similar to: AMD Not Working

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

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'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

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

*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

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",

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

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

On 11/01/2019 09:19, Administrator TOOTAI wrote: > Le 11/01/2019 à 10:12, Neil Youngman a écrit : >> A while back, I posted about detecting when a call was picked up by a >> fax machine. It was suggested that having a "fax" extension and >> "faxdetect=yes" would cause it to jump to the "fax" extension. This >> was not something I could

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have

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

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

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

From: Thierry Reding <treding at nvidia.com> Irrespective of whether or not the device has any usable outputs, the modesetting helpers will try to register all the resources such as CRTCs and planes. Unfortunately, the helpers rely on drm_mode_config_init() to properly set up internal data structures. Since the Tegra GPU does not have a display engine, the Nouveau driver doesn't set

On Thu, Jan 04, 2024 at 05:50:22PM +0100, Robert Stepanek wrote: > Since I am undecided yet if and how to fix this in Xapian I haven't > come up with a pull request. Because trac currently is offline, I > could not file a bug. I hope it's OK to post my analysis here first, > I'll be happy to follow up reporting that bug proper later (should we > conclude that it actually

Hi Group, I have following case scenario. Through call file, Asterisk makes a call to SIP extension. When Extension answers the call, Asterisk reads customer numbers (set in callfile) and calls them one by one untill one of the customers answeres the call. Here customer and SIP extension gets patched and talk to each other. Now if outgoing call is answered by Answering machine,I don't want

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 --

On Tue, Nov 06, 2018 at 05:24:15PM +0100, Thierry Reding wrote: > From: Thierry Reding <treding at nvidia.com> > > Irrespective of whether or not the device has any usable outputs, the > modesetting helpers will try to register all the resources such as CRTCs > and planes. Unfortunately, the helpers rely on drm_mode_config_init() to > properly set up internal data

I have one asterisk server where most of the calls that go through AMD get stuck in it, even if the analysis time of 3 seconds has already ended. It doesn't move to the next priority (which is checking the AMD_STATUS). Executing 'show channels' shows that the calls are stuck in the AMD app. I have another identical server where it doesn't happen. Both server have version 1.2.24

On Sun, Jan 7, 2024, at 7:45 PM, Olly Betts wrote: > I've restarted trac. I now created a pull request: https://github.com/xapian/xapian/pull/329 Should I create a trac issue, too? > Assuming the latter is valid, just removing this block (or removing the > parts of it which are Lu or Ll) should fix the problem as then > tokenisation will switch mode - I tried this and it fixes