similar to: Toll fraud detection/password script

Has anyone created like a broadcast dialplan, if so care to share it. What I'd like to do is create an extension so when someone calls that extension they can leave a voicemail. Right after it is recorded, I need that voicemail to played on all phones on that system... E.g.: 1) Administrator --> Dial special number 2) Record emergency message (e.g. Snow day don't come in) 3) Hang up 4)

Subject says it all... A Simple Asterisk Based Toll Fraud Prevention Script http://www.infiltrated.net/asterisk-ips.html Ramblings for admins/engineers to think about. Doesn't have to cost you umteen thousand dollars for stuff like IPS/IDS. Although a little on the crude side, quite effective. If you care to dabble with MySQL you can create quite an impressive hosts based IPS that is custom

On Dec 10, 2007 10:36 AM, Jeng Yu <jengyu2007 at yahoo.co.uk> wrote: > Thanks, All! And thanks, Oquendo! I will experiment > with this suggestion. I was actually thinking in terms > of a situation where it would be done > non-interactively. > > Jeng > > > --- "J. Oquendo" <sil at infiltrated.net> wrote: > > > Jeng Yu wrote: > > >

Hey all, when you guys have requests from clients to block their CID from showing through, what are others doing? I had a coworker throw in some "Name Here"<0> garbage which none my carriers like. I don't want to do "Private"<12345678910> so any suggestions. -- ==================================================== J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE

Curious to know if anyone's created something similar to the following, if so and you'd care to share an AGI or dialplan, much appreciated. I will be eventually write a script to answer the nTH call. (if I can't find it (why reinvent wheels). Looking to do some testing sending anywhere between 50-200 calls to a machine. I'd like a Snom/Polycom/whatever to pick up after the nTh

Quick and dirty too hungover to think/search... ;) Sorry list AsteriskBoxA -> SBC/PSTN voodoo -> world wide Interweb -> AsteriskBoxB Account on BoxA UsernameJohn<12125551212> Account on BoxB UsernameJohn<102> They're both the same users, had to do some funky trunking (managed firewall provider is playing not in my backyard games)... So anyway, I need to take specific

My Gurus! I'm still playing with asterisk in the lab here. There is a feature that I need in a production asterisk system. I was wondering if it already exists in asterisk. When we want to shutdown a production asterisk system, we would like the shutdown to happen after there are no more calls being processed. In other words, a shutdown command that does the following: - block asterisk

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of J. Oquendo > Sent: Thursday, April 26, 2007 6:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Asterisk brute force watcher (was FYI) > > Steve Totaro wrote: > > I suspect that

Apologies for cross posting but some of us aren't on the other list (vice/versa) and thought both groups would benefit. For those familiar with the VoIP Abuse Project, no need to explain the gist of this. I got tired of parsing through the alerts (lists) I receive via email daily. They're long and sometimes I don't have the time to post them all. So for now, posting VoIP Abuse

Hey all, having an extremely odd issue wondering if anyone else has come across or seen something similar and what your resolution was. I have Asterisk 1.2.12.1 running (don't ask) on a machine. All has been working fine for months on end. The system has a mixture of Polycom, Snom's and Cisco 7960's running. After a brief power outage last Friday, most phones went down but the PBX

Without diving into too many details, does anyone have a simple callback script that does the following: Caller --> Dial Asterisk --> "In order to place this call please enter a callback number to place this call for your pin..." Caller --> Enters DID to call back for pin Asterisk --> stores a number to either a flat file or DB. (From here I can take care of the rest)

/usr/sbin/asterisk -rx "show channels concise"|awk -F : '($11 > 10800) {print "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""}'|sh You don't necessarily have to keep restarting it at midnight. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP "Enough research will tend to

/usr/sbin/asterisk -rx "show channels concise"|awk -F : '($11 > 10800) {print "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""}'|sh You don't necessarily have to keep restarting it at midnight. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP "Enough research will tend to

Hey all, I have a couple of questions. 1) What is the maximum amount of registrations and ongoing calls you've been able to achieve on your Asterisk systems. Please do not respond with marketing hyperbole. I'm looking for real world implementation in the thousands range. For instance, max I recall having on one souped up server was a couple hundred registrations with no more than 70

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

I have an interesting observation which I thought I'd pass along to save other people from spending time trying to 'fix' it. One of my clients uses Charter's so called "business phone service". They provide 'analog' phone lines over IP. In general, they've worked OK. End users were saying that the phone are "cutting out" at times. What

Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061122/054eb688/attachment.htm