similar to: Answering Machine Detection

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hi - I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Hello Everyone, Can someone point me to reliable links on how to tweak Asterisk AMD I am calling a number and have to two files to play depending if it is a real person or an answering machine. Most everytime Asterisk calls it thinks it is an Answering Machine and it starts playing the AMD message, instead of the delivering the "1st real message" Any hints? Thanks in advance, -C

I've set-up a mirror at ftp.gbnet.net/pub/digium It's (sort of) available via www.gbnet.net/public Mirror's the whole of ftp.asterisk.org/pub Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19

This looks like a fun box... a Voip to Dect gateway, I've mailed them for pricing details.... < <http://www.computex.com.tw/news_archive_detail.asp?index=4053> http://www.computex.com.tw/news_archive_detail.asp?index=4053> Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen E: <mailto:info@betel.nl> info@betel.nl The Netherlands W:

Simple Q but I can't find the answer in the archives (and am too lazy to look in the source, but then its 32 Celcius here... Do all digium cards provide the zapata timing? e.g. also the analogs (including the X100P) or only the E1/T1 -ones or do I need to use ztdummy on the analog cards? Thanks, Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen

Hi people! Sorry for bringing up some annoying issue.. yes, it's AMD again... But I was searching the last days for a solution for my problem and didn't really find anything. Now I'm hoping that someone of you has maybe an idea for me. :) My setup: --------- I use the Asterik Manager API to generate outgoing calls (by using "Originate" messages). These outgoing calls

Hi all, I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a channel bank and a digium E1 connected to the PSTN. I get occasional warnings from asterisk: WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error 500 This happens mosttimes in a loop like this: [netland_helpdesk] exten =>

Does anyone know if there's public voice detection algorithms available? I've scoured the net for the last hour or so, and I can't come up with anything except a few proprietary or embedded solutions. I know dsp.c uses goertzel algorithms for DTMF detection, but how does one detect voice? I dunno, maybe detecting voice isn't the way to go. I want to begin playback of a file

I want to setup an Asterisk network using the Cisco ATA 186. What is the best place to order those devices? I'm not finding them anywhere. Al __________________________________ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree

Some switches provide the functionality to try a number till it becomes available. Thus when one dials a number and get a busy, one enters a *XX# code and the switch will call your extension when the called party becomes available. Has somebody already built this in/for Asterisk, otherwise I'll look into it. Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT

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

I just downloaded the latetst CVS. A compile now complains about a missing srv.c & srv.h used in chan_sip.c. Can they be added? -- Betel Consultancy Abelenlaan 19 1185 RT Amstelveen The Netherlands http://www.betel.nl tel. +31 621 858 469

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

Hi I'm trying to get ExtenSpy to work but it wont, I'm dialling a number from my mobile which comes into our server and answering the number on a particular SIP extension which all works fine. I'm then dialling an exten from my own SIP extension which executes the ExtenSpy for the correct extension but I hear nothing. Here is the output in the CLI -- Executing