similar to: How to tune Asterisk AMD - Answering Machine Detection "hacks"

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

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, 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

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

*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

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

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 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

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 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

Hello Everyone!! Wanted to ask for your help in what is the best way to do a callback service with asterisk. I want to be able to read a file containing two number to call and then call the two numbers and bridge them. Thanks in advance, Costa -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We created a dial plan which performs and outbound dial out and deliveres a message to a receipient What call method/option in extensions or anywhere allow us to determine the length of the message. IE, what if a 3 minute message is attempted to be delivered and 45seconds into the call drops or is terminated by the end user. I need to know that the entire message was/was not delivered. What

Hello, I'm trying to install dovecot antispam plugin and I've some problems with dspam user. When I move a mail out of 'SPAM' folder I have this error in my /var/log/messages : Oct 28 17:01:41 tony2 imap: antispam: /usr/bin/dspam --source=error --class=innocent --signature=47de4679174462472577556 ... Oct 28 17:01:41 tony2 dspam[24313]: Unable to determine the destination

Dear I have connected samba 3.6.8 to my Active Directory in the lsass.exe run to 100% When stopping winbind the lsass.exe CPU is down to 0% When set winbindd to debug mode, it seems it try to scan the root user every time. I would to know how to ban nsswitch to query winbindd for system internal users such has root, apache..... Here it is my nsswitch.conf : # # Example configuration of GNU

I have successfully set up the master user on the destination server (2.0.11) and tests have worked. now I'm working on the proxy Before I had the proxy just forward everything to the backend and had the destination server do the authentication. My authentication is done via LDAP but not really sure how to append the master user and password to the users credentials after authentication is