similar to: Can I emulate SIP presence for an extension?

I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a "telemarketer hold" extension which, if I transfer a caller into it, loops around playing music & "please wait" messages, until they give up & hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if

Does such a beastie exist? I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000 respectively), and found them both to be seriously lacking - regular crashes (especially the F3000), poor battery life, and poor reception in particular. However, whilst SIP phones are great, I'd really like an IAX2 phone if there is one, as I can make that work "natively" though

Hi... I have what is, I am sure, a relatively common & straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a "distributed" PBX. Here's the scenario: [MASTER] is in the office. It has unrestricted access to the internet, and a fixed IP address. It has 3 SIP hardphones configured & working, plus a

Hello-- I've been playing with the privacy options on my home/home-office system since August last year, and have some results, gleaned from my CDR records, which over the last 13 months, number a total of 8672, which includes incoming, as well as outgoing calls. Before I start spitting out numbers, let me note that with the current setup, I haven't had to tell a single telemarketer

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

I'm curious is it possible to direct a call to an extension that takes you straight to music on hold, but NOT the standard music on hold. The boss suggested something he wondered if it was possible. Example: Someone calls (Telemarketer), we answer tell them to hold while we 'redirect' them to extension (Someone Important) 666 which is a separate music on hold pool of mp3's from

In my blacklist I have: database show ... /blacklist/Manitoba : advertising ... [incoming] ; First, lets take care of telemarketers exten => 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten => 4,n,Set(goaway=${CALLERID(number):0:2}) exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1) exten => 4,n,GotoIf($["${goaway}" = "V3"

Hi, I have the following requirements I'd like to implement with asterisk: 1. Asterisk notifies interested PC's on the network that there's an incoming call so that the telemarketing app can bring up the customer automagically 2. If a telemarketer makes a call and the customer isn't there and they arrange a callback, the callback is diverted to the originating telemarketers phone

Hi. I am trying to set up my Asterisk box to block anonymous calls. I am having some grief from telemarketer calls to my number and I would like to block it. I see from my CDR's that some of my callers also have "unknown" in their FROM field. I would like to let them through. Only block the FROM "anonymous" that the telemarketers use. Have anyone here done it and

Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c:

Hi folks, I've been playing around with an Asterisk server in my office for a few weeks now, and I've got it pretty much nailed down the way I want it, which is nice. One of the features I'm using is the ability to switch different contexts in & out of the dialplan on a schedule. So, for example, I've got the "official" tel number ringing my desk phone between

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this..... - -- James Finstrom -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -----END PGP

does anyone have a recorder GSM file that emulates the Telco's "if you are a telemarketer please hangup now" recording? I don't see one in the sounds dir. the ZapATEller works great for computerized callers but if a human hears this message asking them to go away they have to. Isn't that right? Dave

Hello-- I submitted of extensions.conf that contains my "telemarketer torture" menus, last week sometime to the mailing list. I got back a note from the mailing list machinery, stating that it was too big, and would be subject to approval. No such approval came, I guess. Either I missed it, or it didn't rate, or the moderator just plain hasn't gotten around to it yet. So, I

Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone

Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? -Adam

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you

Hi, Long story short - I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the number of pulses the dial sends (it could be a squiffy dial, I'm not sure). Not to mention the fact that, in today's modern "want