similar to: voice choppy

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Hi, How can I make calls from Asterisk client to NEC NEAX 2400 traditional phone ? Is it possible to have a connection between Asterisk and NEC NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP. Please help me to find a solution ;;; Thanks & Regards Ram Kumar Customer Support Engineer Barcode Gulf LLC Dubai , UAE Mobile : + 971 50 5594178 Email : Ramkumar@barcodegulf.net

Greetings all, This is my first post to this forum, so please be kind. I am looking to integrate Asterisk as our primary voice mail server (IVS) for our NEC NEAX 2000 IVS. We currently use a 6-port Mitel system. The two communicate via analog TDM ports and a MCI serial connection. Is anyone familiar with setting up a NEAX 2000 with Asterisk as an IVR server? I know I would need six FXO and

hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable and the Digium TE405P using E&M wink signaling. the connection's ok. however when dialing from the NEC to the Asterisk. most of the time the Asterisk only sees the first digit of the dialed number(which is 4 digits). some time if i dialed the 4 digits very fast it might get through. seems like there's a timming

Hello, We use all SCSI PCI card hardware RAIDs on all 4 of our production Asterisk servers. They all have Digium quad T1 cards and they all have from 2 to 4 T1s hooked up to them. We have had no noticable problems with dropped calls/poor quality. What are you looking to do with this system? what kind of traffic will be going through these 4 T1s? MATT--- -----Original Message----- From: Deon

Hi I am trying to run a 'R Batch' job on unix machine but somehow it locks my dterm window. If I kill the window (or logout from my unix account) it kills the R job too. It does'nt seem to happen when I run a 'Splus BATCH' command on my unix machine. Can you please give me a fix for this problem? Thank you very much. Best Regards Vipul Devas Email: vipul.devas at

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

Is there anyone out there who has tried to connect up an asterisk box to make and take calls through a NEC NEAX 2400 using Q.sig or anything like it? Can anyone tell me if it is possible? Thanks in advance, Phil Allred Brooklyn Law School

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)------ADSL_ISP1-------ISP2--------Asterisk-----PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms

Hello, all... I've been trying for a week for transfer ~100 Gig of data using either rsync:// over IPSEC or tunneled through SSH, but it keeps failing... I have done as the documentation says and here are the results: I ran the following command: rsync -avz -vvv --dry-run --rsync-path=/root/bin/rsync-debug.sh -- delete-excluded --exclude-from=server-backup.exclusions --link-

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 -> 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts

Hi all, I was trying to rebuild mtr (http://vault.centos.org/6.7/os/Source/SPackages/mtr-0.75-5.el6.src.rpm) and I keep getting: ==== Executing(%install): /bin/sh -e /var/tmp/rpm-tmp.gu9Ds0 + umask 022 + cd /root/rpmbuild/BUILD + '[' /root/rpmbuild/BUILDROOT/mtr-0.75-5.el6.x86_64 '!=' / ']' + rm -rf /root/rpmbuild/BUILDROOT/mtr-0.75-5.el6.x86_64 ++ dirname

Hi, I have been trying to do numerical differentiation using R. I found some old S code using Richardson Extrapolation which I managed to get to work. I am posting it here in case anyone needs it. ######################################################################## richardson.grad <- function(func, x, d=0.01, eps=1e-4, r=6, show=F){ # This function calculates a numerical approximation

On 8 March 2016 at 16:15, Digimer <lists at alteeve.ca> wrote: > On 08/03/16 07:11 AM, James Hogarth wrote: > > On 8 March 2016 at 10:07, Leon Fauster <leonfauster at googlemail.com> > wrote: > > > >> Am 08.03.2016 um 01:50 schrieb Digimer <lists at alteeve.ca>: > >>> I'm not surprised, given that it is in the repo. That's why I was

Hello, I don't have any experience with channel banks and would appreciate any feedback on my theory outlined below: We have a single T1 entering the building with channels 1-12 being voice lines and 13-24 being a 768k internet connection. This T1 terminates to an Adit 600 (T1-1). Here's what I know. Channels 11-12 go out the Adit 600's 25-pair connector to a wiring block (and

Hello all. I am currently working on setting an S4 domain to replace our aging samba 3 setup. We have found many answers on the net, in various documentation, but when it comes to setting up beyond one node documentation becomes a little thinner. We are setting up a Primary DC with AD, using BIND9_DLZ, also serving dhcp from Primary, and we want to setup a Secondary that is both a