similar to: Answering Machine Detect (AMD) time values

*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

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

Hi all, i'm trying to make AMD, Answer Machine Detection, to work on my outbound context but i can't get it to work, just on inbound context like whe i use the application Answer before AMD, but i need to make AMD to do the detection on an outbound predictive dialer integration. Follow are the inbound and outbound examples. My current environment is Asterisk 1.4beta3 and a Digum

I like to use zgsmplay to preview Asterisk's gsm files. Works like a charm in 1.2. But something changed in 1.4- they play at the wrong speed, very slowly. They sound fine in the normal course of using the Asterisk server. The file sizes seem wacko. The first example is from 1.2, the second from 1.4: $ ls -l im-sorry.gsm -rw-r--r-- 1 1000 1000 1419 Dec 5 2005 im-sorry.gsm $ ls -l

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

Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards,

hey all, I'm getting this warning on the console when I leave a voicemail on my test server: [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing the mailbox with an option is deprecated ('u250@local-vm-users'). [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6553 vm_exec: Please move all leading options to the second argument. This is what voicemail.conf

One more question- is there a reference somewheres for all the built-in Asterisk macros? thanks! -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ?Carla Schroder ?check out my "Linux Cookbook", the ultimate Linux user's ?and sysadmin's guide! http://www.oreilly.com/catalog/linuxckbk/ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

What does option '11. Module Embedding' do in Asterisk 1.4? The default is none of them are selected: [ ] 1. apps [ ] 2. cdr [ ] 3. channels [ ] 4. codecs [ ] 5. formats [ ] 6. funcs [ ] 7. pbx [ ] 8. res -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ?Carla Schroder ?check out my "Linux Cookbook", the ultimate Linux user's

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 list! I have to do a new bare metal installation of a box running Asterisk with bristuff or vzaphfc. The box will be used as a really lightly loaded file server and pbx. Any advise on which architecture I should use? The cpu is a 64 bit capable AMD (the box is running x86_64 now) but is still suffering from echo on the BRI lines. Should I go with the normal x86 or the 64 bit x86_64

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

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a

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

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, 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

How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f - Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061229/638e4426/attachment.htm

Hey all, I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial "ease of use"

We have an Asterisk server with a small outgoing call center. We use AMD and it usually works very well on Zap channels (E1 PRI). We added a couple of SIP trunks to reduce long distance costs but now AMD gets stuck when the call goes out through the SIP channels. Here is an example call using a SIP line: -- Executing [016566275538 at CC2:1] Set("Local/016566275538 at CC2-dad7,2",

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