search for: neax2400

...00 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. What follows are two examples from what I tried in extensions.conf: This works but is not desirable: [nec_pri] ; Digital PRI from the NEAX2400 exten => 2688,1,Wait,1 exten => 2688,2,Dial(SIP/2000,3,Tr) exten => 2688,3,Wait,1 exten => 2688,4,MeetMe,|Mps exten => 2688,5,Hangup This will answer, but there is no audible playback on the channel: [nec_pri] ; Digital PRI from the NEAX2400 exten => 2688,1,Wait,3 exten =>...

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 -------------- next part -------------- An HTML attachment was scrubbed... URL...

...elated. Any help is appreciated. Please see my two examples below: Unless my incoming DID (2000), is pointed to a SIP station that is registered and functional, I do not receive call progress tones on inbound calls. If I point the DID to an application like: [inbound_pri] ; PRI from the NEAX2400 exten => 2000,1,Wait,3 exten => 2000,2,Answer exten => 2000,3,MeetMe,|Mps exten => 2000,4,Hangup I will not hear any initial ringback, and once answered there will be no audio on the channel. If I point the DID to a registered SIP station like: [inbound_pri] ; PRI from the NEAX2...

...nswered 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 hears the .gsm file. >Any assistance in solving this problem is appreciated. > >[nec_pri] >; Digital PRI from the NEAX2400 > >exten => 2688,1,Wait,3 >exten => 2688,2,MeetMe,|Mps >exten => 2688,3,Hangup > I had a similar problem with my system, and I was able to fix the problem by executing Answer before I entered any other applications. Using your previous example: exten => 2688,1,Answer e...

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...

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 aster...

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

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