similar to: TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?

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: >

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

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

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

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

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

OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a "w" in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w5555555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping

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

Does anyone know if R has a function for splines under tension. I know there are numerous packages for spline interpolation within R i just can't find one that lets you determine the tension factor. Any help would be much appreciated! Sam -- View this message in context: http://r.789695.n4.nabble.com/Splines-under-tension-tp2173887p2173887.html Sent from the R help mailing list archive at

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

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)

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

Hi: I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with kewlstart signalling. When an outside caller calls the server, the TDM-400 goes off-hook and provides a ringing tone to the caller. If the caller hangs up before the receiving party answers the phone, Asterisk fails to detect the hang-up. The TDM-400 stays off-hook, hogging the line, while Asterisk rings the

I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power will this drain from the 12 and 5 V connector when all ports are in use? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . -------------

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-----

Hi Is there a test that you can do to test your modules FXS FXO to see if they are blown I have a TDM400 with 2 FXS (green) modules on the the first two ports and 2 FXO (red) modules on the last two ports, but I can't seem to assign a channel to them. Can any one help me?? Thanx Jaco

What is the difference between these to cards? Any feed back good or bad would be great. Jonn

Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can

On 11/28/2017 12:04 PM, Valeri Galtsev wrote: > Thanks, Lamar! that is very instructive. You're welcome. > I was always unimpressed with > persistence of attempts to make more secure (less pickable) cylinder cased > locks (precision, multi-level, pins at a weird locations/angles). The best way to make an unpickable lock is to make the tolerances of the pins and the cylinder

Using a TDM400P in the UK nearly works fine, but I have a last remaining problem in that if the incoming is ringing and then the caller hangs up, asterisk takes another couple of rings before it spots the hangup. This is annoying in that I sometimes get phantom calls late at night (possibly due to call waiting or the exchange doing a half ring to see if we are live). Also I get phantom calls