67.705% -- Worst: 66.187% -- Average: 66.726523% Comparing the boot messages without nohz=off: [0.00] hpet clockevent registered [0.00] Fast TSC calibration failed [0.00] TSC: Unable to calibrate against PIT [0.00] TSC: using HPET reference calibration [0.00] Detected 2593.456 MHz processor. and with nohz=off: [0.00] hpet clockevent registered [0.00] Fast TSC calibration using PIT [0.00] Detected 2593.225 MHz processor. I am encouraged that we seem to be homing in on the problem. I need to read up a bit more on the subject and look at possible power saving issues on this machine. Best regards, Mike On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote: On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote: But on examination the /etc/init.d/dahdi start was only loading the dahdi module. With this in mind I might start looking around the system for things which might cause jitter in the servicing of system timer interrupts: Are you running under a virtualized environment? Are you running a tickless kernel? (maybe try adding nohz=off to your kernel boot parameters) Is there some sort of processor power saving or frequency scaling going on that interrupts the system timer? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
Re: [asterisk-users] Terrible dahdi_test results
Try the card in another machine with a different brand of motherboard. If it works you know it is a hardware issue. Do you have an actual T-1 plugged into your card? If not, try that and see if there is any difference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Leddy Sent: Wednesday, May 14, 2014 3:43 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Terrible dahdi_test results I remembered I have an older box with a Wildcard TE12xP that uses the wcte12xp module with a newer 3.9.11 kernel that works perfectly. I setup the problematic machine with the same kernel in the hope that this might be relevant. Unfortunately the same situation persists. I used the /proc/timer_stats to see how the timers were used: Timer Stats Version: v0.2 Sample period: 10.002 s 311, 15081 kworker/u:0 mod_timer (te12xp_timer) With the TE110P I couldn't find any entry It seems that the timing mechanism is different, it doesn't use mod_timer. I'm running out of ideas. Please help. Thanks, Mike On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote: Thanks again Russ, Just a quick reply for now. No virtualization, but yes I am running a tickless kernel: # # Processor type and features # CONFIG_NO_HZ=y Standard for debian kernels. I booted with nohz=off and the behaviour changed. Unfortunately for the worse: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 66.653% 66.683% 66.683% 66.807% 67.705% 66.666% 66.651% 66.679% 67.516% 66.882% 66.649% 66.657% 66.678% 66.668% 66.672% 66.664% 66.675% 66.675% 66.659% 66.692% 66.631% 66.187% 66.650% 66.710% 66.648% 66.633% 66.714% 66.638% 66.688% 66.794% 66.645% 66.696% --- Results after 32 passes --- Best: 67.705% -- Worst: 66.187% -- Average: 66.726523% Comparing the boot messages without nohz=off: [0.00] hpet clockevent registered [0.00] Fast TSC calibration failed [0.00] TSC: Unable to calibrate against PIT [0.00] TSC: using HPET reference calibration [0.00] Detected 2593.456 MHz processor. and with nohz=off: [0.00] hpet clockevent registered [0.00] Fast TSC calibration using PIT [0.00] Detected 2593.225 MHz processor. I am encouraged that we seem to be homing in on the problem. I need to read up a bit more on the subject and look at possible power saving issues on this machine. Best regards, Mike On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote: On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote: But on examination the /etc/init.d/dahdi start was only loading the dahdi module. With this in mind I might start looking around the system for things which might cause jitter in the servicing of system timer interrupts: Are you running under a virtualized environment? Are you running a tickless kernel? (maybe try adding nohz=off to your kernel boot parameters) Is there some sort of processor power saving or frequency scaling going on that interrupts the system timer? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- _ -- 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
Re: [asterisk-users] Terrible dahdi_test results
Hi Eric, I plugged an E1 into the card and it doesn't make any difference. # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 88.829% 87.806% 88.988% 88.854% 88.944% 88.952% 88.967% 88.841% 88.889% 88.946% 88.933% 88.841% 88.885% 89.050% 88.904% 87.933% 88.912% 88.949% 88.913% 88.886% 88.891% 88.798% 88.746% 89.009% 88.934% 88.870% 88.875% 89.003% 88.925% 88.863% 89.018% 88.093% 88.447% 88.691% 89.034% 88.703% 88.815% 89.011% 88.919% 88.825% etc. I will try the card in an older machine tomorrow. Ironic is that i bought this card because it has a PCI express interface so I can use it in recent servers but it uses an older chipset and driver than I was using. Thanks for the help, Mike On Wed, 2014-05-14 at 15:54 -0400, Eric Wieling wrote: Try the card in another machine with a different brand of motherboard. If it works you know it is a hardware issue. Do you have an actual T-1 plugged into your card? If not, try that and see if there is any difference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Leddy Sent: Wednesday, May 14, 2014 3:43 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Terrible dahdi_test results I remembered I have an older box with a Wildcard TE12xP that uses the wcte12xp module with a newer 3.9.11 kernel that works perfectly. I setup the problematic machine with the same kernel in the hope that this might be relevant. Unfortunately the same situation persists. I used the /proc/timer_stats to see how the timers were used: Timer Stats Version: v0.2 Sample period: 10.002 s 311, 15081 kworker/u:0 mod_timer (te12xp_timer) With the TE110P I couldn't find any entry It seems that the timing mechanism is different, it doesn't use mod_timer. I'm running out of ideas. Please help. Thanks, Mike On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote: Thanks again Russ, Just a quick reply for now. No virtualization, but yes I am running a tickless kernel: # # Processor type and features # CONFIG_NO_HZ=y Standard for debian kernels. I booted with nohz=off and the behaviour changed. Unfortunately for the worse: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 66.653% 66.683% 66.683% 66.807% 67.705% 66.666% 66.651% 66.679% 67.516% 66.882% 66.649% 66.657% 66.678% 66.668% 66.672% 66.664% 66.675% 66.675% 66.659% 66.692% 66.631% 66.187% 66.650% 66.710% 66.648% 66.633% 66.714% 66.638% 66.688% 66.794% 66.645% 66.696% --- Results after 32 passes --- Best: 67.705% -- Worst: 66.187% -- Average: 66.726523% Comparing the boot messages without nohz=off: [0.00] hpet clockevent registered [0.00] Fast TSC calibration failed [0.00] TSC: Unable to calibrate against PIT [0.00] TSC: using HPET reference calibration [0.00] Detected 2593.456 MHz processor. and with nohz=off: [0.00] hpet clockevent registered [0.00] Fast TSC calibration using PIT [0.00] Detected 2593.225 MHz processor. I am encouraged that we seem to be homing in on the problem. I need to read up a bit more on the subject and look at possible power saving issues on this machine. Best regards, Mike On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote: On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote: But on examination the /etc/init.d/dahdi start was only loading the dahdi module. With this in mind I might start looking around the system for things which might cause jitter in the servicing of system timer interrupts: Are you running under a virtualized environment? Are you running a tickless kernel? (maybe try adding nohz=off to your kernel boot parameters) Is there some sort of processor power saving or frequency scaling going on that interrupts the system timer? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- _ -- 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
Re: [asterisk-users] Terrible dahdi_test results
On Wed, May 14, 2014 at 3:41 PM, Mike Leddy m...@loop.com.br wrote: Hi Eric, I plugged an E1 into the card and it doesn't make any difference. Check to see if the card is interrupting 1000 times per second with something like: cat /proc/interrupts | grep wc sleep 1 cat /proc/interrupts | grep wc You could also try manually compiling dahdi_dummy by commenting it back in, in the file: drivers/dahdi/Kbuild Then modprobe dahdi_dummy This module forces the use of the high res timers -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
