search for: res_timing_dahdi

...ystem is version 1.8.7.0. NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so. Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind? SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did watch -n1 date and watched the time tick up, perfectly synchronized to my mobile phone. It might be off by a second or so, I'd have a hard time knowing...

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

...erhaps voipinfo > > I don’t know if you can slow the VM processor speed? I am guessing it > is counting 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     ...

...is 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. Everything else is...

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: 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 for information!! Regards, tbskyd

Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING[22024] app_meetme.c: Unable to write frame to channel Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better results compared to res_timing_timerfd.so). On an idle system (Ce...

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]:

...for the last 40 weeks. I'm wondering of for some reason 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 w...

...based attended transfers. (Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett). NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Fix regression that changed behavior of queues when ringing a queue member. (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) Additionally, this release has...

...ETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. (Patched by tilghman) * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Resolve a memory leak when the Asterisk Manager Interface is disabled. (Reported internally by kmorgan. Patched by russellb) * Reimplemented fax session reservation to reverse...

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

...based attended transfers. (Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett). NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Fix regression that changed behavior of queues when ringing a queue member. (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) Additionally, this release has...

...ETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. (Patched by tilghman) * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Resolve a memory leak when the Asterisk Manager Interface is disabled. (Reported internally by kmorgan. Patched by russellb) * Reimplemented fax session reservation to reverse...

Running asterisk 1.8.4.2, and occasionally we'll have a call drop and the SIP retransmit error show up on the console. I actually think the retransmit error is just a symptom of something else, possibly centered around a timing issue. I tried res_timing_dahdi and that worked for about a week, then suddenly things went haywire and calls wouldn't last for more than a few seconds before being dropped. A restart didn't help. I switched over to res_timing_timerfd.so, and it seems to be working ok, but I'm worried this will happen again after so...

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to

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, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered