similar to: Answer Machine Detection

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

Hi, I have IVR and when I press 1, asterisk calls my mobile phone. If my mobile phone is offline, asterisk transfers to asterisk voicemail. I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail. For that, I used AMD. So I have problems ! Asterisk detects answering machine everytime! How do I do please ? extensions.conf

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Julian

Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering

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 am having a bit of a problem getting AMD to work on a new server. On my regular office server it works like a charm. I am running Asterisk 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and I am using a SIP trunk to send out calls (the same one on both servers). Here is the output of a call on my office server: -- Attempting call on Local/0445540881644 at CC2 for

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

If i use AMD() or the code below, now the problem is the fax machine/modem detection and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting? ---- dave cantera

Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip

I downloaded these 4 files: app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz for use with asterisk 1.4 (these are the 1.4 files) I installed spandsp, copied app_rxfax and app_txfax into /asterisk-1.4beta3/apps my question is what do I do with asterisk.patch? I tried to put it in the apps directory also and run "patch < asterisk.patch" and it gave me an error.

Hello, In Asterisk 1.4 beta 3, the UPGRADE.txt file says: Variables: * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, and ${LANGUAGE} have all been deprecated in favor of their related dialplan functions. You are encouraged to move towards the associated dialplan function, as these

Does anyone know what the time values in amd.conf are? Are they seconds, fractions of seconds, heartbeats, what? ;'initialSilence' is the maximum silence duration before the greeting initial_silence = 25 ; Maximum silence duration before the greeting. It doesn't say in amd.conf or at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD --

Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was

Hello, i use Red Hat Enterprise Linux ES release 4 (Nahant) with Samba Version 3.0.10-1.4E. Samba are running as PDC. If i save a 160 KB Excel file (Excel 2000 and XP) on a Windows 2000 Client, then samba hangs. With a XP Client running all normally. i make several tunning options (socket options, write raw,...) but no change. any idea? thx chris

hi all, when i start * all zap channels get ring signal so i get a huge number of incoming dummy calls when starting *. i'm using TE105P with 4 TA750 full fxo with latest CVS HEAD: zaptel.conf: span=1,0,0,esf,b8zs fxsks=1-24 span=2,0,0,esf,b8zs fxsks=25-48 span=3,0,0,esf,b8zs fxsks=49-72 span=4,0,0,esf,b8zs fxsks=73-96 loadzone=us defaultzone=us zapata.conf: [channels] context=incoming

Probably this is an old question, but i''m not able to find nothing about... So, i''ve just started to play with tc to limit the transfer speed to my hdsl connection. I''m using the tbf and the command # tc qdisc add dev eth0 root tbf rate 10kbit latency 50ms burst 1000 Then i''ve tried to transfer a big (20 Mbyte) file onto my lan, using ftp and the client

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

HI there. Probably my message is a bit off topic but I do not know where else I could be asking. . I'm running a linux terminal server with one client provided with a 3C509B ISA card. I boot the later with a floppy bootable rom image taken from http://www.rom-o-matic.net/ Everything is working just fine. I decided to try to set up a PXE network boot (I'll needed it for installing a