similar to: Using Blacklist

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

Hi All, In theory if I do this; exten => s,1,Zapateller(nocallerid) exten => s,2,Privacymanager exten => s,3,Dial(a bunch of SIP extensions) My callers should only hear the anti-telemarketing tones if they call from a line that has no caller*ID and then get offered an opportunity to enter it, right? What I'm finding is that in the event of no CID the caller gets dumped into the

Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: "One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the "high

Been tinkering and found Zapateller appears to be answering when I didn't expect it to. I have something like so: [incoming] exten => s,1,NoOp ;Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Dial(${RING},20) ... I have a 1x1 analog * installation with a couple IP phones too. I've got the FXO interface connected to the home phone line. When I get an

Hi all, I'm looking for some rudimentary insight on GotoIf() which seems to be failing on me in my dial plan. All I basically wish to do is block a particular caller. Sounds easy enough, but my ternary operator/plan currently is not properly being implemented. Can anyone spot where I'm being a momo? All extensions get forwarded to the following macro: [macro-forward] ; arg1 = phone

I am trying out a new service from www.talkn.com. They use Sipura to terminate your service like most providers. They are looking at directly connecting into asterisk in the future. Right now my configuration is Talkn?Sipura?Asterisk/FXO Card. When someone calls in and get?s answered and when either party hangs up , * releases the port with out a problem. If someone calls in and * answers the

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>

My Asterisk fax detection used to work, but no longer does. OK. So, here's the deal: 1. It appears that the "faxdetect" command cannot be applied channel-by-channel in zapata.conf anymore, as Asterisk appears to the last "faxdetect=" command to ALL channels. 2. My stations are detected and sent to the proper extension; i.e., when I send a fax from one zap extension

Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

Andi - I have experienced the same issue you mention and gotten no reply as to a way to fix it. I finally implemented "blacklist" into my Asterisk and added "Anonymous", "anonymous", "unknown", "Unknown", etc., into my blacklist file. When those come in with an IP address instead of a phone number but have no real name, they get the

Hi list! I'm trying to get PrivacyManager working but for some reason it always thinks that CallerID is present (when it isn't). I get this on the console: == Primary D-Channel on span 1 up -- Accepting voice call from '' to '0711234567' on channel 0/2, span 1 -- Executing Ringing("Zap/2-1", "") in new stack -- Executing

Hi. I'm new to Asterisk and have been working on setting up a development server but have gotten myself a bit confused. I'd like to implement the following logic for calls coming from the PSTN: Check for caller-id yes => keep going no => play SIT and prompt for telephone number Check time of day to see if it's day / night day => ring some phones

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

I'm looking to bock a list of numbers users cant call. Is it possible to pull these from file specified in the dial plan, as apposed to mysql?

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)

Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tried again. Would you believe it worked fine. Has anybody else come

Hi all, I'm looking for an easy way to lookup numbers from the database so I can fork calls from my daughters friends onto her IP phone/answering system. I'm looking for something very similar to LookupBlacklist, but I'm already using LookupBlackist to filter out telemarketers. What I'm doing now is adding multiple exten=> lines to my extensions.conf file to match those

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0