similar to: question on staRt package

Hello! I'm curently using * with two old Teles PCI card (wich, btw, were hard to install and make good use of) with ISDN4Linux. The sound quality is simply perfect. However both dialing in and out through the ISDN line, there seems to be a _little_ bit of echo that eventually gets on your nerves ! Also the echo seems to get a _little_ bigger after a minute or so into the conversation. Now,

On Tue, Sep 01, 2015 at 11:40:14PM -0300, Raphael S Carvalho via Syslinux wrote: > On Tue, Sep 1, 2015 at 8:58 PM, Paulo Alcantara via Syslinux < > syslinux at zytor.com> wrote: > > > Hi, > > > > My name is Paulo Alcantara. I've been working on BIOS/UEFI firmware and > > file systems development for a long time already. For those does not > > know,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello! After LOTS of research on this list and internet in general I managed to get an old Teles PCI card working with Asterisk throught ISDN4Linux. No echos, no delays, simply perfect -- electronic poetry ! :) eheheheheh I just didn't get it to work with CAPI and "chan_capi" but, since isdn4linux is doing such a good job, I'll

Hi, i'm traying to compile asterisk on my pc, a laptop whit SUSE 9.1 and a desktop with SUSE 8.2, with a teles S0 16/3 PnP. With Kernel 2.4 (Desktop) Asterisk run but it's umpossible to compile the driver ISDN-utils for Teles. With kernel 2.6 I can't compile zaptel (not necessary with my laptop) and asterisk, in both cases I receve errors during make or make linux26 (I saw the notes

hi PSTN <--> [Teles ISDN / Asterisk] <--> SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Setup: * Teles 16.3 ISA ISDN card with hisax kernel module *

Hello, folks! This is my first post here. I installed Asterisk from scratch and after reading a lot of information on voip-info and this mailing list I was able to get started. I can make sip-to-sip calls (just on a basic extentions structure, let's say for beginners) but now I'm trying to make this system works with my Teles ISDN BRI PCI card. I can make and receive calls through X-Ten

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help?

Hi all, I'm struggling some time now with this problem. Googling and searching on this topic did not deliver the answer yet, so my last hope is this list. Analogue to the things which are possible with modem.conf, where I can configure the MSN's to act on, I would like to have similar functionality. This is the idea: I have 1 ISDN line, it can be reached by 4 different MSN's. I have

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

Hi I'm using a Teles ISDN passive card configured in modem.conf. when i make call from my sip client (xtex x-pro) to the external world i have more than 1 second of delay and echo very. There is some tuning to do? The performance is better with an active ISDN card or CAPI compatible driver? thanks mark balester

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-----[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from

Dear Sirs, I am an economist, and I have learned about R from an statician. I would like to get some instructions on which packages of R should I install. I am intending to use R for graphical analysis, correlaction, and also estimation - this latter the least, since I have an econometric package which performes it very well. I should also mention that I am going to work mostly with

Hi! I need your advice. My problem is that I have very bad sound quality calling to cellular phone via asterisk router. There are some kind of noises and echo effects when you try to speak louder. I have the following components in my call routing schema: - PBX with E1 port. - asterisk router with TE405P card(32bit/4 E1 ports). - Teles server with PRI interface card(3 E1 ports) and VTM

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

Thomas Hellstrom will be handing over VMware's maintainership of these interfaces to Deep Shah. Signed-off-by: Deep Shah <sdeep at vmware.com> Acked-by: Thomas Hellstrom <thellstrom at vmware.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index e64e5db31497..c9bdbb65e96b 100644 --- a/MAINTAINERS +++

Thomas Hellstrom will be handing over VMware's maintainership of these interfaces to Deep Shah. Signed-off-by: Deep Shah <sdeep at vmware.com> Acked-by: Thomas Hellstrom <thellstrom at vmware.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index e64e5db31497..c9bdbb65e96b 100644 --- a/MAINTAINERS +++

Aprovados lista publicada Jacu?pe: Tangar? da Serra: ANA CAROLINA PINTO COSTA, LISLY KATELLY DE PAULA MARTINS, FRANCISCO HELSON DE LIMA NERES, PAULO RAFAEL PEREIRA SOARES, JO?O CARLOS MOREIRA DE CARVALHO, DAMI?O JOVENAL DOS SANTOS, MARIA GORETTI LIMA FREIRE, JANIMERY BARBOSA DE ABREU MELO. SHYSLAINE ARA?JO BEZERRA, ARIANE SOARES SILVA, LUCAS MOREIRA DIAS, GILSON POLICARPO DE S?, REBECA DE FREITAS

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from