similar to: Is it possible to ID payphone calls?

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

Anyone have a suggestion on how I can have my asterisk box make two SIP or IAX calls and bridge the two together? It seems like it would be easy to setup but the only way I'm finding seems to be setting up meetme rooms. Jess

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

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 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

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

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

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"

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

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

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: >

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 everyone, Can anyone tell me if there's a command prompt on R1.5.0 Windows that will bring up the GUI dialog box that performs the same function as setwd() only interactively? I've hunted around, but can't seem to find anything. Something like file.choose(). Best wishes, Jess _____________________________________ Jess Mar Department of Mathematics University of Queensland QLD

Hello R'ers, I have a large set of data which has many y samples for each unit x. The data might look like: Seconds Response_time ---------- ---------------- 0 0.150 0 0.202 0 0.065 1 0.110 1 0.280 2 0.230 2 0.156 3 0.070 3 0.185 3 0.255 3 0.311 3 0.120 4 .... and so on When I do a basic plot with type=l or the default of points it obviously plots every

My samba shares aren't working with ACL access since the upgrade. Here is an example of the error from /var/log/messages Dec 14 12:44:17 gfm-atlas smbd[14365]: [2006/12/14 12:44:17, 0] smbd/service.c:make_connection_snum(911) Dec 14 12:44:17 gfm-atlas smbd[14365]: '/usr/GFM_Shares/Users/Receiving' does not exist or permission denied when connecting to [GF_Receiving] Error was

Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD

Anyone else having problems installing this package? Any ideas for fixing? Thanks, Jess -- View this message in context: http://r.789695.n4.nabble.com/Forecast-Package-for-2-15-0-tp4601328.html Sent from the R help mailing list archive at Nabble.com.

Hi, Using the functions in the tcltk library (R1.5.0 Windows, tcltk v1.5.0), does anyone know a simple way to create a list box with a scroll bar? I'm stuck on what to supply to the -xscrollcommand in order to do this but I might be on the wrong track. Best wishes, Jess _____________________________________ Jess Mar Department of Mathematics University of Queensland QLD 4072 AUSTRALIA