Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - FIXED...?
...which, after upgrading to Dahdi 2.4.0 seem to be gone. Not sure if 2.3.0.1 was the culprit, or just the fact that sonething was stuck (stopping dahdi took about 5 attempts because it was apparently in use, even if Asterisk had been shutdown). Hopefully this isn't just the calm before the storm and Dahdi will behave under load. Thanks for the help Shaun, I appreciate you helping point me in the right direction. Hopefully this will help someone. Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 2:55 To: 'Shaun Ruffell' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Actually, after further investigation I found that every 10 minutes I get massive irq usage spikes. The CPU is busy servicing hardware interrupts. I do have a 4 port PRI card and a TCE400B card, but tha happens even when asterisk activity=0. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:58 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes When the asterisk process spikes, do you know if there is a corresponding spike in us, sy, ni, hi, or si time? On 09/27/2010 02:53 PM, Mike wrote: That`s exactly what I did, and exactly what didn't happen. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:48 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Just something like echo 1200 /proc/sys/net/ipv4/route/secret_interval should have been sufficient to half the frequency of the CPU spikes. On 09/27/2010 02:42 PM, Mike wrote: I just changed the file, did I need to reboot or do anything for the change to take effect? Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:34 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Hmm..ok, then I'm not sure I'll be able to help. It's just that I know I've seen issues similar to what you're reporting based on the route table flushing pushing some si time into the asterisk process (and it runs at 10 minute intervals by default). So I'm not as familiar with the Asterisk code base to know what sort of periodic tasks may have changed between the versions you reported. On 09/27/2010 02:23 PM, Mike wrote: That wasn't it. I wonder what happens precisely every 10 minutes in Asterisk (or that is accounted to Asterisk's CPU usage). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
(sorry for the series of email, I realize I seem to be having a discussion with myself). I was wrong, the problem isn't fixed. Is having IRQ spikes every 10 minutes (under no load at all) the norm with Dahdi hardware? Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 3:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - FIXED...? ...which, after upgrading to Dahdi 2.4.0 seem to be gone. Not sure if 2.3.0.1 was the culprit, or just the fact that sonething was stuck (stopping dahdi took about 5 attempts because it was apparently in use, even if Asterisk had been shutdown). Hopefully this isn't just the calm before the storm and Dahdi will behave under load. Thanks for the help Shaun, I appreciate you helping point me in the right direction. Hopefully this will help someone. Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 2:55 To: 'Shaun Ruffell' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Actually, after further investigation I found that every 10 minutes I get massive irq usage spikes. The CPU is busy servicing hardware interrupts. I do have a 4 port PRI card and a TCE400B card, but tha happens even when asterisk activity=0. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:58 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes When the asterisk process spikes, do you know if there is a corresponding spike in us, sy, ni, hi, or si time? On 09/27/2010 02:53 PM, Mike wrote: That`s exactly what I did, and exactly what didn't happen. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:48 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Just something like echo 1200 /proc/sys/net/ipv4/route/secret_interval should have been sufficient to half the frequency of the CPU spikes. On 09/27/2010 02:42 PM, Mike wrote: I just changed the file, did I need to reboot or do anything for the change to take effect? Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:34 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Hmm..ok, then I'm not sure I'll be able to help. It's just that I know I've seen issues similar to what you're reporting based on the route table flushing pushing some si time into the asterisk process (and it runs at 10 minute intervals by default). So I'm not as familiar with the Asterisk code base to know what sort of periodic tasks may have changed between the versions you reported. On 09/27/2010 02:23 PM, Mike wrote: That wasn't it. I wonder what happens precisely every 10 minutes in Asterisk (or that is accounted to Asterisk's CPU usage). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] 1.4, 1.6, 1.8 versions and the H323 Gatekeeper functionality
Dear Leif; Thanks a lot for ur kindly reply. Well, based on what I have to decide to go for 1.6 or 1.8 version if I am at 1.4 version? About the H323 gatekeeper functionality, I was hearing there is some work on this direction at the 1.6 and 1.8 versions, but I was would to hear any confirmation regarding this .. no one give any details in this. Regards Bilal - On 10-09-26 01:00 PM, bilal ghayyad wrote: First of all, I am looking to have the H323 Gatekeeper service available at Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing H323 gatekeeper functionality or not? Until 1.4.26.2 version, there is no h323 gatekeeper functionality. So, any implementation for this feature has been done in the other versions? From what I'm aware of, no additional work has been done on the H323 modules available for Asterisk that implements any sort of gatekeeper functionality. Leif. -- _ -- 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
[asterisk-users] 1.6 and 1.8 version A2Billing
Hi All; Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4? Appreciate ur kindly help. Regards Bilal -- _ -- 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] Problems compiling Asterisk on Debian
r...@sangoma-testing:/home# cat /lib/modules/2.6.26-2-amd64/build/.config cat: /lib/modules/2.6.26-2-amd64/build/.config: No such file or directory r...@sangoma-testing:/home# cat /usr/src/linux/.config cat: /usr/src/linux/.config: No such file or directory r...@sangoma-testing:/home# uname -a Linux Sangoma-Testing 2.6.26-2-amd64 #1 SMP Thu Sep 16 15:56:38 UTC 2010 x86_64 GNU/Linux r...@sangoma-testing:/home# uname -r 2.6.26-2-amd64 2010/9/28 Paul Belanger paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 6:57 PM, Danny Dias ing.diasda...@gmail.com wrote: r...@sangoma-testing:/home# ls -la /lib/modules/ total 12 drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 $ cat /lib/modules/2.6.26-2-amd64/build/.config r...@sangoma-testing:/home# ls -la /usr/src/linux lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux - $ cat /usr/src/linux/.config -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Cisco 9971
Sascha, Could you possibly share what you have done to get the 9971 working? I have one of these and would dearly love to get it talking to my Elastix box with or without annoying beep :-) Thanks, DT From: Sascha Ferley sascha.fer...@infineon.net To: asterisk-users@lists.digium.com Sent: Tue, August 31, 2010 3:20:52 AM Subject: [asterisk-users] Cisco 9971 Hi, I am having a weird issue with a Cisco 9971 phone. I managed to get most of it working, including the side car, however one of the issues is that there seems to be some sort of side tone / beep occurring roughly every 13 seconds or so, as if the phone is activated with call waiting. However none of this is activated and is still making the annoying beeping side tone. The phone does require that one runs with tcp=enable and transport=tcp, thus turning on the presence information, which from the logs seems to be refreshing roughly every 10 - 15 seconds. However the Patch has not been compiled in, thus the information being sent to the phone is incorrect and thus am wondering if this is what is causing this annoying side tone. If anyone knows, please let me know or anyone has any experience with the 9971's .. Would be awesome to get this working correctly Thanks -- _ -- 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] NAT issue (i think?)
On Tue, Sep 28, 2010 at 09:28:24AM +0800, Ron wrote: got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. Sounds like NAT problems, do you have qualify enabled for these devices? Also the Linksys devices have a keep alive option (NAT Keep Alive Enable). But even with both these setting enabled NAT gateways sometimes seem to lose track of SIP sessions (I have more trouble with Cisco devices than Linux routers), setting the UDP session timeout to 10m seems to help (default is something like 3m). -- Daniel Tryba -- _ -- 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
[asterisk-users] ISDN - Busy signal on 3rd call
Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html -- _ -- 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] NAT issue (i think?)
hi daniel thank you for your reply, i have enabled NAT keep-alive and NAT-Mapping on the Linksys devices. i have enabled qualify before but it seems it's worst as the linksys devices keeps on rebooting and there was one issue that when i set qualify to all users my server bogged down and asterisk crashed. where do i enable the UDP session timeout? on the linksys devices or the asterisk? TIA Regards Ron On 9/28/10 6:14 PM, Daniel Tryba wrote: On Tue, Sep 28, 2010 at 09:28:24AM +0800, Ron wrote: got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. Sounds like NAT problems, do you have qualify enabled for these devices? Also the Linksys devices have a keep alive option (NAT Keep Alive Enable). But even with both these setting enabled NAT gateways sometimes seem to lose track of SIP sessions (I have more trouble with Cisco devices than Linux routers), setting the UDP session timeout to 10m seems to help (default is something like 3m). -- _ -- 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
[asterisk-users] Inbound calls from TRUNK
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html xmlns:v=urn:schemas-microsoft-com:vml xmlns:o=urn:schemas-microsoft-com:office:office xmlns:w=urn:schemas-microsoft-com:office:word xmlns:m=http://schemas.microsoft.com/office/2004/12/omml; xmlns=http://www.w3.org/TR/REC-html40;}
{\*\htmltag2 \par }
{\*\htmltag2 \par }
{\*\htmltag34 head}
{\*\htmltag1 \par }
{\*\htmltag1 \par }
{\*\htmltag161 meta name=Generator content=Microsoft Word 12 (filtered medium)}
{\*\htmltag1 \par }
{\*\htmltag241 style}
{\*\htmltag241 \par !--\par /* Font Definitions */\par @font-face\par \tab \{font-family:Cambria Math;\par \tab panose-1:2 4 5 3 5 4 6 3 2 4;\}\par @font-face\par \tab \{font-family:Calibri;\par \tab panose-1:2 15 5 2 2 2 4 3 2 4;\}\par @font-face\par \tab \{font-family:Tahoma;\par \tab panose-1:2 11 6 4 3 5 4 4 2 4;\}\par /* Style Definitions */\par p.MsoNormal, li.MsoNormal, div.MsoNormal\par \tab \{margin:0in;\par \tab margin-bottom:.0001pt;\par \tab font-size:11.0pt;\par \tab font-family:Calibri,sans-serif;\}\par a:link, span.MsoHyperlink\par \tab \{mso-style-priority:99;\par \tab color:blue;\par \tab text-decoration:underline;\}\par a:visited, span.MsoHyperlinkFollowed\par \tab \{mso-style-priority:99;\par \tab color:purple;\par \tab text-decoration:underline;\}\par p.MsoAcetate, li.MsoAcetate, div.MsoAcetate\par \tab \{mso-style-priority:99;\par \tab mso-style-link:Balloon Text Char;\par \tab margin:0in;\par \tab margin-bottom:.0001pt;\par \tab font-size:8.0pt;\par \tab font-family:Tahoma,sans-serif;\}\par span.EmailStyle17\par \tab \{mso-style-type:personal-compose;\par \tab font-family:Calibri,sans-serif;\par \tab color:windowtext;\}\par span.BalloonTextChar\par \tab \{mso-style-name:Balloon Text Char;\par \tab mso-style-priority:99;\par \tab mso-style-link:Balloon Text;\par \tab font-family:Tahoma,sans-serif;\}\par .MsoChpDefault\par \tab \{mso-style-type:export-only;\}\par @page Section1\par \tab \{size:8.5in 11.0in;\par \tab margin:1.0in 1.0in 1.0in 1.0in;\}\par div.Section1\par \tab \{page:Section1;\}\par --\par }
{\*\htmltag249 /style}
{\*\htmltag1 \par }
{\*\htmltag241 !--[if gte mso 9]xml\par o:shapedefaults v:ext=edit spidmax=1026 /\par /xml![endif]--}
{\*\htmltag241 !--[if gte mso 9]xml\par o:shapelayout v:ext=edit\par o:idmap v:ext=edit data=1 /\par /o:shapelayout/xml![endif]--}
{\*\htmltag1 \par }
{\*\htmltag41 /head}
{\*\htmltag2 \par }
{\*\htmltag2 \par }
{\*\htmltag50 body lang=EN-US link=blue vlink=purple}\htmlrtf \lang1033 \htmlrtf0
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag96 div class=Section1}\htmlrtf {\htmlrtf0
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0 Hi ,
{\*\htmltag244 o:p}
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0
{\*\htmltag244 o:p}
{\*\htmltag84 nbsp;}\htmlrtf \'a0\htmlrtf0
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0 I have configured sip trunk and put it
{\*\htmltag84 nbsp;}\htmlrtf \'a0\htmlrtf0 in route for
{\*\htmltag4 \par }\htmlrtf \htmlrtf0 asterisk extensions
{\*\htmltag244 o:p}
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0 How can I allow anonymous
{\*\htmltag84 nbsp;}\htmlrtf \'a0\htmlrtf0 calls from trunk to
{\*\htmltag4 \par }\htmlrtf \htmlrtf0 extensions .
{\*\htmltag244 o:p}
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0 All calls as a forbidden sip request
{\*\htmltag244 o:p}
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0
{\*\htmltag244 o:p}
{\*\htmltag84 nbsp;}\htmlrtf \'a0\htmlrtf0
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0
{\*\htmltag244 o:p}
{\*\htmltag84 nbsp;}\htmlrtf \'a0\htmlrtf0
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
\htmlrtf \par
\htmlrtf0
{\*\htmltag72 /p}
{\*\htmltag0 \par }
{\*\htmltag0 \par }
{\*\htmltag64 p class=MsoNormal}\htmlrtf {\htmlrtf0
{\*\htmltag244 o:p}
{\*\htmltag84 nbsp;}\htmlrtf \'a0\htmlrtf0
{\*\htmltag252 /o:p}\htmlrtf\par}\htmlrtf0
Re: [asterisk-users] NAT issue (i think?)
You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060 register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ron nha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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] func SHARED, how to use?
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Please check if in Asterisk 1.6 the Syntax for passing arguments to the M
option of Dial() has changed.
[macro-test]
exten = s,n,Set(SHARED(foo,${CHANNEL})=456 )
exten = s,n,Set(SHARED(foo,${ARG1})=456)
Philipp
--
_
-- 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
[asterisk-users] What's the meaning of this?
Hello, I'm checking this: [Sep 28 13:32:46] NOTICE[30360] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 [Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 [Sep 28 13:32:46] NOTICE[30361] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 What should i do? the calls are going down. And the Telco says that the E1's are ok -- _ -- 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
[asterisk-users] SIP X.25
Hi List. It is possible to travel over the X.25 protocol on Asterisk SIP? -- Atenciosamente Daviramos Roussenq Fortunato -- _ -- 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] NAT issue (i think?)
On Tue, Sep 28, 2010 at 07:08:36PM +0800, Ron wrote: i have enabled qualify before but it seems it's worst as the linksys devices keeps on rebooting and there was one issue that when i set qualify to all users my server bogged down and asterisk crashed. Never seen these problems before related to qualify, how many devices are there? where do i enable the UDP session timeout? on the linksys devices or the asterisk? Neither, that is a property of the NAT device. You should figure out when the problem happens wether the NAT device has any knowledge of the UDP session between the SIP device and Asterisk. -- Daniel Tryba -- _ -- 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
[asterisk-users] SIP X.25
Hi List. It is possible to travel over the X.25 protocol on Asterisk SIP? -- Atenciosamente Daviramos Roussenq Fortunato -- _ -- 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] NAT issue (i think?)
Hi Danny On the pap2 by default it is set to 3600 and i have not change that. by the way, is the NAT keep-alive same with the NOTIFY message? coz i am seeing my asterisk respond to those as bad event could that be causing it to loose the registration? here's the registration from ngrep: U 78.65.34.12:5094 - 12.34.56.78:5060 REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport. From: Kristine sip:456...@sip.mydomain.com;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com. Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. Max-Forwards: 70. Contact: Kristine sip:456...@78.65.34.12:5094;expires=3600. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . U 12.34.56.78:5060 - 78.65.34.12:5094 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094. From: Kristine sip:456...@sip.mydomain.com;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com. Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. On 9/28/10 7:24 PM, Danny Dias wrote: You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ronnha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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] func SHARED, how to use?
28.09.2010 15:35, Philipp von Klitzing ?:
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Hello!
Thank you!
I can pass this constant , but I need RTCP stats
And this
[macro-test]
exten = s,1,NoOp(test ${CHANNEL} )
exten = s,n,Set(SHARED(foo,${ARG1})=${CHANNEL(rtpqos,audio,all)} )
gives me stats on beginning of call, not at it's end...
This is mentioned here http://www.voip-info.org/wiki/view/Asterisk+RTCP
; - These RTCP stats provided by CHANNEL(rtpqos) apparently only show data for the last RTCP message we got!
But, looks like ${RTPAUDIOQOSJITTER} is not available at all:
-- Executing [...@macro-test:2] Set(SIP/6052-02b2,
SHARED(foo,DAHDI/9-1)= ) in new stack
As you see it is empty.
Hmm,looks like I need to call macro at the end of call.
Is it possible?
Thank you!
At least now I understand something ;-)
--
_
-- 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] SIP X.25
On 09/28/2010 07:05 AM, Daviramos Roussenq Fortunato wrote: Hi List. It is possible to travel over the X.25 protocol on Asterisk SIP? Your question doesn't make any sense, but we can try to answer anyway: SIP is transported over UDP or TCP, which are transported over IP. It is possible to transport IP over X.25 networking, although I doubt anyone uses X.25 for that purpose any more. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] func SHARED, how to use?
28.09.2010 16:19, Dmitry Melekhov ?: btw, about bridged variables- they are really what I need. Looks like there is bug in asterisk- if call is dropped from dahdi side- there is no info in these variables. I think I have to fill bug. Thank you! I got what I want :-) Thank you again! -- _ -- 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
[asterisk-users] E1 check with nagios, how to?
We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, --- Dario Quiroz (71) 9275-9080 gtalk: darioqui...@gmail.com --- -- _ -- 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] What's the meaning of this?
Danny Dias wrote: PRI got event: HDLC Abort (6) Google is your friend: http://lists.digium.com/pipermail/asterisk-users/2005-June/107299.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Inbound calls from TRUNK
Thanks ,it solved by adding insecure=very regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Tuesday, September 28, 2010 2:16 PM To: Asterisk; Asterisk List Subject: [asterisk-users] Inbound calls from TRUNK Hi , I have configured sip trunk and put it in route for asterisk extensions How can I allow anonymous calls from trunk to extensions . All calls as a forbidden sip request Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- 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] E1 check with nagios, how to?
what do you want to monitor? I ended up with MRTG graphing the Incoming/Ringing/Established calls. On Tue, Sep 28, 2010 at 4:22 PM, Dario Quiroz darioqui...@gmail.com wrote: We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, --- Dario Quiroz (71) 9275-9080 gtalk: darioqui...@gmail.com --- -- _ -- 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 -- Mvh, Aurimas Skirgaila -- _ -- 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] E1 check with nagios, how to?
Are you monitoring some dahdi hardware or a separate black box? If dahdi, you could write a nagios plugin in shell with something like this: ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l` and then set the appropriate exit code if ALARMS is not 0. -M On Tue, Sep 28, 2010 at 9:22 AM, Dario Quiroz darioqui...@gmail.com wrote: We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, --- Dario Quiroz (71) 9275-9080 gtalk: darioqui...@gmail.com --- -- _ -- 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] E1 check with nagios, how to?
Enjoy...you can ignore certain T1/E1 ports if you pass in the name of the
port as an argument (I use this on ports that aren't yet connected to a
telco, but I don't want to get an alert on). I execute it via NRPE on the
Asterisk box. It will give you descriptions of which ports are bad, so you
don't need to guess. :)
#!/usr/bin/perl -w
#
# Copyright (C) 2010 Local Matters, Inc.
# http://www.localmatters.com/
# Author: Joel C. Maslak
#
# Licensed under GPL version 3
#
use strict;
use Carp;
my %ignore;
MAIN: {
my @out = `/usr/sbin/dahdi_scan`;
for my $ig (@ARGV) {
$ignore{$ig} = 1;
}
my $alarm;
my $desc;
my @alarms;
for my $line (@out) {
chomp($line);
if ($line =~ /^alarms=/) {
$alarm = $line;
$alarm =~ s/^alarms=//;
}
if ($line =~ /^description=/) {
$desc = $line;
$desc =~ s/^description=//;
if (!defined($ignore{$desc})) {
if ($alarm ne 'OK') {
push @alarms, $desc: $alarm Alarm;
}
}
}
}
if (scalar(@alarms) 0) {
my $out = join '; ', @alarms;
print Circuits in alarm: $out\n;
exit(2);
} else {
print All monitored circuits OK\n;
exit(0);
}
}
On Tue, Sep 28, 2010 at 9:17 AM, Mark Deneen mden...@gmail.com wrote:
Are you monitoring some dahdi hardware or a separate black box?
If dahdi, you could write a nagios plugin in shell with something like
this:
ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l`
and then set the appropriate exit code if ALARMS is not 0.
-M
On Tue, Sep 28, 2010 at 9:22 AM, Dario Quiroz darioqui...@gmail.com
wrote:
We need to monitorate the E1 with nagios, somebody did this? any ideia?
Thanks in advance!
--
Atenciosamente,
---
Dario Quiroz
(71) 9275-9080
gtalk: darioqui...@gmail.com
---
--
_
-- 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
--
_
-- 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] E1 check with nagios, how to?
Bro,
What OS u use ??
Maybe you can use SNMP .. snmpd its a good application
linux: apt-get install smtpd
freebsd: pkg_add -rv net-snmpd
ABS[]s
2010/9/28 Joel Maslak jmas...@antelope.net
Enjoy...you can ignore certain T1/E1 ports if you pass in the name of the
port as an argument (I use this on ports that aren't yet connected to a
telco, but I don't want to get an alert on). I execute it via NRPE on the
Asterisk box. It will give you descriptions of which ports are bad, so you
don't need to guess. :)
#!/usr/bin/perl -w
#
# Copyright (C) 2010 Local Matters, Inc.
# http://www.localmatters.com/
# Author: Joel C. Maslak
#
# Licensed under GPL version 3
#
use strict;
use Carp;
my %ignore;
MAIN: {
my @out = `/usr/sbin/dahdi_scan`;
for my $ig (@ARGV) {
$ignore{$ig} = 1;
}
my $alarm;
my $desc;
my @alarms;
for my $line (@out) {
chomp($line);
if ($line =~ /^alarms=/) {
$alarm = $line;
$alarm =~ s/^alarms=//;
}
if ($line =~ /^description=/) {
$desc = $line;
$desc =~ s/^description=//;
if (!defined($ignore{$desc})) {
if ($alarm ne 'OK') {
push @alarms, $desc: $alarm Alarm;
}
}
}
}
if (scalar(@alarms) 0) {
my $out = join '; ', @alarms;
print Circuits in alarm: $out\n;
exit(2);
} else {
print All monitored circuits OK\n;
exit(0);
}
}
On Tue, Sep 28, 2010 at 9:17 AM, Mark Deneen mden...@gmail.com wrote:
Are you monitoring some dahdi hardware or a separate black box?
If dahdi, you could write a nagios plugin in shell with something like this:
ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l`
and then set the appropriate exit code if ALARMS is not 0.
-M
On Tue, Sep 28, 2010 at 9:22 AM, Dario Quiroz darioqui...@gmail.com wrote:
We need to monitorate the E1 with nagios, somebody did this? any ideia?
Thanks in advance!
--
Atenciosamente,
---
Dario Quiroz
(71) 9275-9080
gtalk: darioqui...@gmail.com
---
--
_
-- 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
--
_
-- 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
On 09/28/2010 02:37 AM, Mike wrote: I was wrong, the problem isn't fixed. Is having IRQ spikes every 10 minutes (under no load at all) the norm with Dahdi hardware? Interrupt spikes on 10 minute intervals is not (should not be) the norm with DAHDI hardware. What is the make/model of the PRI card that you are using? Do you know what devices are on the interrupt line on which you see the spike? The TCE400 will generate many interrupts when channels are setup and destroyed and will generate an interrupt for each packet to be transcoded. When you have more than 40 channels open the card will switch off interrupts and just poll the interface. So, given that, it's possible that if the call load is dipping below 40 channels and then going back over you could see an interrupt spike, but that doesn't sound like what you're describing. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] TELUS British Columbia PRI Settings
I'd just like to thank everyone that helped me get this running. I thought I had a FAS PRI but it turns out it was NFAS so there was no dchannel on the second PRI. When getting the 2nd PRI changed, I received an email containing the cheat sheet TELUS employees use which I thought I should share, though quite a few settings will be obvious to Asterisk gurus. The TELUS cheat sheet Asterisk PRI provisioning = T1 span timing is provisioned in /etc/zaptel.conf or /etc/dahdi/system.conf --- For syntax see [system.conf.sample](http://svn.digium.com/svn/dahdi/tools/trunk/system.conf.sample) ### Defining a span as esf/b8zs span=1,0,0,esf,b8zs ### Defining a span as sf/ami span=1,0,0,d4,ami PRI B/D channels are provisioned in /etc/zaptel.conf or /etc/dahdi/system.conf -- For syntax see [system.conf.sample](http://svn.digium.com/svn/dahdi/tools/trunk/system.conf.sample) bchan=1-23 dchan=24 PRI interface options are provisioned in /etc/asterisk/zapata.conf or /etc/asterisk/chan_dahdi.conf --- For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) [channels] ;--- PRI interface options ; Switchtype: Only used for PRI. ; ; national:National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess:ATT 4ESS ; 5ess:Lucent 5ESS ; euroisdn:EuroISDN (common in Europe) ; ni1: Old National ISDN 1 ; qsig:Q.SIG switchtype=national context=incoming-from-dahdi ; defined in extensions.conf signalling=pri_cpe ; should be pri-cpe for customer side ### Defining channel groups for selection Once you've defined groups in chan_dahdi.conf.sample they can be referenced in extensions.conf [channels] ;--- PRI interface options . . . group=1 ; reference for hunting method in extensions.conf channel=1-23 ; follows b-channel provisioning ### TON / NPI For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; dynamic:Dynamically selects the appropriate dialplan ; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) pridialplan For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) * national : Interpret the digits as a national number. * international : A fully formed E.164 phone number. [channels] . . . ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for ; the dialed number. For most installations, leaving this as 'unknown' (the ; default) works in the most cases. In some very unusual circumstances, you ; may need to set this to 'dynamic' or 'redundant'. Note that if you set one ; of the others, you will be unable to dial another class of numbers. For ; example, if you set 'national', you will be unable to dial local or ; international numbers. pridialplan=unknown ; Asterisk default prilocaldialplan For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) * national : Interpret the digits as a national number. * international : A fully formed E.164 phone number. ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's ; numbering plan). In North America, the typical use is sending the 10 digit ; callerID number and setting the prilocaldialplan to 'national' (the default). prilocaldialplan=national ; Asterisk default Hunting (selection of a specific trunk) --- ### Invoking hunting in Asterisk dialplans For syntax see [extensions.config.sample](http://svn.digium.com/svn/asterisk/trunk/configs/extensions.conf.sample) ; g: select the lowest-numbered non-busy DAHDI channel ;(aka. ascending sequential hunt group). ; G: select the highest-numbered non-busy DAHDI channel ;(aka. descending sequential hunt group). exten = 1,1,Dial(DAHDI/g1/8675309) ; ascending sequential hunt group (LIDL towards provider) exten = 2,1,Dial(DAHDI/G1/8675309) ; descending sequential hunt group (MIDL towards provider)
Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
How do I go about finding which device is causing those interrupts? (to make sure it`s actually Digium hardware and not something else?) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Tuesday, September 28, 2010 12:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED All Digium, on a HP DL360G5. I am using the TE420 (4 ports, PCIe) and a TCE400B If you are asking for cat /proc/interrupts (sorry, not much of a H/W guy) that would be the answer: CPU0 CPU1 CPU2 CPU3 0: 3692453723 0 0 0IO-APIC-edge timer 1:673333 0 0IO-APIC-edge i8042 8: 1 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 4 0 0 0IO-APIC-edge i8042 14: 29975595 42033164 0 0IO-APIC-edge ide0 82:775 3344 0 0 IO-APIC-level uhci_hcd:usb4 90:54507743045902 0 0 IO-APIC-level uhci_hcd:usb5, hpilo 114: 354830040 364865092 43218758 878551 PCI-MSI-X cciss0 130: 1525729157 0 0 0 PCI-MSI eth0 138: 39424975 287760152 0 0 PCI-MSI eth1 146:9481462 54600537 0 0 IO-APIC-level ipmi_si 169: 261607605 3446075481 482400759 18842256 IO-APIC-level uhci_hcd:usb1, tce4000 177: 19031 1594156509 1263212294 834831045 IO-APIC-level uhci_hcd:usb2, wct4xxp 185: 0 0 0 0 IO-APIC-level uhci_hcd:usb3 NMI:111720613720781264324 955544 LOC: 3692453666 3692453584 3692453552 3692453473 ERR: 0 MIS: 0 If, as a non-HW guy, I understand correctly, it means that each card is sharing interrupts with a USB port. Nothing is plugged in those USB ports. If I remember correctly, the HP BIOS didn't let me do what I wanted exactly. Would disabling the USB ports help? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, September 28, 2010 12:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED On 09/28/2010 02:37 AM, Mike wrote: I was wrong, the problem isn't fixed. Is having IRQ spikes every 10 minutes (under no load at all) the norm with Dahdi hardware? Interrupt spikes on 10 minute intervals is not (should not be) the norm with DAHDI hardware. What is the make/model of the PRI card that you are using? Do you know what devices are on the interrupt line on which you see the spike? The TCE400 will generate many interrupts when channels are setup and destroyed and will generate an interrupt for each packet to be transcoded. When you have more than 40 channels open the card will switch off interrupts and just poll the interface. So, given that, it's possible that if the call load is dipping below 40 channels and then going back over you could see an interrupt spike, but that doesn't sound like what you're describing. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
On 09/28/2010 11:40 AM, Mike wrote: How do I go about finding which device is causing those interrupts? (to make sure it`s actually Digium hardware and not something else?) Can you let something like 'while true; do date interrupt_log.txt; cat /proc/interrupts interrupt_log.txt; echo *** interrupt_log.txt; sleep 10; done' Run and see IRQ line has a jump in total interrupts (on any CPU) about the time you notice the IRQ spike from your snmp log? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
Shaun, Thanks for all the help. I tried but couldn't see a definite spike anyways. I used mpstat to see the irq% for each CPU, and although it shows me spikes to 100% for one CPU (not always the same one is spiking irq%) the intr/s isn't actually spiking. It`s like suddently there aren't more interrupts, but the IRQ% time for the CPU shoots up. What can make it do that? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, September 28, 2010 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED On 09/28/2010 11:40 AM, Mike wrote: How do I go about finding which device is causing those interrupts? (to make sure it`s actually Digium hardware and not something else?) Can you let something like 'while true; do date interrupt_log.txt; cat /proc/interrupts interrupt_log.txt; echo *** interrupt_log.txt; sleep 10; done' Run and see IRQ line has a jump in total interrupts (on any CPU) about the time you notice the IRQ spike from your snmp log? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] SIP X.25
On Tue, 2010-09-28 at 09:06 -0300, Daviramos Roussenq Fortunato wrote: Hi List. It is possible to travel over the X.25 protocol on Asterisk SIP? -- Hi Daviramos, You gotta be joking! X.25 can only be found in telecom-musea's nowadays. Latest development was the X.31 and X.32 interface towards the D-channel of isdn, which also has been discarded for 10 years or more. Forget about X.25. Due to the granularity if the timers it can not cope with linespeeds above 64Kbps. I did both level-2 and level-3 coding (from the infamous red-book) for my former employer, about twentyfive years ago! hw -- _ -- 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] TELUS British Columbia PRI Settings
On Tue, Sep 28, 2010 at 11:23:16AM -0500, jeremy.hellst...@synovate.com wrote: I'd just like to thank everyone that helped me get this running. I thought I had a FAS PRI but it turns out it was NFAS so there was no dchannel on the second PRI. When getting the 2nd PRI changed, I received an email containing the cheat sheet TELUS employees use which I thought I should share, though quite a few settings will be obvious to Asterisk gurus. The TELUS cheat sheet Asterisk PRI provisioning = T1 span timing is provisioned in /etc/zaptel.conf or /etc/dahdi/system.conf --- For syntax see [system.conf.sample](http://svn.digium.com/svn/dahdi/tools/trunk/system.conf.sample) ### Defining a span as esf/b8zs span=1,0,0,esf,b8zs ### Defining a span as sf/ami span=1,0,0,d4,ami Both have timing=0 . That is: that device sets the timing on the T1 line. Shouldn't it be '1' instead? PRI B/D channels are provisioned in /etc/zaptel.conf or /etc/dahdi/system.conf -- For syntax see [system.conf.sample](http://svn.digium.com/svn/dahdi/tools/trunk/system.conf.sample) bchan=1-23 dchan=24 PRI interface options are provisioned in /etc/asterisk/zapata.conf or /etc/asterisk/chan_dahdi.conf --- For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) [channels] ;--- PRI interface options ; Switchtype: Only used for PRI. ; ; national:National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess:ATT 4ESS ; 5ess:Lucent 5ESS ; euroisdn:EuroISDN (common in Europe) ; ni1: Old National ISDN 1 ; qsig:Q.SIG switchtype=national context=incoming-from-dahdi ; defined in extensions.conf signalling=pri_cpe ; should be pri-cpe for customer side ### Defining channel groups for selection Once you've defined groups in chan_dahdi.conf.sample they can be referenced in extensions.conf [channels] ;--- PRI interface options . . . group=1 ; reference for hunting method in extensions.conf channel=1-23 ; follows b-channel provisioning Note that anything after that 'channel=' line has no effect on those channels. ### TON / NPI For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; dynamic:Dynamically selects the appropriate dialplan ; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) pridialplan For syntax see [chan_dahdi.conf.sample](http://svn.digium.com/svn/asterisk/trunk/configs/chan_dahdi.conf.sample) * national : Interpret the digits as a national number. * international : A fully formed E.164 phone number. [channels] . . . ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for ; the dialed number. For most installations, leaving this as 'unknown' (the ; default) works in the most cases. In some very unusual circumstances, you ; may need to set this to 'dynamic' or 'redundant'. Note that if you set one ; of the others, you will be unable to dial another class of numbers. For ; example, if you set 'national', you will be unable to dial local or ; international numbers. pridialplan=unknown ; Asterisk default As of 1.6.0 . On 1.4 the default is 'national'. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
I have one of my 4 PRI ports RED (that`s normal: there is a cable from my DS3 MUX to it but there is no active PRI). I configured it for futur use, thinking that it could just start running when that particular PRI was activated without any asteisk modifications. Could this be the cause, or am I just grasping at straws? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Tuesday, September 28, 2010 13:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED Shaun, Thanks for all the help. I tried but couldn't see a definite spike anyways. I used mpstat to see the irq% for each CPU, and although it shows me spikes to 100% for one CPU (not always the same one is spiking irq%) the intr/s isn't actually spiking. It`s like suddently there aren't more interrupts, but the IRQ% time for the CPU shoots up. What can make it do that? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, September 28, 2010 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED On 09/28/2010 11:40 AM, Mike wrote: How do I go about finding which device is causing those interrupts? (to make sure it`s actually Digium hardware and not something else?) Can you let something like 'while true; do date interrupt_log.txt; cat /proc/interrupts interrupt_log.txt; echo *** interrupt_log.txt; sleep 10; done' Run and see IRQ line has a jump in total interrupts (on any CPU) about the time you notice the IRQ spike from your snmp log? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
On 09/28/2010 12:53 PM, Mike wrote: Thanks for all the help. I tried but couldn't see a definite spike anyways. I used mpstat to see the irq% for each CPU, and although it shows me spikes to 100% for one CPU (not always the same one is spiking irq%) the intr/s isn't actually spiking. It`s like suddently there aren't more interrupts, but the IRQ% time for the CPU shoots up. What can make it do that? Now we're back to my original assertion that I believe what you're seeing is related to the way % cpu time is calculated statistically via a timer interrupt. I would bet that you have a steady amount of cycles dedicated to servicing the interrupt handler but every 10 minutes there are a confluence of factors that affect the calculation. You might be able to use a tool like cyclictest [1] to see if there really is some inordinate amount of scheduling latency every 10 minutes as opposed to some sampling anomaly. [1] https://rt.wiki.kernel.org/index.php/Cyclictest -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
Shaun, Thanks, don`t think I don't appreciate all your time, I really do. So you`re saying it's possibly just fake smoke from a non-existent fire, and that nothing is actually wrong despite a high load one one of the CPUs? That`s what it really is: I am seeing smoke and am looking for a fire, maybe I'm just being paranoid. That`s what trending and SNMP alarms are for I suppose ;-) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, September 28, 2010 14:57 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED On 09/28/2010 12:53 PM, Mike wrote: Thanks for all the help. I tried but couldn't see a definite spike anyways. I used mpstat to see the irq% for each CPU, and although it shows me spikes to 100% for one CPU (not always the same one is spiking irq%) the intr/s isn't actually spiking. It`s like suddently there aren't more interrupts, but the IRQ% time for the CPU shoots up. What can make it do that? Now we're back to my original assertion that I believe what you're seeing is related to the way % cpu time is calculated statistically via a timer interrupt. I would bet that you have a steady amount of cycles dedicated to servicing the interrupt handler but every 10 minutes there are a confluence of factors that affect the calculation. You might be able to use a tool like cyclictest [1] to see if there really is some inordinate amount of scheduling latency every 10 minutes as opposed to some sampling anomaly. [1] https://rt.wiki.kernel.org/index.php/Cyclictest -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
On 09/28/2010 02:06 PM, Mike wrote: Thanks, don`t think I don't appreciate all your time, I really do. So you`re saying it's possibly just fake smoke from a non-existent fire, and that nothing is actually wrong despite a high load one one of the CPUs? That`s what it really is: I am seeing smoke and am looking for a fire, maybe I'm just being paranoid. That`s what trending and SNMP alarms are for I suppose ;-) Perhaps. It's certainly my best guess based on what you've said. Something else you could do without two much trouble to shift things around and perhaps change what you're seeing is force all the wct4xxp interrupt onto a single CPU and tell IRQ balance to not use that CPU for it's normal work. i.e. looking at the output from 'cat /proc/interrupts' that you provided before I see that the wct4xxp driver is attached to IRQ 177. So in /etc/sysconfig/irqbalance set: IRQBALANCE_BANNED_INTERRUPTS=177 IRQBALANCE_BANNED_CPUS=8 to prevent irqbalance from using CPU3 or trying to balance IRQ 177, and then echo 8 /proc/irq/177/smp_affintity To force IRQ 177 onto CPU3 exclusively. Restart irqbalance and after which the quad span should only be interrupting on CPU3 and the other three cores are free to handle all the other ones. It would be interesting to know if you still % hi spikes every 10 minutes like this. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
Hi Mike, I`m not sure, but I suggest the following tests. The cards generate interrupts even with no load. And in real life they have different clocks. A very small clock difference could accumulate and make the interrupts happen at same time. USB , TCE400 or PRI card interrupt routines could be having problems at this moment. 1) Separe usb interrupts from digium boards, try to do it by changing board slots. (or disabling usb in BIOS). 2) Try to run your system using one digium board at once to determine which one is generating the problem or if the problem occurs only with both present. Luis A P Barbosa 2010/9/28 Mike l...@net-wall.com (sorry for the series of email, I realize I seem to be having a discussion with myself). I was wrong, the problem isn't fixed. Is having IRQ spikes every 10 minutes (under no load at all) the norm with Dahdi hardware? Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 3:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - FIXED...? ...which, after upgrading to Dahdi 2.4.0 seem to be gone. Not sure if 2.3.0.1 was the culprit, or just the fact that sonething was stuck (stopping dahdi took about 5 attempts because it was apparently in use, even if Asterisk had been shutdown). Hopefully this isn't just the calm before the storm and Dahdi will behave under load. Thanks for the help Shaun, I appreciate you helping point me in the right direction. Hopefully this will help someone. Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 2:55 To: 'Shaun Ruffell' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Actually, after further investigation I found that every 10 minutes I get massive irq usage spikes. The CPU is busy servicing hardware interrupts. I do have a 4 port PRI card and a TCE400B card, but tha happens even when asterisk activity=0. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:58 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes When the asterisk process spikes, do you know if there is a corresponding spike in us, sy, ni, hi, or si time? On 09/27/2010 02:53 PM, Mike wrote: That`s exactly what I did, and exactly what didn't happen. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:48 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Just something like echo 1200 /proc/sys/net/ipv4/route/secret_interval should have been sufficient to half the frequency of the CPU spikes. On 09/27/2010 02:42 PM, Mike wrote: I just changed the file, did I need to reboot or do anything for the change to take effect? Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:34 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Hmm..ok, then I'm not sure I'll be able to help. It's just that I know I've seen issues similar to what you're reporting based on the route table flushing pushing some si time into the asterisk process (and it runs at 10 minute intervals by default). So I'm not as familiar with the Asterisk code base to know what sort of periodic tasks may have changed between the versions you reported. On 09/27/2010 02:23 PM, Mike wrote: That wasn't it. I wonder what happens precisely every 10 minutes in Asterisk (or that is accounted to Asterisk's CPU usage). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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:
Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
Thanks. Will try all that. Night time work again ;-) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, September 28, 2010 15:37 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED On 09/28/2010 02:06 PM, Mike wrote: Thanks, don`t think I don't appreciate all your time, I really do. So you`re saying it's possibly just fake smoke from a non-existent fire, and that nothing is actually wrong despite a high load one one of the CPUs? That`s what it really is: I am seeing smoke and am looking for a fire, maybe I'm just being paranoid. That`s what trending and SNMP alarms are for I suppose ;-) Perhaps. It's certainly my best guess based on what you've said. Something else you could do without two much trouble to shift things around and perhaps change what you're seeing is force all the wct4xxp interrupt onto a single CPU and tell IRQ balance to not use that CPU for it's normal work. i.e. looking at the output from 'cat /proc/interrupts' that you provided before I see that the wct4xxp driver is attached to IRQ 177. So in /etc/sysconfig/irqbalance set: IRQBALANCE_BANNED_INTERRUPTS=177 IRQBALANCE_BANNED_CPUS=8 to prevent irqbalance from using CPU3 or trying to balance IRQ 177, and then echo 8 /proc/irq/177/smp_affintity To force IRQ 177 onto CPU3 exclusively. Restart irqbalance and after which the quad span should only be interrupting on CPU3 and the other three cores are free to handle all the other ones. It would be interesting to know if you still % hi spikes every 10 minutes like this. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED
Thanks. The problem is I only have two PCIe slots, and they are both taken. I`ll definitely try disabling USB, I have no need for it right now. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Antonio Prata Barbosa Sent: Tuesday, September 28, 2010 17:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - NOT FIXED Hi Mike, I`m not sure, but I suggest the following tests. The cards generate interrupts even with no load. And in real life they have different clocks. A very small clock difference could accumulate and make the interrupts happen at same time. USB , TCE400 or PRI card interrupt routines could be having problems at this moment. 1) Separe usb interrupts from digium boards, try to do it by changing board slots. (or disabling usb in BIOS). 2) Try to run your system using one digium board at once to determine which one is generating the problem or if the problem occurs only with both present. Luis A P Barbosa 2010/9/28 Mike l...@net-wall.com (sorry for the series of email, I realize I seem to be having a discussion with myself). I was wrong, the problem isn't fixed. Is having IRQ spikes every 10 minutes (under no load at all) the norm with Dahdi hardware? Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 3:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes - FIXED...? ...which, after upgrading to Dahdi 2.4.0 seem to be gone. Not sure if 2.3.0.1 was the culprit, or just the fact that sonething was stuck (stopping dahdi took about 5 attempts because it was apparently in use, even if Asterisk had been shutdown). Hopefully this isn't just the calm before the storm and Dahdi will behave under load. Thanks for the help Shaun, I appreciate you helping point me in the right direction. Hopefully this will help someone. Mike -Original Message- From: Mike [mailto:l...@net-wall.com] Sent: Tuesday, September 28, 2010 2:55 To: 'Shaun Ruffell' Subject: RE: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Actually, after further investigation I found that every 10 minutes I get massive irq usage spikes. The CPU is busy servicing hardware interrupts. I do have a 4 port PRI card and a TCE400B card, but tha happens even when asterisk activity=0. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:58 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes When the asterisk process spikes, do you know if there is a corresponding spike in us, sy, ni, hi, or si time? On 09/27/2010 02:53 PM, Mike wrote: That`s exactly what I did, and exactly what didn't happen. Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:48 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Just something like echo 1200 /proc/sys/net/ipv4/route/secret_interval should have been sufficient to half the frequency of the CPU spikes. On 09/27/2010 02:42 PM, Mike wrote: I just changed the file, did I need to reboot or do anything for the change to take effect? Mike -Original Message- From: Shaun Ruffell [mailto:sruff...@digium.com] Sent: Monday, September 27, 2010 15:34 To: Mike Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Hmm..ok, then I'm not sure I'll be able to help. It's just that I know I've seen issues similar to what you're reporting based on the route table flushing pushing some si time into the asterisk process (and it runs at 10 minute intervals by default). So I'm not as familiar with the Asterisk code base to know what sort of periodic tasks may have changed between the versions you reported. On 09/27/2010 02:23 PM, Mike wrote: That wasn't it. I wonder what happens precisely every 10 minutes in Asterisk (or that is accounted to Asterisk's CPU usage). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com
Re: [asterisk-users] NAT issue (i think?)
Hello Ron.. The answer that i see here is only a trying to a Register...means the REGISTRATION procedures are taking a significant amount of time. You should get a 200 OK Can you lease make a simple draw of your architecture? seems to be a NAT problem, that's for sure REgards! 2010/9/28 Ron nha...@gmail.com Hi Danny On the pap2 by default it is set to 3600 and i have not change that. by the way, is the NAT keep-alive same with the NOTIFY message? coz i am seeing my asterisk respond to those as bad event could that be causing it to loose the registration? here's the registration from ngrep: U 78.65.34.12:5094 - 12.34.56.78:5060 REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport. From: Kristine sip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com ;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com sip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. Max-Forwards: 70. Contact: Kristine sip:456...@78.65.34.12:5094;expires=3600. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . U 12.34.56.78:5060 - 78.65.34.12:5094 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094. From: Kristine sip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com ;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com sip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. On 9/28/10 7:24 PM, Danny Dias wrote: You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ronnha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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 -- _ -- 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
[asterisk-users] Use modprobe to find E1/T1 jumper setting on PRI card
Does anyone know if I could use modprobe command to find out rather than set the jumper on a Digium PRI card? I want to find out the jumper settings on the card without opening the box which will cause down time. Thanks. -- _ -- 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
