Re: [asterisk-users] Asterisk choking on voice messages announcements
On 04/21/2010 07:13 PM, bruce bruce wrote: How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? What version of DAHDI are you using? If you're using a relatively recent version (2.2.0+) dahdi_dummy works better on virtual machines because it uses the wall time to adjust itself as opposed to just assuming it's going to be called at a fixed interval. But, one issue with this is that if the host machine is unable to keep accurate wall time then dahdi_dummy can be confused about how much time is really passing. The big jumps in accuracy make me think that NTP may be adjusting the host clock in several second increments? The firewall changes aren't interfering with NTP or anything like that are they? Just a thought... -- 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
[asterisk-users] Asterisk choking on voice messages announcements
Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. 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
Re: [asterisk-users] Asterisk choking on voice messages announcements
Are your sound files being transcoded or played back in their native formats? On 04/21/2010 12:25 PM, bruce bruce wrote: Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Asterisk choking on voice messages announcements
Yes, it's all g.711 ulaw. On Wed, Apr 21, 2010 at 1:37 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: Are your sound files being transcoded or played back in their native formats? On 04/21/2010 12:25 PM, bruce bruce wrote: Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Asterisk choking on voice messages announcements
Are you running asterisk in a virtual machine? -- _ -- 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 choking on voice messages announcements
yes, it's on Amazon. On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock rrb3...@gmail.com wrote: Are you running asterisk in a virtual machine? -- _ -- 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 choking on voice messages announcements
Then use a timing source if the version is correct (1.6.1 or 2), or install dahdi-dummy, which can be quite some amount of work On Wed, Apr 21, 2010 at 12:35 PM, bruce bruce bruceb...@gmail.com wrote: yes, it's on Amazon. On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock rrb3...@gmail.com wrote: Are you running asterisk in a virtual machine? -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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 choking on voice messages announcements
So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- 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 choking on voice messages announcements
Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.com wrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- 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 choking on voice messages announcements
Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com wrote: Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.com wrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- 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 choking on voice messages announcements
On 22 Apr 2010, at 00:36, bruce bruce wrote: Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Thats.. Interesting... -- _ -- 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 choking on voice messages announcements
On 04/21/2010 05:36 PM, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.com mailto:rrb3...@gmail.com wrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 What in the world? Bruce, that is a measure of accuracy of your timing source. I believe that is the issue. What is this running on? -- _ -- 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 choking on voice messages announcements
It's running on an Amazon instance. No changes to system made and it was working find previously. Here is an output of top: [r...@ip-10-251-123-3 ~]# top top - 19:59:48 up 6:52, 1 user, load average: 0.78, 0.95, 0.99 Tasks: 49 total, 2 running, 47 sleeping, 0 stopped, 0 zombie Cpu(s): 0.0%us, 0.0%sy, 0.0%ni, 98.7%id, 0.0%wa, 0.0%hi, 0.0%si, 1.3%st Mem: 1740948k total, 399504k used, 1341444k free, 105300k buffers Swap: 917496k total,0k used, 917496k free, 161544k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1 root 15 0 2132 752 648 S 0.0 0.0 0:00.05 init 2 root RT 0 000 S 0.0 0.0 0:00.00 migration/0 3 root 34 19 000 S 0.0 0.0 0:00.00 ksoftirqd/0 4 root RT 0 000 S 0.0 0.0 0:00.00 watchdog/0 5 root 10 -5 000 S 0.0 0.0 0:00.00 events/0 6 root 10 -5 000 S 0.0 0.0 0:00.00 khelper 7 root 11 -5 000 S 0.0 0.0 0:00.00 kthread 9 root 20 -5 000 S 0.0 0.0 0:00.00 xenwatch 10 root 10 -5 000 S 0.0 0.0 0:00.00 xenbus 17 root 20 -5 000 S 0.0 0.0 0:00.00 kblockd/0 19 root 20 -5 000 S 0.0 0.0 0:00.00 kseriod 52 root 25 0 000 S 0.0 0.0 0:00.00 pdflush 53 root 15 0 000 S 0.0 0.0 0:00.02 pdflush 54 root 20 -5 000 S 0.0 0.0 0:00.00 kswapd0 55 root 20 -5 000 S 0.0 0.0 0:00.00 aio/0 671 root 10 -5 000 S 0.0 0.0 0:00.19 kjournald 695 root 10 -5 000 S 0.0 0.0 0:00.00 kauditd 720 root 18 -4 2380 672 424 S 0.0 0.0 0:00.23 udevd 1439 root 12 -5 000 S 0.0 0.0 0:00.00 kmpathd/0 1445 root 12 -5 000 S 0.0 0.0 0:00.00 kmirrord 1463 root 10 -5 000 S 0.0 0.0 0:00.00 kjournald 1719 root 17 0 2392 572 288 S 0.0 0.0 0:00.00 dhclient 1804 root 18 0 10576 1040 752 S 0.0 0.1 0:00.34 rsyslogd 1808 root 25 0 1772 416 352 S 0.0 0.0 0:00.00 rklogd 1829 root 15 0 6948 1072 688 S 0.0 0.1 0:00.24 sshd 1858 root 25 0 2640 1208 1040 S 0.0 0.1 0:00.00 mysqld_safe 1916 mysql 15 0 118m 19m 4904 S 0.0 1.1 0:00.47 mysqld 1957 root 15 0 9480 1860 784 S 0.0 0.1 0:00.00 sendmail 1967 smmsp 18 0 8260 1488 632 S 0.0 0.1 0:00.00 sendmail 1976 root 18 0 24728 7612 4636 S 0.0 0.4 0:00.11 httpd 1992 root 18 0 3072 1128 584 S 0.0 0.1 0:00.00 crond 2005 asterisk 18 0 25476 7296 3568 S 0.0 0.4 0:00.09 httpd 2006 asterisk 15 0 25496 7300 3556 S 0.0 0.4 0:00.04 httpd 2007 asterisk 15 0 25816 7364 3596 S 0.0 0.4 0:00.11 httpd 2008 asterisk 20 0 29348 9876 4432 S 0.0 0.6 0:00.04 httpd 2009 asterisk 15 0 24888 5244 2092 S 0.0 0.3 0:00.09 httpd 2010 asterisk 17 0 25496 7300 3540 S 0.0 0.4 0:00.08 httpd 2011 asterisk 17 0 25480 7344 3572 S 0.0 0.4 0:00.07 httpd 2012 asterisk 15 0 25496 7252 3516 S 0.0 0.4 0:00.03 httpd On Wed, Apr 21, 2010 at 7:56 PM, Sean Brady sbr...@gtfservices.com wrote: On 04/21/2010 05:36 PM, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com wrote: Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.comwrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want
Re: [asterisk-users] Asterisk choking on voice messages announcements
On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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 choking on voice messages announcements
I know that anything lower than 99% is bad. But *-400 *? Anything care of comment? Thanks, On Wed, Apr 21, 2010 at 7:45 PM, Steve Howes steve-li...@geekinter.netwrote: On 22 Apr 2010, at 00:36, bruce bruce wrote: Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Thats.. Interesting... -- _ -- 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 choking on voice messages announcements
How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? P.S. there is only few calls load at anytime on this server. Thanks On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 choking on voice messages announcements
On Wed, 21 Apr 2010, bruce bruce wrote: It's running on an Amazon instance. No changes to system made and it was working find previously. Maybe you could correlate the fluctuations in your timing source with the attacks on Randy and Fred's systems. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 choking on voice messages announcements
On Wed, Apr 21, 2010 at 6:13 PM, bruce bruce bruceb...@gmail.com wrote: How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? P.S. there is only few calls load at anytime on this server. Here are few ideas: 1. I have seen complaints that as Amazon loads up its virtual machines, that neighboring VM's running on the same hardware are sucking up CPU cycles and reducing the performance of the other VM's on board. One guy was complaining that to get the same performance he got a few months ago, he has to move to a more powerful machine, which costs more . You might move up to a more expensive, faster VM and see if it helps. 2. I don't know exactly how Dahdi gets its timing, but I do know that it has two methods; one involves HIGH RES TIMERS compiled into the kernel. The other when the high-res stuff isn't included. You can decompress /proc/config.gz into a local file and look for HIGHRES to be defined. If it isn't you might try to find a kernel with it defined, and see if it helps. 3. If you are on 1.6.1 or 1.6.2 (too tired to look up which), you could try using another method of generating timing than dahdi_dummy. I suspect that they may just reflect code already in Dahdi_dummy... but this seems like something you might want to become knowledgeable about! murf Thanks On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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
