similar to: AMD reporting NOTSURE most of the time

Displaying 20 results from an estimated 300 matches similar to: "AMD reporting NOTSURE most of the time"

2014 Mar 28
1
AMD with analog lines - DIALSTATUS empty
Hello, I would like to use AMD on outgoing calls using analog line. I tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other end is analog number behind another cisco/asterisk, also tested calling a mobile number with the same result. What I did: dial is done like exten => s,n,Dial(SIP/<IP gw>/<dialed number>,,M(myMacro)), which tell Asterisk to
2007 Oct 12
1
question about PSTN pickup
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
2006 Nov 21
2
Answer Machine Detection
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
2009 May 07
1
Macro arguments on app_queue
hi list, i have a question about the args of queue: when we use Queue() app, there are some arguments than can use. help from CLI: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) well.. i'm trying to identify who has taken the call on a queue, and, when agent conected, launch a macro with some args based on who takes the call i do: exten =>
2012 Feb 02
1
amd detect answering machine
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
2019 Jan 11
2
Detecting a fax
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 get to work. I have now created a very simple example. In sip.conf I have "faxdetect = yes". My example extension is: [test]
2007 Jun 12
1
Answering machine detection after Dial()
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
2009 Apr 23
9
AMD Not Working
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
2019 Jan 11
4
Detecting a fax
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
2017 Nov 04
2
ntfs user mappings?
. DOMAIN_ADMIN_PASSWD.sh echo ${PASSWD} | kinit ${ADMIN}@${DOMAIN} echo -n > /etc/ntfs-3g.usermap for DOMAIN_USER in $(wbinfo -u);do RPCLOOKUPID=$(rpcclient -P -c "lookupnames ${DOMAIN_USER}" ${DOMAIN}) if [ "${RPCLOOKUPID:0:7}" != "ERROR: " ] && [ "${RPCLOOKUPID:0:7}" != "Failed " ];then SID=$(echo ${RPCLOOKUPID}|awk '{print
2017 Nov 03
2
ntfs user mappings?
That looks easier I was working on ldap to convert but I'll try ldb-tools I was off on a bash mission here is what I had so far it isn't correct so I'll keep working on it #!/bin/bash if [ "$(echo $1|wc -c)" = "41" ];then hex=$(echo $1|base64 -d| od -x -w28 --endian=big|head -n1|sed 's/^0000000 //'|sed 's/ //g') echo ${hex} hex_chunk=$(echo
2017 Nov 05
3
ntfs user mappings?
On Sat, 4 Nov 2017 18:42:36 -0600 Jeff Sadowski <jeff.sadowski at gmail.com> wrote: > I decided to continue trying the ldap route as well > > littlehex2int() > { > hex=$1 > hex_chunk=$(echo ${hex}|cut -c$2-$3) > little=$(echo ${hex_chunk}|awk '{print > substr($0,7,2)substr($0,5,2)substr($0,3,2)substr($0,1,2)}') > echo "ibase=16; ${little}" |
2009 Jan 07
2
How to use AMD "Answering Machine Detect" ?
Hi everybody, Happy New Year ! I'm trying to detect if a call was answered by a machine (linke voicemail systems) or a human. I would like to use AMD (Answering Machine Detect) command, but with my configuration it was not possible get there. Follow my dialplan: exten => _[789].,1,NoCDR exten => _[789].,n,Dial(SIP/${EXTEN}@111,60) exten => _[789].,n,AMD
2007 Jan 24
3
setting up AMD
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
2018 Jul 13
2
Withholding Answer Supervision
Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at
2010 Aug 07
2
AMD setup in Astersik
In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf ============================== [ext-queues] include => ext-queues-custom exten => 5000,20,Macro(user-callerid,) ; changed the priority to 20 ............... ============================== In extension_custom.conf
2007 Jun 25
0
Faktortel in Sydney outage
Warning to anyone using or contemplating using Faktortel services (www.faktortel.com.au <http://www.faktortel.com.au/> an Australian ITSP) Their Sydney numbers have been down now for 9 days. No one at technical support can confirm the moving restore time. Oh also in case you ask why don't I drop them as a provider. I got told (and confirmed) that none of the faktortel numbers
2005 May 28
1
3 goes and your out
Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>
2005 Feb 02
2
different IAX ports for different contexts
I have a problem with my asterisk@home installation (configured with AMP) My question is this, can you have different ports for different contexts within IAX? [Faktortel] port = 5036 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=all ; Allow all codecs register => XXXXX:XXXXX@iax.faktotel.com/EXTEN
2005 Feb 11
2
transferring a IAX call into a conference
When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number