similar to: ways to get more accuracy from ztdummy

Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

Another request make me test my t1 card, which has no quality problems, but all that I get is: [root@SIP2-MI zaptel-1.2.1]# ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

I have some DID numbers that come into Asterisk via PRI, then connect to a Panasonic DBS PBX via PRI. Outbound calls from the PBX go out via PRI to Asterisk, then out to the Telco's PRI. The sync LEDs on my PBX show that it is synced to Asterisk via the PRI. I have users complaining about random hangups. What is the best way toi approach finding the issue? ref: zttest looks good. ./zttest

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff

> output of zttest: > Opened pseudo zap interface, measuring accuracy... > 99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.951172% > 99.951172% Ah. Don't be scared. This is actually -correct- when using RTC. Let me demonstrate: [root@dhcp5 zaptel]# ./zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8195 sample intervals 99.963379% 8192

I just needed to test a dialup modem connection (don't ask) and I had a modem connected to a TDM card (FXS port) which then dialled out via a E1 PRI on a TE4xxp card. See my log below: atdt0198xxxxxx CONNECT 36000 V42bis ** Dial IP ** Username: xxxx Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATH0 OK ATDT0198xxxxxx CONNECT 31200 V42bis ** Dial

HiI don't have any digium cards and only want to install ztdummy with all asterisk functions. What I will need with ztdummy and what I can disable?Thanks a lotDmitri************************************ Zaptel Module Selection ************************************* Press 'h' for help.

I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk "gets busy" and uses

Hi, I'm still strugling with getting an easy to use conference system implemented. I did have app_conference running, but today I upgraded asterisk to 1.0.9 and it stopped working. I've tried following the instructions for compiling app_conference on 1.0.7 but it didn't work. So I went back to ztdummy (I've not had any luck getting this to compile on FC2). Anyhoo, I've

I've modified ztdummy to work under 2.6 (basically ditched all the uhci stuff and added a kernel timer instead). How do I test my changes are doing anything useful? zttest gives: --- Results after 14 passes --- Best: 99.975586 -- Worst: 99.975586 Is that good/bad/terrible...? Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle <tmh@nodomain.org> Key ID:

> I'm still having my (apparantly) timing problem, but I'll do some more > testing and make a separate thread for that. I'm generating an > outbound call through Asterisk. The inbound audio is good, but the > outbound audio is sometimes staticy. This seems to happen only at the > start of the call. These are short test calls, just playing a "weasels > ate our

I'm using asterisk 1.4.11 and Zaptel version 1.4.5.1 with kernel 2.6.22. I have the ztdummy module loaded, which is using zaptel and rtc. When the ztdummy module is loaded, sounds are not heard when using the asterisk "background" command. When the ztdummy module is unloaded, which also removes zaptel and rtc, sounds are heard. I've also tested this under kernel 2.6.21 with the

We are having intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option

I've written a Smalltalk application which starts R on another machine and then communicates with it via sockets. The functions involved are: slaveStart <- function(masterHost, masterPort){ inPort <- availablePort() sendSocket(as.character(inPort), masterHost, masterPort) socketSlave(inPort) } socketSlave <- function(inPort){ ## listens on inPort.

Hello, can someone explain these errors Oct 27 15:34:05 vhost01 named[1316]: zone #ZONE#/IN/auth: refresh: retry limit for master IPV6-MASTER#53 exceeded (source IPV6-THIS#0) Oct 27 15:34:05 vhost01 named[1316]: zone #ZONE#/IN/auth: Transfer started. Oct 27 15:34:06 vhost01 named[1316]: zone #ZONE#/IN/auth: refresh: retry limit for master IPV6-MASTER#53 exceeded (source IPV6-THIS#0) Oct 27

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have