similar to: monitoring VM via speaker and grabbing connection

Reason. I have a fax/ans phone with handset, that lets you monitor the caller, so if you wish, you can pickup the call. The asterisk is undergoing testing, it will then be online tested at the house so I can get more familiar in setting components up, e.g. sip phones, voicemail, transfers etc. But, I really need the monitoring of the Voicemall being left, with the ability to pick up the

Hi everyone, I'm in the very early stages of rolling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any

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

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

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi i have an @home box with Uniden UIP200's and the speaker phone works great for recording my voicemail box and recording the digital receptionist menu but when i make calls out the PSTN the mic on the speaker phone acts like it only picks up when my voice is really high, make a noise and get closer to the mic all the sudden it starts working then if i back off it quits and visa-versa, but I

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

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

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

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

AstriCon 2009 is still 5 months away, but the time will fly! We're looking for speaker presentations for 2009's conference - do you have something you want to talk about? Submit your talk proposal today - the proposal window closes on June 1. AstriCon is the single largest conference dedicated to Asterisk applications and business, and this year we hope to have even more

Hello All, I've just been made aware of Asterisk and have installed it. Things seem to be working: I can dial from the console the internal numbers and hear the answerphone messages and do other interesting stuff. however when I hangup the call I see a continuous stream of messages saying "Sound: Recording overrun" appear on ALL my linux consoles and I can't see what I'm

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

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

Hi all! Using C6 on an Asus M5A99X EVO R2.0 motherboard with AMD six core FX6300. That board uses Realtek ALC892 audio chipset. For the first time since I got that motherboard I tried plugging in headphones to the front panel speaker output. I was somewhat displeased to note that the speakers don't mute when that happens. I recall seeing info on that in Fedora, but not Centos, and

Hello, I noticed that for one specific Speaker, there are codebook entries in all codebooks, that "fit" the speaker. So if one had a look at a histogram of the used codebook line numbers for one speaker, the histograms would look very much the same for different speech samples (of course, the speech samples should be long enough, more than a minute of speech ought to be sufficient). I

This message is long overdue. I've commented on similar issues when other people have mentioned similar items, but I am finally taking the time to to ask this question. Did anyone else have the sound from their internal speaker stop working with the release of CentOS 7.2? I have tested this with multiple machines with the same card and the same issue. The card is identified as: !!PCI

We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. The phone lines at church are also used by our private school during the week. So I am thinking that I need to setup a schedule so that the lines used for the service will

Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card. I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best }