search for: outpulsing

Hey All, AT&T is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have

Hi, I want to make sure I'm clear on something. All of the phone systems we have setup have been either analog lines, or PRI lines. We have a customer who is going to use a T1 line with 4-digit outpulse. Do I need to do anything special to get Asterisk to recognize the outpulse, or will it just work like DNIS does?

We are in the prossess of ordering a ISDN (PRI) line from Verizon. There are a couple of questions on the application that I don't understand. I'm still a newbie here but we're taking the big plunge. If someone could give me some direction here I would be greatly appreciative. The questions on their app that I don't understand are: 1.) The Verizon Co will outpulse 10

We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both) What is the number of digits

Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me:

We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk

Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and "Advantex" the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the

Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me:

I'm ordering a PRI T1 for use with Asterisk and a Digium Wildcard TE405P. The provider is asking me a number of questions about how I want to configure the line. Here goes: 1. Dial Tone - No, Yes - Precise, Yes - SCC 2. Framing - SF, ESF 3. Line Coding - AMI, B8ZS 4. Signaling Type - Ground Start, E&M, Loop Start w/Ring, Loop Start w/o Ring 5. Pulse Mode - DTMF, MF 6. Outpulse Start -

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>

Hi, I'm seeing rather odd behaviour on a new box with TE110P card. I'm running the TE110P span with ccs,hdb3,crc4 in pri_net, connected to a second machine with a TE410P in pri_cpe. The span is idle. I'm using pri intense debug span 1 and can see the RRs going back and forth. So - things are running along with the span showing "Provisioned, Up, Active" in pri show

...list. - enter a Class of Restriction group number in COR Group Number (this determines which extensions *cannot* access this trunk, I am using a COR that permits all Mitel extensions to access the Sip trunk and therefore call Asterisk users successfully) - enter any required digits in Digits Before Outpulsing. (If this field is left blank, digits will be sent out as "Enbloc".) Route number = 10 Routing medium = SIP Trunk Trunk group number = empty SIP Peer profile = Asterisk Route Type = PSTN access via DPNSS - ARS Digits dialed Assignment: Digits Dialed = 2 (2 is the first digit of my Aster...

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number