similar to: Timer_fd, pthreads, or DAHDI timer for timing under 1.8.11.0?

Displaying 20 results from an estimated 1000 matches similar to: "Timer_fd, pthreads, or DAHDI timer for timing under 1.8.11.0?"

2010 Jul 16
4
chan_local - Asterisk 1.6.2.6
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
2015 Feb 12
1
1.8.11.0 - CLI error res_timing_timerfd
Hi all Sometimes (about every three months) some of my Asterisk 1.8 boxes will start running this message thousands of times in the CLI: [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12
2013 Oct 02
2
Dahdi_dummy is more accurate than core timer?
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
2016 Apr 05
3
Best timing source?
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
2016 Apr 06
2
Best timing source?
On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> 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 >>
2016 Apr 05
5
Best timing source?
On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> 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 >>
2011 Mar 15
1
Ast 1.8_CentOS5.5 with timerfd as timing source
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?
2012 Jul 23
2
Intel D2500cc video problem: no scrolling
Dear colleagues, in the process of upgrading my home router I'm trying to utilise Intel D2500CC mini-ITX motherboard[1]. Both stable/9 and head suffer from no scrolling issue: from the beginning of booting the kernel, all output overrides the last line on the screen, while others keep the loader phase. Provided the board has multiple serial ports and the final target (headless
2015 Jan 07
3
Asterisk executable suddenly about 40KB larger - modules not working
Hi all I have a strange issue with 1.8.11.0 on a production Asterisk machine at our head office, and the same issue with a production machine at a branch office. Every now and then, on the head office machine, ODBC CEL and CDR logging will stop working. On examination in the CLI, Asterisk behaves as if the config files for ODBC in the /etc directory are just gone. Repeated tests have then
2013 Nov 04
1
[LLVMdev] DominanceFrontier/PostDominanceFrontier for PRE
On Sun, Nov 3, 2013 at 9:19 AM, Christopher Wood <christopherwood07 at gmail.com> wrote: > On Sun, Nov 3, 2013 at 1:02 AM, Daniel Berlin <dberlin at dberlin.org> wrote: >> >> As for a "better" way to implement PRE, it depends on what algorithm >> you want to use. If you just want to write a PRE pass, that's easy >> enough without dominance
2016 Nov 11
6
Asterisk 11.24.1 garbled audio
>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
2014 Dec 12
1
Corrupt MixMonitor recordings - .gsm format
Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted
2015 Mar 02
1
System() command refuses to execute bash script
Hi All I'm using this extension to try and get Asterisk 1.8.11.0 to run a bash script: exten=>802,n,System(/bin/sh -f /root/wireless.sh) This file is -rwxr-xr-x 1 root root 171 Mar 2 16:23 wireless.sh e.g. root owns the file, and it has execute permissions for all users. Asterisk runs as root as well. Asterisk executes the command without any errors at max verbosity. The file
2019 Jan 14
2
Various extensions ring once and go to voicemail
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
2009 Feb 14
1
Asterisk 1.6.x timing API
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
2015 Feb 19
0
TimerFD errors if MTU size is set incorrectly - SIP trunk
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
2011 May 06
1
is res_timing_timerfd module stable in 1.8?
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
2012 Mar 29
0
Asterisk 1.8.11.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: *
2011 Jan 24
3
Asterisk on Debian Lenny with timerfd
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?
2015 Aug 13
2
One way audio - doesn't seem to be NAT issue
Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would