search for: outpulse

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 any idea what she was referring to? Is this ANI? Outgoing Caller ID? Something else? While I've done many POTS line setups, this is my first PRI install, so I'd also welcome any &...

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?

...uple 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 digits to the PBX, is this satisfactory? 2.)DNLS Digit Dialing - Prefix and Delete - (if yes, what will it be?) It's also asking for the PBX manufacturer / system model number and the PBX software / release. Should I just fill in Asterisk and the realease for these questions? Lastly,...

...k 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 outpulsed? Is there a version number on the TE110P card? Thank you in advanced to anyone who might be able to answer these questions for me. I tried to explain to Verizon that it would work and that the system is flexible but they want to make sure of it before they setup the circuit. Michael ------...

...the TE110p is RJ45 - correct?) Dial Tone: None | Yes-Precise | Yes-SCC Framing: SF | ESF (I'm assuming ESF) Line Coding: AMI | B8ZS Signaling Start: Ground Start | E&M | Loop Start w/ring | Loop Start w/o ring (which of these does kewlstart deal with?) Pulse Mode: DTMF | MF (I assume DTMF) Outpulse Mode: Wink | Immediate | Seizure (If Seizure, then Origination or Digit Collection) Will ANI delivery be required for Toll-Free service? (I'm assuming Yes if we want to pass our caller id?) Thanks a ton for your time, JT

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 1.4.19. Watching the...

...ng to the PBX from the Central Office (CO) Asending (ASEQ), Desending (DSEQ), Most Idle (MIDL), Least Idle (LIDL) 10. Direct Inward Dial to PBX - DID (Y/N): - Number or Number Blocks routed toward PBX (i.e. 972-668-20xx) - Porting Coordination Required (Y/N)? - Number of digits to outpulse (None, 4, 7, 10 digits) to PBX? 11. Inband Ringing required from the CO when terminating calls to PBX (Y/N)? 12. NON-PRI DS1Switching - Start Signaling and Digit Pulsing for In-Band Signaling for NON-PRI DS1: - CO Incoming Start Signaling for calls originating at the PBX (ISTARTSG)...

...the TE110p is RJ45 - correct?) Dial Tone: None | Yes-Precise | Yes-SCC Framing: SF | ESF (I'm assuming ESF) Line Coding: AMI | B8ZS Signaling Start: Ground Start | E&M | Loop Start w/ring | Loop Start w/o ring (which of these does kewlstart deal with?) Pulse Mode: DTMF | MF (I assume DTMF) Outpulse Mode: Wink | Immediate | Seizure (If Seizure, then Origination or Digit Collection) Will ANI delivery be required for Toll-Free service? (I'm assuming Yes if we want to pass our caller id?) Thanks a ton for your time, JT

...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 - Wink, Immediate, Seizure 7. If Seizure then - Origination, Digit Collection. I'm not sure of the correct answers to any of these. I do know that I want to be able to get caller ID and the number dialed for the application that I'm building. I think from other things I've r...

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

hallo everyone, fyi ... working SIP Trunk configuration between Asterisk and Mitel 3300 ICP attached. let's refine further, please test and share your feedback, regards, Joseph Okoegwale Abuja, Nigeria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070709/4f7f15e1/attachment.htm --------------

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