search for: timerfd

Hi All Just finished setting up a vm with centos 5.5 and asterisk 1.8.3 Using timerfd as a timing source. Has anyone got a similar setup in production ? How's performance? Thanks, Neeraj?

Hello All, I'm sure this has been talked about and based on some searching of archives, I'd discovered that to be able to use timerfd, one needs to have a kernel version >=2.6.27? Is this true? If yes, then is there anyone who's got it working in Lenny 5.0.7? Do I need to download and build the linux kernel (currently at 2.6.37) from scratch and get access to the TimerFD source? Should I even bother with it for app_confBr...

I am currently having a voice quality problem with one of our Asterisk servers. We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions. I am looking at the timing source for Asterisk and it is currently using timerfd even though we have an E1 card installed. Is timerfd better than dahdi? Any recommendations to test if timing may be a problem for voice quality and DTMF? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hi all Is there a relation between the above? I'm having a problem where I suspect my internet access provider (through whom I go to a SIP trunk provider) have got MTU size problems. My asterisk (1.8.11.0) is constantly going into the situation where a TimerFD error is spammed in the CLI, load goes up and up until the system is completely unusable. I have an admission by the ISP that their MTU size on their fiber NTU "may" be incorrect. Can an incorrect MTU size on an ethernet connection cause Asterisk to experience errors with the kernel tim...

Hi Guys Regarding this I found the following links which appear relevant: https://issues.asterisk.org/jira/browse/ASTERISK-19347 https://issues.asterisk.org/jira/browse/ASTERISK-18223 It seems that this issue is related to that, NOT to a too-large MTU size. Don't know if anybody can comment? Is the MTU size a red-herring as relates the timer-fd errors? Thank you very much Stefan

...a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >> timerfd even though we have an E1 card installed. Is timerfd better than >> dahdi? Any recommendations to test if timing may be a problem for voice >> quality and DTMF? > > What is the scenario and the channels involved? Timing is only used > for things such as playback, music on hold,...

...a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >> timerfd even though we have an E1 card installed. Is timerfd better than >> dahdi? Any recommendations to test if timing may be a problem for voice >> quality and DTMF? > > What is the scenario and the channels involved? Timing is only used > for things such as playback, music on hold...

Hi guys I have some questions regarding the above 1. Why are there different options for timing? 2. What are the differences between these types of timing sources? 3. When should you use what? 4. Is one timer type more "reliable" for an Asterisk system under heavy loads than another while NOT using any DAHDI hardware - SIP only? I'm of course referring to the familiar

>Information on timing sources can be found here: >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces >As noted on that page, ConfBridge can use any timing interface Asterisk >provides, and is not limited to the DAHDI timing interface. Generally, >timerfd is a good timing interface. >That aside, I would try to rule out external issues with the garbled audio >before changing the timing source. Things like: > - Analysis of the RTP traffic (along with potential jitter) > - CPU utilization with an active conference (95% idle doesn't mea...

...iption                              Use Count  Status Support Level res_timing_dahdi.so            DAHDI Timing Interface                   0          Running              core res_timing_pthread.so          pthread Timing Interface                 0          Running          extended res_timing_timerfd.so          Timerfd Timing Interface                 1          Running              core 3 modules loaded     This will show you what module Asterisk is using for timing. You can try doing a noload on the two you do not need. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Cháv...

...0-0000170e [Mar 4 17:29:38] NOTICE[25790] app_chanspy.c: Attaching SIP/209-00001715 to SIP/210-00001714 [Mar 4 17:31:06] NOTICE[25812] app_chanspy.c: Attaching SIP/209-0000171c to SIP/210-0000171a [Mar 4 17:31:19] ERROR[25818] acl.c: Cannot create socket [Mar 4 17:31:19] ERROR[25818] res_timing_timerfd.c: Failed to create timerfd timer: Too many open files [Mar 4 17:31:19] WARNING[25818] channel.c: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n [Mar 4 17:31:19] WARNING[25818] chan_sip.c: Unable to allocate AST channel structure for SIP...

Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using "res_timing_dahdi" or I can use "res_timing_timerfd" to get some benefit if I upgrade to 1.8? thank a lot f...

...othesis. In the past this was a simple matter of running dahdi_test which would report the accuracy. I'm not sure how to interpret the results of "timing test" in the Asterisk CLI. If I increase the number of ticks per second the results are erratic while under load. I'm using the timerfd module in Asterisk with a 1000HZ tick kernel and high res timers enabled. I've tried both hpet and tsc as system clock sources, both exhibit the same breaks in audio. It sounds like someone presses the mute button in the middle of a sentence. Any insight is appreciated! Here are the specs on...

...ng the problem? I've looked at various upgrade instructions and haven't seen anything, but it's easy to miss details in the reams of info out there! In any case, the log looks like this: == Manager 'veco' logged on from 127.0.0.1 [Feb 10 14:14:36] ERROR[15569]: res_timing_timerfd.c:105 timerfd_timer_open: Failed to create timerfd timer: Function not implemented [Feb 10 14:14:36] ERROR[15569]: res_timing_timerfd.c:105 timerfd_timer_open: Failed to create timerfd timer: Function not implemented -- Executing [conference at veco:1] Answer("Local/conference at veco-...

Thank you for a quick answer, Dmitry! We have tried the settings you suggested but nothing helped. The machine is running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620 v3 @ 2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(. ??, 18 ???. 2017 ?. ? 10:49, Dmitriy Ermakov <demonihin at gmail.com>: > Hi, Kseniya! > > I have three installations of Asterisk (as FreePBX but I think it is not > important). They work fine. > > I have made some set...

Hi all I am using asterisk 11.24.1 on a centos 5 machine. kernel 2.6.18 flavor. (x86_64). I have about SIP 150 endpoints on it. when I send a message I'm getting garbled audio. I used to have a single PRI card in the box - but something happened and that connection no longer worked. I removed the card and also removed the system.conf and chan_dahdi entries. I am using ConfBridge in a PA

Hi all! Does anybody have experience with asterisk on Hyper-V? My test setup with Ubuntu 16 and asterisk 13.1 (ubuntu repo) shows sound distortion. I have analyzed the RTP flow with wireshark and I see high skew and delta values when the traffic leaves the hypervisor, however everything is okay when a capture is taken from a VM itself. I have read that there can be timing problems with Hyper-V. I

...you for a quick answer, Dmitry! We have tried the settings you suggested but nothing helped. The machine is running?4.4.0-104 kernel, 4 cores,?Intel(R) Xeon(R) CPU E5-2620 v3 @ 2.40GHz, clocksource is?hyperv_clocksource_tsc_page, timing module is res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(.? ??, 18 ???. 2017 ?. ? 10:49, Dmitriy Ermakov <demonihin at gmail.com <mailto:demonihin at gmail.com>>: Hi, Kseniya! I have three installations of Asterisk (as FreePBX but I...