search for: payphones

I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones

I've got a dumb Western Electric payphone and some homebuilt hardware to control the coin relay which is accessible to Asterisk through the AGI interface. I'd like to be able to set the state of the coin relay to collect at the end of a call if a called party answers. [Hey, I admit this project is being persued just for the fun of it ] Looking through the documentation, there is a way to

Hello I have a 800 DID setup to dial into my Asterisk server and I'm wondering if it's possible to ID when it's a payphone or not? I suspect it's not since I'm getting calls from someone else's SIP or IAX box. If I had a digium card installed and connected to a couple lines would I be able to get this information and parse it? Thanks, Jess

...xperience with PRI delivery of ANI II information? Specifically, I want to know if it's possible from within Asterisk to know if the inbound call (which may or may not be to an 800 number) came from a payphone or not. I know with some 800 providers it's possible to block inbound calls from payphones (due to the FCC surcharge etc) but was wondering how accessible that information is once the call hits my box. Thanks in advance Paul

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

...200,1004/300 unobtainable = 400 ring = 400+450/400,0/200,400+450/400,0/2000 callwaiting = 400/100,0/4000 ; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000 ; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it ; is and I've not used a payphone for years) creditexpired = 400/125,0/125 ; These two are used to confirm/reject service requests on exchanges that ; don't do voice announcements. confirm = 1400 switching = 400/200,0/400,400/2000,0/400 ; This is...

Great idea Dean, "I would also suggest that maybe instead of looking to set up within the disaster zones that you consider setting up in the relocation zone (eg where people are being sent to) yes they have payphones there but having a bank of 20 grandstream phones connected a T1 is a far smarter and more effective solution and probably more meaningful." > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of De...

...successfully drilled being the lock cylinder itself), and one on the door to the circuitry (which included the programming port to set the per-call rate for use with a standard subscriber line, instead of the dedicated pay lines, as well as the coin-counter electronics).? They were used on many payphones twenty years ago or so.

I am looking for the most stable version of Asterisk to use with ASTCC for a production environment. It does not appear that any of the Stable versions will be suitable since they do not support US PRI ANI Info digit collection and hence could not apply surcharges for payphone use, etc. Is there a specific CVS Head version date that includes the II updates and has proven to be stable enough to

I've just been doing some tests using the manager API to originate an outgoing call via a X100P and connect the call to an extension: Action: Originate Channel: Zap/1/01234567890 Context: local-extensions Exten: 6000 Priority: 1 I've noticed that extension is getting called as soon as the outgoing call has been placed, rather than when it is answered. Is the X100P capable of detecting

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: > [incoming] > ; incoming calls from the FXO port are directed to this context from zapata.conf > > exten => s,1,Answer() > exten => s,2,Dial(SIP/polycom) And zapata.conf: >

A client wants to put phones in a semi-public place, using a Calling Card solution. What kind of hardware is suitable? I'm looking for a wall mounted booth, a SIP phone that can't easily be broken, but I might just use cheap analog phones and a channel bank. What do you suggest for the calling card software? This installation will be outside the US. -- Chris Mason NetConcepts (264)

On Tue, November 28, 2017 9:21 am, Lamar Owen wrote: > On 11/27/2017 02:02 PM, m.roth at 5-cent.us wrote: >> Pete Biggs wrote: >>> - don't run ssh on 22, use a different port. >> I consider that pointless security-through-obscurity. > Security through obscurity it may be, but it isn't pointless. Tarpits are in a similar class; they don't help with security

I haven't managed to work this one out yet, so any assistance appreciated ... We want to be able to set the outgoing caller-id on, BRI according to the extension but haven't worked how with asterisk ? we have several hundred inbound numbers on these BRI so we are able o use any one these to sett on outdial. One other point I have been told should work, bu have no way of trying... In

We are using SPA-2002s and PAP2Ts to service our VoIP customers. It seems that one of our customers (female) has a voice that is just right that it generates DTMF tones when she talks... I know I've seen this sporatically, but this seems to happen often on her line, and I'm curious if there are any settings on the device to alter this behaviour?

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

I am using asterisk 1.6 at home and would like to send incoming calls without caller id immediately to voicemail (i don't want to use the privacy manager where people have to enter a number). The config examples i found are all for the pretty obsolete 1.0 and 1.2 versions of asterisk. Would anyone be willing to share a config example? Thanks!

Dear all I am now going to do some text analysis using R. However, the data is very noisy that I need to clean it first. I don't have much experience in the text cleaning process. Is anyone would provide help on this? If you are able to provide some similar code which was done before would be greatly appreciated. May content is mainly the Feedback data through *Phone call record*: