similar to: Interface with NEC NEAX 2400

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

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

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

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

On 2002.10.19 19:44 Ruben I Safir wrote: NYLXS is having a watershed week this week and I'd like to personally extend my hope to everyone to participate and prosper from all of our activities. On Sunday, We are papering Long Island City and Astoria with Flyers for the Free Software Business Demo at LaGaurdia College. We are meeting at the front Entrance of the Queens Citicorp Building

I have a recording studio in Brooklyn NY and I am seeking help in some lessons and setting up. I have been learning linux and asterisk very well thanks to lurking in the group and info on the voip-info. I would like some one to come over and give me a tutorial, check my box out, make sure everything is secure and answer some really dumb questions that I have. In return I will make available

Hello I have a Free Software Fax Client I use for Samba based faxes called Respond. I need it altered a little to be a little more professional. I need it to pop a window to create a cover letter and attach it to the fax. I don't program in Windows, so I can someone paid to do this from a client. Please email if interested. Thanx! Ruben -- __________________________ Brooklyn Linux

This patch cleans up some 80-odd warnings I was seeing when compiling the xenpci driver, all completely trivial. It also replaces the use of asm for calling cpuid with the __cpuid compiler intrinsic, and modifies calling code accordingly. I''m pretty sure my mailer going to mangle the inline patch (below) so I''m attaching the pristine version. (will try to fix for next time) --

Hey Guys We've been using the respond program with SMB fax services and it's worked pretty good. But I have one need which is not addressed by this software, which I'm hoping maybe I can pay someone to hack into the windows code. I need Respond to send a cover letter prior to being sent out, independent of the program queueing the print job in windows. I was thinking I might be

>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() >If I assign the DID to ring extension SIP/2000 and then after time-out send >it to MeetMe() or Playback() it works and the caller

I posted a problem earlier thinking it was due to a lack of sound card. Several members stated that you do not need a sound card to play audio to a PRI channel. I did some further testing and discovered that there is a problem with call progress tones or signaling on my PRI. I think that the reason I am not hearing audio from the MeetMe() or Playback() apps. is because the the calling side of

----- Forwarded message from Brooklyn Linux Solutions CEO ----- >From ruben@www2.mrbrklyn.com Thu Mar 15 23:17:00 2001 From: Brooklyn Linux Solutions CEO <ruben@www2.mrbrklyn.com> Message-Id: <200103160417.f2G4H0O09806@www2.mrbrklyn.com> Subject: Some Users Work - Some don't In-Reply-To: <200103160226.f2G2Q1i09494@www2.mrbrklyn.com> from Brooklyn Linux Solutions CEO at

Can the logs be set up to monitor who is copying files accross the netwrok? Ruben -- __________________________ Brooklyn Linux Solutions __________________________ http://www.mrbrklyn.com - Consulting http://www.nylxs.com/radio - Free Software Radio Show and Archives http://www.brooklynonline.com - For the love of Brooklyn http://www.nylxs.com - Leadership Development in Free Software

Hello We've been running samba with LPD hooked up to two HP 5SI prionters through a deskject Direct card. Occassionally, it stop working. And the clients get an access deny messages which seems to be meaningless. This usually happens when the printer jam. But it doesn't seem to come back to a stable configuration either with the W2K drivers or the deskject direct cards. THe print

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

Hi, Is the stars command in the base package or do I need to download? I am looking to make star/radar/spider charts. Thanks! Britt Britt Aronovich Marketing Analyst BAM (Brooklyn Academy of Music) ... Peter Jay Sharp Building 30 Lafayette Ave. Brooklyn, NY 11217-01486 ... P: 718.724.8038 E: baronovich@BAM.org<mailto:baronovich@BAM.org> BAM.org Facebook.com/BAMstage

We have a basic application that runs a SIP channel to pick up a call and process it. We are using Broadvoice and it's been working great. We recently rebooted the machine and now when a call comes in Asterisk picks up the call but does not process it. Asterisk seems to send the call back to Broadvoice. Nothing at all has been changed in the configuration to warrant this. Below is the

On 08/10/2017 04:41 PM, Frank-Ulrich Sommer wrote: > I can't see any security advantages of a self signed cert. I then you fail to understand the history, like when Microsoft's certs were undermined because the third party authentication agency gave the keys to 2 guys that knocked on the door and asked for them... -- So many immigrant groups have swept through our town that

thatis not secure and you might as well use gmail It is not so hard to just get a static IP and put a mail server up. fit/pc has perfect boxes for this, run on about 15w/h power On Sun, Dec 10, 2017 at 02:42:36PM -0800, Kenneth Porter wrote: > On 12/10/2017 12:39 PM, Stephan H wrote: > >I have dovecot postfix setup on my home server as well. I use a virtual > >server in the