similar to: Asterisk to NEC NEAX

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

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

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

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

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) --

>-----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

as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

Topology: PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server When I make a call to a VoIP user from the PSTN, the call gets routed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my

Trying to install Java JRE, but getting this error: ronv at speedy:~/temp$ wine jre-1_5_0_16-windows-i586-p.exe err:service:validate_service_config Service L"Mscromw" has an unknown service type err:service:scmdatabase_load_services Invalid configuration of service L"Mscromw" - skipping fixme:sfc:SfcIsKeyProtected ((nil), (null)) stub fixme:advapi:RegisterEventSourceA

Forgive me if this question has been asked/answered in another post. And let me reiterate what other users have frequently said - Asterisk is great, and I really appreciate all the work you folks have put into it. How have some of you gone about integrating Asterisk with a legacy office PBX, such that the end-user can use a regular office (digital handset) and dialing is fairly seamless ? Our

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

I am currently attempting to get Asterisk working with our existing NEC IPK 192 system. The * box has a Wildcard X100P clone card in it and is hooked to an analog port on the NEC system. The analog port is shared between an elevator phone and a credit card machine. I have one SIP extension configured in * which the softphone on my Windows PC is registered to. With the softphone I am able to

Hi All: We have cooperation PBX as NEC, I am wondering how to replace our legacy ACD by using Asterisk, but not losing tie line connection to NEC. Does anyone have any idea how to make it work? Many thanks, Tilein

http://bugzilla.mindrot.org/show_bug.cgi?id=465 djm at mindrot.org changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution| |FIXED ------- Additional Comments From djm at mindrot.org 2003-02-24 15:46

I have an nec electra elite 192 with a t1 card; and am looking for suggestions as to integrating them (can't throw out the system yet!). I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart signaling, and 2 working bt102 grandstream ip phones (thanks again Matt for your "start from scratch"

I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with CCIS). Things work pretty well with the exception of issues on stations on the SV8300. When I call from Asterisk to a SV8300 station and I send my extension as the caller ID number, it shows up on the SV8300 as "OPERATOR". I've tried a few different TON/NPI values with no difference. If I set callerid to a

Hi, We are currently in the process of installing an Asterisk box between our legacy PBX (NEC Xen Master) and the Telco (Telstra). Originally the PBX was connected to the telco via 8x BRIs. Our new Asterisk box has been setup with 2 OpenVox B800P cards. Each card has the 4 ports in TE mode and 4 ports in NT mode. The NEC PBX is connected to the Asterisk boxes NT ports and the Asterisk box is