similar to: Use of "timing test"

Hello, We use Asterisk extensively for conferencing - for the last 8 years or so this has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 350 concurrent users. Right around that number DAHDI hit's a hard coded memory limit and kicks allocation errors in the log. [Jun 22 10:04:13] WARNING[9095] app_meetme.c: Unable to open DAHDI pseudo channel: Cannot allocate

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

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 >>

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

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

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?

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 >>

Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [root at bin]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference:

Hi I wonder if anyone has any sugestions I have had a TDM400 for a couple of years, and I have always had problems with noise on the line, so tonight I have been doing some research and have found that if I load the CPU dahdi_test has almost perfect results and no noise dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.997% 99.999% 99.998% 99.997% 99.999% 99.998% 99.998%

Hello, I am trying to get a Wildcard TE110P to work in a relatively modern HP Proliant DL385p Gen8 server. Being a potent 12 core Opteron server I expected no problems. Much to my dismay the dahdi_test results are constantly terrible: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 89.101% 89.195% 89.142% 88.957% 88.953% 89.115% 89.089% 89.134% 89.066% 89.021% 88.933% 89.044%

Hello, Does anyone know how to test the timing device? I've tried the following but with no luck. Zaptel is installed. I'm trying to use ztdummy as a timer. [root at TemplateAsteriskServer ~]# dahdi_test Unable to open dahdi interface: No such file or directory [root at TemplateAsteriskServer ~]# zttool Unable to open /dev/zap/ctl: No such file or directory Thanks Dan

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?

Hi Eric, > On 12 avr. 2016, at 17:48, Eric Cooper <ecc at cmu.edu> wrote: > > On Tue, Apr 12, 2016 at 04:36:58PM +0200, Bertrand LUPART - Linkeo.com wrote: >> I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : >> [...] >> However, i can't find any dahdi startup script, neither init.d neither systemd

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

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

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]

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

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <div class="moz-text-flowed" style="font-family: -moz-fixed; font-size: 13px;" lang="x-western">Hi, <br> <br> I try to find a solution for this problem : <br>

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really