search for: res_timing_pthread

...like: > - Analysis of the RTP traffic (along with potential jitter) > - CPU utilization with an active conference (95% idle doesn't mean that >some core isn't pegged) > - Any potential transcoding issues or codec issues >Matt Hi Matt - thanks. Looks like I am ONLY loading: res_timing_pthread res_timing_dahdi But I dont think the res_timing(x) is working on CentOS 5. res_timing_timerfd does not even seem to be compiled on this box. How do I tell for sure what its using and if its good. All I saw in the asterisk log was the two res_timing_pthread and res_timing_dadhi being loaded. Ev...

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

...ing something much faster than it used to > > Cheers Duncan > > *CLI> module show like timing Module Description                              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 d...

...January 15, 2019 12:05 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.so DAHDI Timing Interface 4 2 modules loaded So what would you suggest? (And thanks in advance.) Thomas I've had some good experience with res_timing_dahdi both when we ourselves were stil...

starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these "timing" modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? sean

Folks, I've read some sources claiming that Asterisk does not need DAHDI for timing in 1.6.1. Is this true? Searching the web, all I can find is pages celebrating the fact but no actual documentation on which version it was introduced in and how one would go about configuring an external time source. I'm having a devil of a job trying to compile DAHDI on a hosted Xen VM and thought I

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

...ason the Music on Hold service is now using the DAHDI timing module because when I do "module show like timing" I see: CLI> module show like timing Module Description Use Count res_timing_dahdi.so DAHDI Timing Interface 33 res_timing_pthread.so pthread Timing Interface 0 2 modules loaded I believe that the pthread used to show a use count of at least 1 with the Music On Hold service using that timing source. I suspec that if I restart the Asterisk service everything will come back up the way that it did last time. However, I&...

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

...oot 15 0 3332 1112 572 S 0.0 0.2 0:01.14 crond 16282 root 25 0 4756 1008 820 S 0.0 0.2 0:00.00 safe_asterisk 22514 root 25 0 494m *445m* 6612 S 0.0 *87.0* 663:08.66 asterisk virtual1_ast1*CLI> memory show summary 4644 bytes in 2 allocations in file 'res_timing_pthread.c' 4096 bytes in 1 allocations in file 'chan_unistim.c' 484 bytes in 1 allocations in file 'res_clialiases.c' 96 bytes in 2 allocations in file 'devicestate.c' 244 bytes in 1 allocations in file 'iax2-provision.c' 660 bytes...

Hello, Is it possible to install Asterisk on Vmware(centos) from source. Is there any difference or disadvantage for this compared to asterisk running on physical machine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100811/05a14968/attachment.htm

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

...of the playback file. To eliminate encoding as an issue, I have only codec_ulaw/format_pcm loaded and the recording is ulaw. I've niced down the asterisk process to -19 even though I don't see asterisk taking more than 3% cpu. Is this behavior indicative of a timing problem? loading res_timing_pthread.so makes things horribly worse. i don't believe any other software timer is available for Solaris/sparc, right ? other thoughts ? Thanks, -- Jeremy Kister http://jeremy.kister.net./

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

...where the codecs of the called party leg were not properly sent back to the call leg when reinvited (issue #13569). * Fix broken attended transfers (issue #15183). * Add flags to chanspy audiohook so that audio stays in sync (issue #13745). * Resolve issues with choppy sound when using res_timing_pthread (issue #14412) Additionally, an update to chan_iax2 related to issue AST-2009-001 is included in this beta release. For more information, see: http://downloads.asterisk.org/pub/security/AST-2009-001.html For a full list of changes in this beta, please see the ChangeLog: http://svn.digium.c...

...ses issue #16982. Reported, patched by dmitri) * Send AgentComplete manager event for attended transfers. (Closes issue #16819. Reported, patched by elbriga) * Correct manager variable 'EventList' case. (Closes issue #17520. Reported, patched by kobaz) In addition, changes to res_timing_pthread that should make it more stable have also been implemented. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10 Thank you for your continued support of Asterisk!

...where the codecs of the called party leg were not properly sent back to the call leg when reinvited (issue #13569). * Fix broken attended transfers (issue #15183). * Add flags to chanspy audiohook so that audio stays in sync (issue #13745). * Resolve issues with choppy sound when using res_timing_pthread (issue #14412) Additionally, an update to chan_iax2 related to issue AST-2009-001 is included in this beta release. For more information, see: http://downloads.asterisk.org/pub/security/AST-2009-001.html For a full list of changes in this beta, please see the ChangeLog: http://svn.digium.c...

Hello, i just made the reproducible watching: I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via T.38 -> Audiocodes Mediant 2000 (FW 5.60.43.5) -> PSTN Fax With Internal timing Enabled, the Fax break after the first quarter from the first page is transfered. With Internal timing Disabled, the fax is transferred flawless. Both test with pthread timing module on a QEMU

...ses issue #16982. Reported, patched by dmitri) * Send AgentComplete manager event for attended transfers. (Closes issue #16819. Reported, patched by elbriga) * Correct manager variable 'EventList' case. (Closes issue #17520. Reported, patched by kobaz) In addition, changes to res_timing_pthread that should make it more stable have also been implemented. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10 Thank you for your continued support of Asterisk!

Hi, I used asterisk on Debian7 and it was good experience. Now, i'm using osx on mac mini. I'd like to install asterisk 12. I tried to compile it and after lot of searches, I got it. All sip accounts log in. I can call but I haven't any sounds. - for IVR, - for voicemail, - for out and in calling... Please help me... Thank you in advance.... AMICALEMENT? ______________ Manu -