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
I think that there is a much easier way to detect if the far end is a fax.
In your dialplan, include a section that uses the 'fax' extension. If the
call jumps to that extension, then the far end is a fax machine.
Regards;
John V.
-Original Message-
From:
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
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
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
Sometimes you have to play some audio before calling AMD or other audio
functions for whatever reason... Like play 100ms of silence in a .wav file
immediately after answer. This causes RTP to be sent out to the carrier.
John
From: asterisk-users-boun...@lists.digium.com
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
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
Hi,
I am using my own number and not hanging up and audio is also coming
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 9:14 PM, Ruddy Gbaguidi plugwo...@micnes.comwrote:
Maybe the customer hangs up during the AMD analysis or
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
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.
Maybe the customer hangs up during the AMD analysis 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:
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
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
Al,
Are you doing voice broadcasting
that is, delivering a pre-recorded message, possibly giving a live caller other
options? Just curious. Ive been working on a
voice-broadcasting application myself and Ive had mixed success with
app_amd.c. It does work very well in some cases, but
Michael -Correct, I am attempting to do voice broadcasting. I did try background detect also but could never get that to work either. The only method I got to work was the one which came with the Teleyapper scripts (I based mine off this). In the Teleyapper instance it simply repeated the
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