search for: payphon

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...

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...

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

Quickie: Does anyone out there have experience 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

...25.153.150:5060;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Max-Forwards: 68 From: "39937841 39937841" <sip:39937841;cpc=payphone at 192.168.225.2:5060 ;user=phone>;tag=02e3a8c0-33807b-t-2 To: <sip:D39539988120 at 192.168.225.2:5060;user=phone> Call-ID: 2915b6e4-02e3a8c0-0000be53 at 192.168.225.2 CSeq: 2 INVITE Contact: <sip:10.1.1.10;line=sr-N6IAzBMsz.MwzxPfPxFsMJZfWBc7MBVuOBV-W.y6MxV*> User-Agent: NetCentrex...

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 second window to...

...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...

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...

...d others, such as the Kaba dimple locks used on Cisco Metro 1500 DWDM gear for power switches (the lasers are powerful enough to permanently damage your eyes in short order in those). > Whereas > there exists "disk based design" (should I say Abloy?), ... I had an old Bell System payphone with Abloy locks.? Very difficult to bypass or pick, and requiring very different techniques than are used with pin-tumbler locks.? There were two locks: one on the coin door (activated four large rectangular bolts, one on each side, with the only common point that could be successfully drille...

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 run in production? Thanks for the help. Karl Putz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-us...

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>

...el => 4 Pretty straightforward stuff -- a call comes in on the PSTN line, the Asterisk answers the call, then rings the extension. The caller hears a ring tone throughout the entire process. The rub is that Asterisk has, in reality, taken the PSTN line off hook. Not great if the caller is at a payphone. What if nobody answers the extension? The caller is out his money (50 cents in most of the US, 35 cents in Alberta and 25 cents in the rest of Canada ;) ) So I had the idea of doing things a bit differently, like so: > [incoming] > ; incoming calls from the FXO port are directed to this c...

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

...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 Australia, caller-id NAME is not used except to indicate the calling service such as PAYPHONE, MOBILE, RESTRICTED etc.. I have been told that we should be also able to set NAME on outbound as well as number, yet funny enough have yet to find any commercialsystems which do this, so I wonder if the information is true ? Gary .

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?

...TSP 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" <sip:xxx6002182@xxx.yyy.34.201;isup-oli=70>;tag=as6fbc6e76 When going out through level 3, they sometimes strip the isup-oli tag. I've tried 3 other ITSP's, by calling the MCI test number 18889996365, and they must be stripping it as well, because they have all also failed....

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!

...To <g-ccdfa at adfae.com> cc <g-ccdfa at adfae.com> Subject ase Mewrr asdffID:dde_20120624_15988015_11653024 * (keep this part)* CUSTOMER DETAILS Name : Mr dffa Company : da Address : ff Home No. : Office No. : Payphone Ext : Mobile No. : Fax No. : Email : CASE DETAILS Division : * dsaf (RIM) (keep this part)* Category 1 : * dsaf (RIM) (keep this part)* Category 2 : * dsaf (RIM) (keep this part)* Category 3 : Veh Reg Num : COMMENTS 24/06/2012 09:15:23 AM (Name) -...