search for: 75ms

Hello, Working a few servers that are transferring data across country with a 75ms delay on a GIGE connection. We can tune the tcp buffers on linux to improve the connections using iperf. Does rsync use the tcp buffers of the OS or does it override these settings? Thanks, Neal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.samb...

Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while

I have been in the process of trying to troubleshoot a phone system that is doing IAX trunking to a provider. The average RTA is 75ms with spikes from time to time and jitter from time to time as well. My question is this; How much can one trust this types of samples when going over the internet? I mean who knows who is doing what kind of ICMP rate limiting or dropping ICMP all together? What is a good measurement or troubleshoot...

I'm experiencing touch tone recognition issues when calling some outside phone systems. For instance, if I call my Nextel phone, and try to press * to enter my voicemail, Nextel's system does not "hear" the DTMF tone. I've also experienced other outside phone systems for which I am unable to use their touch tone menus. Oddly, this isn't the case with all outside systems.

Hello, Currently if you install a package twice: install.packages("testit") install.packages("testit") R will build the package from source (depending on what OS you're using) twice by default. This becomes especially burdensome when people are using big packages (i.e. lots of depends) and someone has a script with: install.packages("tidyverse") ... ... later

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

Hi. I wrote in a reply to a mail on here recently that you can''t set mpu (minimum packet unit) on HTB as you can on CBQ. I''ve just noticed that there is a patch on devik''s site which does mpu and overhead. http://luxik.cdi.cz/~devik/qos/htb/ For dsl users mpu is, for practical purposes going to be 106 - overhead is still variable though, depending on packet size.