similar to: 1.8.11.0 - CLI error res_timing_timerfd

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

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

We are running load capacity tests using Amazon AWS configurations. For the tests, we are basically scaling up calls to a second Asterisk box. First box that is calling the second box plays music on hold for 60 seconds, then hangs up the call. My initial thought was jitter problems, but that doesn't seem to be the case. I believe I found the cause while looking at the asterisk logs. I am

Hi, we have a 2 node / 3 node RAC installation with OCFS. We have the following observations. 1. TIME STAMP Issue We have noticed that the time stamp which is shown on the datafiles doesnt remain the same even after a shutdown normal /shutdown immediate i.e If I shutdown all RAC instances ( A , B , C) using shutdown normal / immediate, the timestamp on the datafiles are not the same. Even

>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

Hello List! My Asterisk stopped making SIP-calls today, I could call from external, and saw Call coming in over PRI, but calling the SIP/Device wont work. I saw 5 open channels - all chan_spy. Only a restart helped. In the messages-file i found from yesterday: [Mar 4 17:28:01] NOTICE[25769] app_chanspy.c: Attaching SIP/209-0000170fto SIP/210-0000170e [Mar 4 17:29:38] NOTICE[25790]

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

Im setting up a very basic rails app and have a question about the needed migrations & associations. Basically, my app is an occasion reminder service that emails users when occasions that they select or input are coming up. Occasions will be selected from a checkbox type list or alternatively manually input by the user (name and date of the occasion). Users can have_many occassions.

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Dear R-users, I have written the following code to generate some trellis plots. It works perfectly fine except that it is quite slow when it is apply to my typical datasets (over several thousands of lines). I believe the problem comes from the loops I am using to subset my data.frame. I read in the archives that the tapply function is often more efficient than a loop in R. Unfortunately ,

On several occassions lately, my home Asterisk box has stopped registering with my VoIP provider. I haven't been able to reproduce the problem, and the log doesn't contain anything useful. How can I increase the log verbosity for SIP registration-related events? I've looked through logger.conf and tried searching with Google, but I haven't been able to find a clear answer. This

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the "Forward" softkey just before she enters the "Page All" keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have

Dear R-Gurus: This afternoon I was installing the "Hmisc" package. I use R in Linux (Fedora Core 1 (yarrow)) on a Compaq presario with 128 MB RAM laptop. Opening an R session as a root user (superuser), I issued install.packages("Hmisc") and waited. R downloaded the package, installation was going on, and on the standard output I could see that a list of

I''m working on a Rails project that needs to be deployed in the near future. To enable prompt responses to bug reports, me and my coding partner were thinking to use the following SVN repository lay-out/policy. We''re using trunk/ for our main-line development. Whenever we deploy something from our trunk, we first want to make a branch (e.g. branches/1.x) and then derive a tag

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

I am using CentOS 6, kernel 3.10 from elrepo.org kernels (3.10.102-1.el6.elrepo.x86_64). Asterisk version is 11.21.2 and Asterisk 13.X.X (I can't get it's version now). Is it possible that your network switches' interfaces which are connected to Hyper-V Server are 100% busy? It is possible that my installation works well because my Hyper-V server is not high-load server so it has

I have had two users on two separate systems indicate that they could "not hear a new message" When I investigate I find that the user has marked a message for deletion. The .WAV .wav and gsm files are gone but the .txt file remains thus giving asterisk and the user the impression that a new message exists when it does not. Has anyone else encountered this issue? Is there a fix?

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have