Is there any reason why there isn't a setting for min_number_of_words
+ after_greeting_silence. We have an issue where we get one ring followed
by silence and Asterisk thinks it's a human.
from the logs:
[2018-06-13 11:29:31] VERBOSE[30789][C-00034a08] app_amd.c: -- AMD:
initialSilence [3500]
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] on behalf of Tech Support
[aster...@voipbusiness.us]
Sent: 09 February 2018 14:53
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] AMD and fax detection
I think
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neil Youngman
Sent: Friday, February 09, 2018 09:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AMD and fax detection
Is it feasible to enhance AMD to detect and report if the far end sends fax
tones?
I am
Is it feasible to enhance AMD to detect and report if the far end sends fax
tones?
I am guessing that, as it is using DSP to detect sounds and periods of silence,
the same DSP can also report if a CED tone is sent.
If it is feasible, is there a document describing the DSP interface that
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
Le 28/03/2014 15:40, Richard Mudgett a écrit :
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I
tested with SPA3102 and cisco2811 as gw and asterisk
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI ad...@tootai.netwrote:
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
Sent: Friday, February 03, 2012 8:36 AM
Subject: Re: [asterisk-users] amd detect answering machine
Hi,
do noop(${AMDCAUSE}) after exten = 1,1,AMD() , run some test calls and find
out why the call was detected as Answering Machine and adjust amd.conf
accordingly. if I recall correctly
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
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
You would have to make the tolerance of variance fairly high. There are many
reasons why pickup time by a mechanical device such as an answering machine or
a fax machine may vary quite significantly.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta,
Thank you, Alex
yes, I expect the pickup time to vary within 1 second (it's just a guess).
If I have to tolerate higher bias, so I would start doubting about
the efficiency of this method.
On Mon, May 16, 2011 at 4:00 PM, Alex Balashov abalas...@evaristesys.comwrote:
You would have to make the
On 20/08/10 1:52 AM, Tino wrote:
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Record?
--
Cheers,
Matt Riddell
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
Yes, we need to record the message
On Wed, Aug 25, 2010 at 12:35 PM, Matt Riddell li...@venturevoip.comwrote:
On 20/08/10 1:52 AM, Tino wrote:
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Record?
--
Cheers,
Matt
On 25/08/10 7:14 PM, Tino wrote:
Yes, we need to record the message
:D So use the Record() application :D
--
Cheers,
Matt Riddell
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/exchange.php (Full ITSP
I took a quickdirty solution in your case, when I wanted to pick up samples
for analyzing AMD. That was full recording of all outgoing calls
(application Monitor() ), and then I've selected only the phone numbers
which were detected as Answering Machines.
On Wed, Aug 25, 2010 at 10:14 AM, Tino
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
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
Hi Tino,
I think you can do it by using dummy queue number. for example create 500
queue in freepbx. and replace your goto command in ext-queues-custom with
exten = 5000,n,Goto(ext-queues,500,1)
Regards
On Sat, Aug 7, 2010 at 7:06 PM, Tino t...@sparksupport.com wrote:
In my Asterisk server
Hi,
You can use /etc/asterisk/extensions_override_freepbx.conf file if you
dont want your dialplan to get overridden.
Regards,
Rishi
On Saturday 07 August 2010 07:36 PM, Tino wrote:
In my Asterisk server following things have been done to detect
answering machines before the answered
You can include the label of the context in the custom area instead of
including a different context
i.e. [ext-queues](+)
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Not sure if it affects the order of processing or if that matters
Cheers Duncan
On
Hi
I am using the AMD application in a power dialing.
All works well when I use an internal extension but when I try to use an
external number, the AMD every times returns non human status. Also the
AMDCAUSE returns Total-Time-5500. I am using the default parameters in
AMD.CONF.
Anybody has
-users-boun...@lists.digium.com] On Behalf Of Tetra
Informatica
Sent: Monday, June 21, 2010 3:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AMD
Hi
I am using the AMD application in a power dialing.
All works well when I use an internal extension but when I try to use
On Thu, Apr 15, 2010 at 4:06 PM, Baji Panchumarti
baji.panchuma...@gmail.com wrote:
Steve, Chris :
I too had this problem and the solution was not tweaking
the AMD parameters, but playing a short audio file (even
a really really short one) before executing the AMD function.
The key
Steve, Chris :
I too had this problem and the solution was not tweaking
the AMD parameters, but playing a short audio file (even
a really really short one) before executing the AMD function.
The key is executing the Background step before AMD()
Please see sample dialplan below :
On Tue, Mar 23, 2010 at 9:06 PM, Steve Moran s...@matara.net wrote:
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
On 24/03/10 3:06 PM, Steve Moran wrote:
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
On 24/03/10 3:06 PM, Steve Moran wrote:
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
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
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1, agi://
127.0.0.1:4577/call_log) in new stack
It looks like your channel has been hungup during the AMD application,
not that the AMD application is hanging up the call. The source is your
friend (http://www.asterisk.org/doxygen/asterisk1.4/app__amd_8c.html):
00205 /* If we fail to read in a frame, that means they hung up */
00206
I changed my VOIP, and now things are ok.
But didnt understand, how can VOIP can affect it ?
On Wed, Feb 24, 2010 at 11:53 PM, David @ULC ucoms2...@gmail.com wrote:
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
On 1/05/2009 10:10 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
I have tried to play the message 3 times, it played upto 30 seconds.
We have installed amd based on the information given in below link
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD
We are using
: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: May-03-09 6:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMD Not Working
On 1/05/2009 10:10 p.m., Sam Hawkin wrote:
Hi
Hi,
Thanks for your reply.
I have tried to play the message 3 times, it played upto 30 seconds.
We have installed amd based on the information given in below link
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD
We are using the asterisk 1.2.4 in production server, we cannot
Hi,
Thanks for your reply.
We donot kept any absolute time out's.
And we have remove the AMD and kept only the play back,
it works fine.
Any help is highly appreciated.
Thanks.
On Wed, Apr 29, 2009 at 6:35 AM, Matt Riddell li...@venturevoip.com wrote:
On 28/04/2009 4:56 p.m., Sam Hawkin
On 30/04/2009 2:25 a.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
We donot kept any absolute time out's.
And we have remove the AMD and kept only the play back,
it works fine.
Any help is highly appreciated.
Ok, so when you remove AMD and keep playback, how long is the message.
Hi,
Thanks for your reply.
We I remove the AMD it plays the message in the 12 seconds.
It takes 16 seconds before AMD disconnects.
We are using Asterisk 1.2.4
Any help is highly appreciated.
Thanks.
On Thu, Apr 30, 2009 at 3:00 AM, Matt Riddell li...@venturevoip.com wrote:
On 30/04/2009
On 30/04/2009 4:26 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
We I remove the AMD it plays the message in the 12 seconds.
It takes 16 seconds before AMD disconnects.
We are using Asterisk 1.2.4
Any help is highly appreciated.
Few things:
1. Play the message twice without AMD
On 28/04/2009 4:56 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
I have tried as you suggested.
In h extension it is giving Status as AMD_HANGUP.
That normally means that the remote end disconnected the call - if I
were you I'd do a SIP debug to find out why the call is being
On 27/04/2009 4:22 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
I have tried as you suggested, I does not even come upto NoOp()
It hangups after AMD.
I have decreased the silence threshold from 256 to 100 and 50.
Try the NoOp in the h extension:
exten = h,1,NoOp(Status: ${AMDSTATUS}
Hi,
Thanks for your reply.
I have tried as you suggested.
In h extension it is giving Status as AMD_HANGUP.
Below is the log
-- Executing Answer(SIP/sip-874d, ) in new stack
-- Executing AMD(SIP/sip-874d, ) in new stack
-- AMD: SIP/sip-874d (null) (null) (Fmt: 4)
Apr 28
On 25/04/2009 4:29 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply
I have tried the HUMAN as you suggested , but still my problem does not
get solved.
I am answering the call and then running the amd.
Below is the log.
Few things.
1. Put an answer before the AMD line.
2. Put a
Hi,
Thanks for your reply.
I have tried as you suggested, I does not even come upto NoOp()
It hangups after AMD.
I have decreased the silence threshold from 256 to 100 and 50.
below is the log.
-- Executing Answer(SIP/sip-38ea, ) in new stack
-- Executing AMD(SIP/sip-38ea, ) in new stack
or you don’t have
any audio coming to asterisk through your sip channel.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sam Hawkin
*Sent:* April-23-09 11:00 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users
Hi,
Thanks for your reply
I am using my own number and not hanging up. and sip debug is also not
showing much
information regarding the failure.
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Fri, Apr 24, 2009 at 4:58 AM, Steve Totaro
Hi,
Thanks for your reply
We are using the Asterisk 1.2.4.
and below the dialplan path. we are orginating the call to
my number and connection it to context cdtest and extension 1.
[cdtest]
exten = 1,1,NoOp( cb amd issue testing )
exten =
Hello,
Well, depending on the version of app_amd that you used when you added
it to Asterisk 1.2, you might need to use HUMAN and MACHINE as the
possible AMDSTATUS instead of AMD_PERSON and AMD_MACHINE. The
AMDSTATUS was changed at some point in the app_amd code, not sure why
they changed it, but
Hi,
Thanks for your reply
I have tried the HUMAN as you suggested , but still my problem does not get
solved.
I am answering the call and then running the amd.
Below is the log.
-- AMD: SIP/sip-58ab (null) (null) (Fmt: 4)
Apr 25 00:26:07 NOTICE[27310]: app_amd.c:134 isAnsweringMachine: AMD
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
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:
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.
@lists.digium.com
Subject: [asterisk-users] 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
On 24/04/2009 3:00 a.m., Sam Hawkin wrote:
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.
I'd say that the remote end of the call is hanging up - do a SIP debug
so you can see what
On Thu, Apr 23, 2009 at 6:12 PM, Matt Riddell li...@venturevoip.com wrote:
On 24/04/2009 3:00 a.m., Sam Hawkin wrote:
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.
I'd say
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
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
Add an answer() and a playback of 1 second of silence or something else
to make sure the RTP is nailed up. AMD can/will hang if it has no media
to analyze.
Carlos Chavez wrote:
We have an Asterisk server with a small outgoing call center. We use
AMD and it usually works very well on
Hi -I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is
Lougher
Sent: Wednesday, June 21, 2006
9:24 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AMD
Machine Detect
Hi -
I have been developing an auto-dialling application similar to
Voiceshot, Call-em-all etc. The one thing I am now struggling to get working
: [Asterisk-Users] AMD Machine Detect Hi -I have been developing an
auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c
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