[asterisk-users] AMD min amount of words
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] greeting [2000] afterGreetingSilence [400] totalAnalysisTime [5000] minimumWordLength [80] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] maximumWordLength [5000] [2018-06-13 11:29:31] VERBOSE[30789][C-00034a08] app_amd.c: -- AMD: Channel [SIP/d1-d57a]. Changed state to STATE_IN_SILENCE [2018-06-13 11:29:32] VERBOSE[30789][C-00034a08] app_amd.c: -- AMD: Channel [SIP/d1-d57a]. Word detected. iWordsCount:1 [2018-06-13 11:29:32] VERBOSE[30789][C-00034a08] app_amd.c: -- AMD: Channel [SIP/d1-d57a]. Detected Talk, previous silence duration: 1040 [2018-06-13 11:29:32] VERBOSE[30789][C-00034a08] app_amd.c: -- AMD: Channel [SIP/d1-d57a]. Changed state to STATE_IN_SILENCE [2018-06-13 11:29:33] VERBOSE[30789][C-00034a08] app_amd.c: -- AMD: Channel [SIP/d1-d57a]. HUMAN: silenceDuration:400 afterGreetingSilence:400 TIA Dovid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD and fax detection
A quick test to a fax number doesn't seem to jump to the 'fax' extension. I checked faxdetect=both on chan_dahdi.conf. I seem to remember trying this some years back and finding that fax detect only worked for CNG tones, not CED tones? Neil Youngman 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 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: asterisk-users-boun...@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 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 someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD and fax detection
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: asterisk-users-boun...@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 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 someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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 someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 execute myMacro when the
call is answered by calling party.
[myMacro]
exten = s,1,NoOP(Executed when call is answered)
same = n,AMD()
same = n,NoOp(Dial status=${DIALSTATUS})
same = n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same = n,MacroExit()
Problem is that [myMacro] is executed as soon as the call is going out
from the gw (cisco or linksys) and before called party answered.
DIALSTATUS is empty (should be ANSWER), AMDSTATUS=NOTSURE and
AMDCAUSE=TOOLONG-5000 which seems OK as DIALSTATUS isn't reliable.
The same dialplan using a SIP trunk is working as expected.
So question is, why, when using analog line, I dont get the right behavior.
Thanks for any hint
--
Daniel
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Re: [asterisk-users] AMD with analog lines - DIALSTATUS empty
Le 28/03/2014 15:40, Richard Mudgett a écrit :
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I
tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as
well as 11.8.1 Other end is analog number behind another
cisco/asterisk, also tested calling a mobile number with the same
result.
What I did: dial is done like exten = s,n,Dial(SIP/IP
gw/dialed number,,M(myMacro)), which tell Asterisk to execute
myMacro when the call is answered by calling party.
[myMacro]
exten = s,1,NoOP(Executed when call is answered)
same = n,AMD()
same = n,NoOp(Dial status=${DIALSTATUS})
same = n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same = n,MacroExit()
Problem is that [myMacro] is executed as soon as the call is going
out from the gw (cisco or linksys) and before called party
answered. DIALSTATUS is empty (should be ANSWER),
AMDSTATUS=NOTSURE and AMDCAUSE=TOOLONG-5000 which seems OK as
DIALSTATUS isn't reliable.
The same dialplan using a SIP trunk is working as expected.
So question is, why, when using analog line, I dont get the right
behavior.
Thanks for any hint
Analog lines don't have a reliable way to know when the far end
actually answers. Polarity
reversals could signal when the far end actually answers, but it isn't
normally available or
standardized. Thus, the line is considered answered when dialing is
complete.
OK, so it's a no way.
Thanks for your answer
--
Daniel
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Re: [asterisk-users] AMD with analog lines - DIALSTATUS empty
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 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 execute myMacro when the call
is answered by calling party.
[myMacro]
exten = s,1,NoOP(Executed when call is answered)
same = n,AMD()
same = n,NoOp(Dial status=${DIALSTATUS})
same = n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same = n,MacroExit()
Problem is that [myMacro] is executed as soon as the call is going out
from the gw (cisco or linksys) and before called party answered. DIALSTATUS
is empty (should be ANSWER), AMDSTATUS=NOTSURE and AMDCAUSE=TOOLONG-5000
which seems OK as DIALSTATUS isn't reliable.
The same dialplan using a SIP trunk is working as expected.
So question is, why, when using analog line, I dont get the right behavior.
Thanks for any hint
Analog lines don't have a reliable way to know when the far end actually
answers. Polarity
reversals could signal when the far end actually answers, but it isn't
normally available or
standardized. Thus, the line is considered answered when dialing is
complete.
Richard
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Re: [asterisk-users] amd detect answering machine
Hi,
I'd try to change somethings about my asterisk configuration.
I think, it's near OK !
AMD (Asterisk Machine detect) is right but it's detect if the caller is machine
or not.
I'd like to detect the called. for example to detect if my called answers me or
if it's his answering machine phone.
Here is my extensions.conf code :
[ServeurPro]
exten = s,1,Ringing()
exten = s,2,Wait(2)
exten = s,3,Answer()
exten = s,4,Set(NbInvalide=0)
exten = s,5,Set(NbEssai=0)
exten = s,6,background(${ChmAudio}/ServeurProBienvenu)
exten = s,7,WaitExten(2)
exten = 1,1,AMD()
exten = 1,2,GotoIf($[${AMDSTATUS}==MACHINE]?1,4)
exten = 1,3,Dial(SIP/0682416304@ippi_outgoing2,40,m(Attente))
exten = 1,4,Voicemail(801@FloriePro,us)
exten = i,1,Set(NbInvalide=$[${NbInvalide}+1]})
exten = i,2,Gotoif($[${NbInvalide} 3]?:6)
exten = i,3,Playback(${ChmAudio}/ErreurSaisie)
exten = i,4,Playback(${ChmAudio}/RetourMenu)
exten = i,5,Goto(s,6)
exten = i,6,Playback(${ChmAudio}/ErreurSaisie)
exten = i,7,Playback(${ChmAudio}/Aurevoir)
exten = i,8,Hangup()
exten = t,1,Set(NbEssai=$[${NbEssai}+1])
exten = t,2,Gotoif($[${NbEssai} 3]?:5)
exten = t,3,Playback(${ChmAudio}/DemandeIncomprise)
exten = t,4,Goto(s,6)
exten = t,5,PlayBack(${ChmAudio}/Aurevoir)
exten = t,6,Hangup()
In asterisk cli :
If I'm calling mobile phone, I'm talking when it's ringing and AMD detects me
like MACHINE OR Not sure. If I'm saying just one word, AMD detects me like
human.
I'd like it doesn't detect me but my called.
Please help me !
Sorry for my bad english...
AMICALEMENT
Manu
SITES WEBS
Mon site web Officiel (Manu-dpk.net)
Ecoutez Radio DPK
CONTACT
- E-mail : manuli...@manu-dpk.net
- Messenger (WLM) : m...@manu-dpk.net
- Skype : manu-dpk
PS : Pour le respect de l'environnnement, n'imprimez ce mail qu'en cas de
nécessité.
- Original Message -
From: Aurimas Skirgaila
To: Etann ; Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, February 03, 2012 8:36 AM
Subject: Re: [asterisk-users] amd detect answering machine
Hi,
do noop(${AMDCAUSE}) after exten = 1,1,AMD() , run some test calls and find
out why the call was detected as Answering Machine and adjust amd.conf
accordingly. if I recall correctly, you can also see the AMD flow in Asterisk
in verbose mode.
I'd suspect low silence_threshold . I usually set it 384, but it's very
dependent on carrier.
On Thu, Feb 2, 2012 at 5:51 PM, Etann manuli...@manu-dpk.net wrote:
Hi,
I have IVR and when I press 1, asterisk calls my mobile phone.
If my mobile phone is offline, asterisk transfers to asterisk voicemail.
I'd like asterisk detects my mobile voicemail and if my mobile voicemail
answers, asterisk transfers to asterisk voicemail.
For that, I used AMD.
So I have problems ! Asterisk detects answering machine everytime!
How do I do please ?
extensions.conf
[ServeurPro]
exten = s,1,Ringing()
exten = s,2,Wait(2)
exten = s,3,Answer()
exten = s,4,Set(NbInvalide=0)
exten = s,5,Set(NbEssai=0)
exten = s,6,background(${ChmAudio}/ServeurProBienvenu)
exten = s,7,WaitExten(2)
exten = 1,1,AMD()
exten = 1,2,GotoIf($[${AMDSTATUS}=MACHINE]?1,4)
exten = 1,3,Dial(SIP/@ippi_outgoing2,40,r)
exten = 1,4,Voicemail(801@FloriePro,us)
exten = i,1,Set(NbInvalide=$[${NbInvalide}+1]})
exten = i,2,Gotoif($[${NbInvalide} 3]?:6)
exten = i,3,Playback(${ChmAudio}/ErreurSaisie)
exten = i,4,Playback(${ChmAudio}/RetourMenu)
exten = i,5,Goto(s,6)
exten = i,6,Playback(${ChmAudio}/ErreurSaisie)
exten = i,7,Playback(${ChmAudio}/Aurevoir)
exten = i,8,Hangup()
exten = t,1,Set(NbEssai=$[${NbEssai}+1])
exten = t,2,Gotoif($[${NbEssai} 3]?:5)
exten = t,3,Playback(${ChmAudio}/DemandeIncomprise)
exten = t,4,Goto(s,6)
exten = t,5,PlayBack(${ChmAudio}/Aurevoir)
exten = t,6,Hangup()
exten = h,1,noOp(Statut AMD : ${AMDSTATUS})
amd.conf
[general]
initial_silence = 2500 ; Maximum silence duration before the greeting.
; If exceeded then MACHINE.
greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
after_greeting_silence = 500 ; Silence after detecting a greeting.
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to
decide
; on a HUMAN or MACHINE
min_word_length = 120 ; Minimum duration of Voice to considered as a word
between_words_silence = 50 ; Minimum duration of silence after a word to
consider
; the audio what follows as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded then MACHINE
silence_threshold = 256
Thank you for your reply and for help
[asterisk-users] amd detect answering machine
Hi,
I have IVR and when I press 1, asterisk calls my mobile phone.
If my mobile phone is offline, asterisk transfers to asterisk voicemail.
I'd like asterisk detects my mobile voicemail and if my mobile voicemail
answers, asterisk transfers to asterisk voicemail.
For that, I used AMD.
So I have problems ! Asterisk detects answering machine everytime!
How do I do please ?
extensions.conf
[ServeurPro]
exten = s,1,Ringing()
exten = s,2,Wait(2)
exten = s,3,Answer()
exten = s,4,Set(NbInvalide=0)
exten = s,5,Set(NbEssai=0)
exten = s,6,background(${ChmAudio}/ServeurProBienvenu)
exten = s,7,WaitExten(2)
exten = 1,1,AMD()
exten = 1,2,GotoIf($[${AMDSTATUS}=MACHINE]?1,4)
exten = 1,3,Dial(SIP/@ippi_outgoing2,40,r)
exten = 1,4,Voicemail(801@FloriePro,us)
exten = i,1,Set(NbInvalide=$[${NbInvalide}+1]})
exten = i,2,Gotoif($[${NbInvalide} 3]?:6)
exten = i,3,Playback(${ChmAudio}/ErreurSaisie)
exten = i,4,Playback(${ChmAudio}/RetourMenu)
exten = i,5,Goto(s,6)
exten = i,6,Playback(${ChmAudio}/ErreurSaisie)
exten = i,7,Playback(${ChmAudio}/Aurevoir)
exten = i,8,Hangup()
exten = t,1,Set(NbEssai=$[${NbEssai}+1])
exten = t,2,Gotoif($[${NbEssai} 3]?:5)
exten = t,3,Playback(${ChmAudio}/DemandeIncomprise)
exten = t,4,Goto(s,6)
exten = t,5,PlayBack(${ChmAudio}/Aurevoir)
exten = t,6,Hangup()
exten = h,1,noOp(Statut AMD : ${AMDSTATUS})
amd.conf
[general]
initial_silence = 2500 ; Maximum silence duration before the greeting.
; If exceeded then MACHINE.
greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
after_greeting_silence = 500 ; Silence after detecting a greeting.
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
; on a HUMAN or MACHINE
min_word_length = 120 ; Minimum duration of Voice to considered as a word
between_words_silence = 50 ; Minimum duration of silence after a word to
consider
; the audio what follows as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded then MACHINE
silence_threshold = 256
Thank you for your reply and for help!
AMICALEMENT
Manu
SITES WEBS
Mon site web Officiel (Manu-dpk.net)
Ecoutez Radio DPK
CONTACT
- E-mail : manuli...@manu-dpk.net
- Messenger (WLM) : m...@manu-dpk.net
- Skype : manu-dpk
PS : Pour le respect de l'environnnement, n'imprimez ce mail qu'en cas de
nécessité. --
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[asterisk-users] AMD tweaking
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 about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD tweaking
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, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On May 16, 2011, at 8:56 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: 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 about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD tweaking
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 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, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On May 16, 2011, at 8:56 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: 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 about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD message
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) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD message
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 Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD message
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 Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD message
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 t...@sparksupport.com wrote: 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 Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMD message
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 introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 added following amd dialplan
===
[ext-queues-custom]
exten = 5000,1,Answer()
exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human)
exten = 5000,n(machine),Verbose(3, We found an answring machine)
exten = 5000,n,Set(AMP=${CALLERID(num)})
exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = 5000,n,System(not showing the actual command)
exten = 5000,n,Goto(ext-queues,5000,20)
exten = 5000,n(human),Verbose(3, We've got a human on the line!)
exten = 5000,n,Goto(ext-queues,5000,20)
===
This setup is working fine but the problem is that when i reload freepbx,
extension_additional.conf will go to its original form
and the changes made will be lost. Is there any way to make the changes in
extension_additional.conf conf permanent . Or is there any alternative
method for this ?
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Re: [asterisk-users] AMD setup in Astersik
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 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 added following amd dialplan
===
[ext-queues-custom]
exten = 5000,1,Answer()
exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human)
exten = 5000,n(machine),Verbose(3, We found an answring machine)
exten = 5000,n,Set(AMP=${CALLERID(num)})
exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = 5000,n,System(not showing the actual command)
exten = 5000,n,Goto(ext-queues,5000,20)
exten = 5000,n(human),Verbose(3, We've got a human on the line!)
exten = 5000,n,Goto(ext-queues,5000,20)
===
This setup is working fine but the problem is that when i reload freepbx,
extension_additional.conf will go to its original form
and the changes made will be lost. Is there any way to make the changes in
extension_additional.conf conf permanent . Or is there any alternative
method for this ?
--
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--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
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Re: [asterisk-users] AMD setup in Astersik
Hi,
You can use /etc/asterisk/extensions_override_freepbx.conf file if you
dont want your dialplan to get overridden.
Regards,
Rishi
On Saturday 07 August 2010 07:36 PM, Tino wrote:
In my Asterisk server following things have been done to detect
answering machines before the answered call connects to the agents in
queue.
In extension_additional.conf
==
[ext-queues]
include = ext-queues-custom
exten = 5000,20,Macro(user-callerid,); changed the priority to 20
...
==
In extension_custom.conf added following amd dialplan
===
[ext-queues-custom]
exten = 5000,1,Answer()
exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human)
exten = 5000,n(machine),Verbose(3, We found an answring machine)
exten = 5000,n,Set(AMP=${CALLERID(num)})
exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = 5000,n,System(not showing the actual command)
exten = 5000,n,Goto(ext-queues,5000,20)
exten = 5000,n(human),Verbose(3, We've got a human on the line!)
exten = 5000,n,Goto(ext-queues,5000,20)
===
This setup is working fine but the problem is that when i reload
freepbx, extension_additional.conf will go to its original form
and the changes made will be lost. Is there any way to make the
changes in extension_additional.conf conf permanent . Or is there any
alternative method for this ?
--
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Re: [asterisk-users] AMD setup in Astersik
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 8/08/2010, at 7:43 AM, Rushikesh wrote:
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 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 added following amd dialplan
===
[ext-queues-custom]
exten = 5000,1,Answer()
exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human)
exten = 5000,n(machine),Verbose(3, We found an answring machine)
exten = 5000,n,Set(AMP=${CALLERID(num)})
exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = 5000,n,System(not showing the actual command)
exten = 5000,n,Goto(ext-queues,5000,20)
exten = 5000,n(human),Verbose(3, We've got a human on the line!)
exten = 5000,n,Goto(ext-queues,5000,20)
===
This setup is working fine but the problem is that when i reload
freepbx, extension_additional.conf will go to its original form
and the changes made will be lost. Is there any way to make the
changes in extension_additional.conf conf permanent . Or is there any
alternative method for this ?
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[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 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 some idea? Thanks Sergio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD
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 [mailto:asterisk-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 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 some idea? Thanks Sergio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
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 is executing the Background step before AMD() You're right, that does seem to make difference. I tried a couple of test calls using your solution and my AMD() function correctly detected HUMAN where it had been getting NOTSURE. Thanks for the help, I'll do some more testing as soon as I can. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
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 :
exten = s,n,Answer()
exten = s,n,Background(BLANK_AUDIO)
exten = s,n,AMD()
;
exten = s,n,GotoIf($[${AMDSTATUS} = HANGUP ]?Hungup:)
exten = s,n,GotoIf($[${AMDSTATUS} = MACHINE]?${app_id}_V,1:)
;
exten = s,n,GotoIf($[${AMDSTATUS} =
HUMAN]?${app_id}_L,1:${app_id}_V,1)
In my case the BLANK_AUDIO sound file is 0.1 secs
of silence.
This script tends to detect more calls as ANS m/c over
live pickups.
If you prefer false positive in the other direction (more calls
detected as Live over ans m/c), then change the order of
tests.
Hope that helps.
-baji.
--
On Sat, Apr 10, 2010, Chris Gentle wrote:
On Tue, Mar 23, 2010, 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 week, and 70% said NOTSURE).
Hi. Did you ever resolve this? I am having the same problem as you when I
use AMD with outgoing calls through my Vitelity line. Sending the calls out
PSTN seems to work as normal. I tried tweaking the threshold setting as
someone else pointed out but it didn't make any difference.
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Re: [asterisk-users] AMD reporting NOTSURE most of the time
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 50,000 outbound calls last week, and 70% said NOTSURE). Hi. Did you ever resolve this? I am having the same problem as you when I use AMD with outgoing calls through my Vitelity line. Sending the calls out PSTN seems to work as normal. I tried tweaking the threshold setting as someone else pointed out but it didn't make any difference. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
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 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 lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 Looks like it's missing the first word - some VoIP providers take a while to pass audio - might be that there is a delay in your dialplan or that the first words of audio are simply not transmitted. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
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 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 lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 Alternatively your threshold might be too high - do a few tests to your own phone and make sure it recognizes the individual words. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMD reporting NOTSURE most of the time
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 lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 My AMD.conf settings are all set to default:- [general] initial_silence = 2500 ; Maximum silence duration before the greeting. ; If exceeded then MACHINE. greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE. after_greeting_silence = 800; Silence after detecting a greeting. ; If exceeded then HUMAN total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide ; on a HUMAN or MACHINE min_word_length = 100 ; Minimum duration of Voice to considered as a word between_words_silence = 50 ; Minimum duration of silence after a word to consider ; the audio what follows as a new word maximum_number_of_words = 5 ; Maximum number of words in the greeting. ; If exceeded then MACHINE silence_threshold = 256 Just wondering if any of you AMD users have any ideas as to what I should check. When I view this on the console I see that it jumps to too long almost immediately:- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] maximumWordLength [5000] -- AMD: Channel [SIP/faktortel-1385]. Changed state to STATE_IN_SILENCE -- AMD: Channel [SIP/faktortel-1385]. Too long... -- AMD: Channel [SIP/faktortel-1385]. Too long... Thanks Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMD: HANGUP
*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
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD(Local/91441425477...@default-b9f2,1,
2000|2000|1000|5000|120|50|4|256) in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
[50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI(Local/91441425477...@default-1e22,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-1e22,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---) in
new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI(Local/91441425477...@default-86e4,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-86e4,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---) in
new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
vici*CLI
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten = 8369,1,Playback(sip-silence)
exten = 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten = 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten = 8369,4,AGI(VD_amd.agi,${EXTEN})
exten = 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,7,Hangup
Amd.conf has :
*Code:*
; initial_silence: Maximum silence duration before the greeting. If exceeded
then MACHINE.
; greeting: Maximum length of a greeting. If exceeded then MACHINE.
; after_greeting_silence: Silence after detecting a greeting. If exceeded
then HUMAN
; total_analysis_time: Maximum time allowed for the algorithm to decide on a
HUMAN or PERSON
; min_word_length: Minimum duration of Voice to considered as a word
; between_words_silence: Minimum duration of silence after a word to
considere the audio what follows as a new word
; maximum_number_of_words: Maximum number of words in the greeting. If
exceeded then MACHINE
[AnsweringMachineDetector]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
--
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Re: [asterisk-users] AMD: HANGUP
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 if (!(f = ast_read
http://www.asterisk.org/doxygen/asterisk1.4/channel_8c.html#7ef6737309dc9e8b6c4a7cb4800638b1(chan)))
{
00207 if (option_verbose
http://www.asterisk.org/doxygen/asterisk1.4/group__main__options.html#ga294d0efa6a89c1a3d162787cac4fff5
2)
00208 ast_verbose
http://www.asterisk.org/doxygen/asterisk1.4/logger_8c.html#81d26348827b996085d4cb6be3e2c348(VERBOSE_PREFIX_3
http://www.asterisk.org/doxygen/asterisk1.4/logger_8h.html#24b0f46e22f4ea3226fa082e955dd4ef
AMD: HANGUP\n);
00209 if (option_debug
http://www.asterisk.org/doxygen/asterisk1.4/group__main__options.html#g40f8fb2e731031d99f732f515cec680f)
00210 ast_log
http://www.asterisk.org/doxygen/asterisk1.4/logger_8c.html#93dd824dff97fe84941d6d71b7a3710e(LOG_DEBUG
http://www.asterisk.org/doxygen/asterisk1.4/logger_8h.html#6ff63e8955665c4a58b1598f2b07c51a,
Got hangup\n);
00211 strcpy(amdStatus, HANGUP);
00212 break;
00213 }
So basically check that the channel is not being hungup during
application execution.
Regards,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
David @ULC escribió:
*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 http://127.0.0.1:4577/call_log) in
new stack
-- AGI Script agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD(Local/91441425477...@default-b9f2,1,
2000|2000|1000|5000|120|50|4|256) in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null)
(Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI(Local/91441425477...@default-1e22,2,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15
completed, returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-1e22,1,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
completed, returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI(Local/91441425477...@default-86e4,2,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15
completed, returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-86e4,1,
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
http://127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---
completed, returning 0
vici*CLI
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten = 8369,1,Playback(sip-silence)
exten = 8369,2,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten = 8369,4,AGI(VD_amd.agi,${EXTEN})
exten = 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,7,Hangup
Amd.conf has :
*Code:*
Re: [asterisk-users] AMD: HANGUP
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,
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
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD(Local/91441425477...@default-b9f2,1,
2000|2000|1000|5000|120|50|4|256) in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null) (Fmt:
64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
[50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI(Local/91441425477...@default-1e22,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-1e22,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI(Local/91441425477...@default-86e4,2, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI(Local/91441425477...@default-86e4,1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---)
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
vici*CLI
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten = 8369,1,Playback(sip-silence)
exten = 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten = 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten = 8369,4,AGI(VD_amd.agi,${EXTEN})
exten = 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten = 8369,7,Hangup
Amd.conf has :
*Code:*
; initial_silence: Maximum silence duration before the greeting. If
exceeded then MACHINE.
; greeting: Maximum length of a greeting. If exceeded then MACHINE.
; after_greeting_silence: Silence after detecting a greeting. If exceeded
then HUMAN
; total_analysis_time: Maximum time allowed for the algorithm to decide on
a HUMAN or PERSON
; min_word_length: Minimum duration of Voice to considered as a word
; between_words_silence: Minimum duration of silence after a word to
considere the audio what follows as a new word
; maximum_number_of_words: Maximum number of words in the greeting. If
exceeded then MACHINE
[AnsweringMachineDetector]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
--
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Re: [asterisk-users] AMD Not Working
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 the asterisk 1.2.4 in production server, we cannot go to new version. There are major changes (even in the 1.2 branch) since then, and Asterisk is now up to 1.6. How did AMD go? Is it a backport? Hope your machine is not accessible via the Internet as 1.2.4 is likely to have quite a few security vulnerabilities in it. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hope your machine is not accessible via the Internet as 1.2.4 is likely to have quite a few security vulnerabilities in it. And they are ? the core itself ? LOL.. Im sure 1.6 has more than 1.2 or 1.4, but then again.. i was just reading a thread here I have connected my Asterisk-box directly to my internetconnection. I have disabled my firewall. Still I am unable to register with my IAX-provider. Can someone I think code 18 here, hence people getting hacked.. Probably with default settings that include (longdistance and international ) ;) -Original Message- From: 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, 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 go to new version. There are major changes (even in the 1.2 branch) since then, and Asterisk is now up to 1.6. How did AMD go? Is it a backport? Hope your machine is not accessible via the Internet as 1.2.4 is likely to have quite a few security vulnerabilities in it. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 go to new version. Any help is highly appreciated. Thanks. On Thu, Apr 30, 2009 at 11:09 AM, Matt Riddell li...@venturevoip.comwrote: 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 (you might be being disconnected after 15 seconds) 2. I thought AMD wasn't present in 1.2. Is it a backport? 3. 1.2.4 is quite an old version, any chance you could upgrade it to a more recent version? There have been many bugs fixed since 1.2.4 was released. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 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 disconnected. You don't have any absolute timeouts or anything? The other thing to test would be to skip AMD for the moment and just play some audio instead and see if it hangs up in that case. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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. Secondly, how long does it take before you are disconnected with AMD. Oh, and which version of Asterisk are you running? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 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. Secondly, how long does it take before you are disconnected with AMD. Oh, and which version of Asterisk are you running? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 (you might be being disconnected after 15 seconds) 2. I thought AMD wasn't present in 1.2. Is it a backport? 3. 1.2.4 is quite an old version, any chance you could upgrade it to a more recent version? There have been many bugs fixed since 1.2.4 was released. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 disconnected. You don't have any absolute timeouts or anything? The other thing to test would be to skip AMD for the moment and just play some audio instead and see if it hangs up in that case. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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} Cause: ${AMDCAUSE})
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
___
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Re: [asterisk-users] AMD Not Working
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 00:53:41 NOTICE[5837]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
-- Executing NoOp(SIP/sip-874d, Status: AMD_HANGUP Cause: ) in new
stack
vm3*CLI
Any help is highly appreciated.
Thanks.
On Mon, Apr 27, 2009 at 5:04 PM, Matt Riddell li...@venturevoip.com wrote:
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} Cause: ${AMDCAUSE})
--
Kind Regards,
Matt Riddell
Director
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Re: [asterisk-users] AMD Not Working
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 NoOp(Status: ${AMDSTATUS} Cause: ${AMDCAUSE}) after
3. Decrease the silence threshold
--
Kind Regards,
Matt Riddell
Director
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Re: [asterisk-users] AMD Not Working
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
-- AMD: SIP/sip-38ea (null) (null) (Fmt: 4)
Apr 27 00:14:25 NOTICE[20035]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [50]
-- AMD: HANGUP
vm3*CLI
Any help is highly appreciated.
Thanks.
On Mon, Apr 27, 2009 at 3:55 AM, Matt Riddell li...@venturevoip.com wrote:
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 NoOp(Status: ${AMDSTATUS} Cause: ${AMDCAUSE}) after
3. Decrease the silence threshold
--
Kind Regards,
Matt Riddell
Director
___
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Re: [asterisk-users] AMD Not Working
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 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] 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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly appreciated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 stot...@totarotechnologies.com wrote: On Thu, Apr 23, 2009 at 6:12 PM, Matt Riddell li...@venturevoip.comwrote: 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 happens - the best way to test things like this is by calling your own number - that way you can guarantee it doesn't hang up :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) You can also run Orecx on the localhost (for very small production or lab systems) or on a different host via mirrored switch port and then listen to all calls (SIP and other VoIP), or RTPTap via Sangoma cards). I have done this many times to catch intermittent problems that are continuously reported by users but cannot be readily reproduced. I just ask that the user log the time of the call and what they experienced, then I can listen to the recording, ascertain all the critical info that users leave off trouble reports, and figure out the commonalities. Obviously, all due notice/permission and/or legal disclosures should be made/given before recording anything. It is great for troubleshooting (and yes, calls do get crossed and all kinds of other strangness in Asterisk, you know, what you write off as user error :-) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 = 1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten = s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten = s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com wrote:
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.
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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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Re: [asterisk-users] AMD Not Working
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 that might be your issue.
Also, since you are calling your own number you might want to do an
Answer on the call before running AMD, not sure if that would cause
the hangups you are seeing or not, but it's something to try.
MATT---
On 4/24/09, Sam Hawkin gvrt...@gmail.com wrote:
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 =
1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten =
s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten =
s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com wrote:
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.
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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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Re: [asterisk-users] AMD Not Working
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 using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
vm3*CLI
Any help is highly appreciated.
Thanks.
On Fri, Apr 24, 2009 at 4:03 PM, Matt Florell astma...@gmail.com wrote:
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 that might be your issue.
Also, since you are calling your own number you might want to do an
Answer on the call before running AMD, not sure if that would cause
the hangups you are seeing or not, but it's something to try.
MATT---
On 4/24/09, Sam Hawkin gvrt...@gmail.com wrote:
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 =
1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten =
s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten =
s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com
wrote:
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.
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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500]
afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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[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 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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly appreciated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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. 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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP What version of Asterisk are you running this on? What is the dialplan path that this is running through? MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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: asterisk-users@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 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 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly appreciated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 happens - the best way to test things like this is by calling your own number - that way you can guarantee it doesn't hang up :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
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 that the remote end of the call is hanging up - do a SIP debug so you can see what happens - the best way to test things like this is by calling your own number - that way you can guarantee it doesn't hang up :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) You can also run Orecx on the localhost (for very small production or lab systems) or on a different host via mirrored switch port and then listen to all calls (SIP and other VoIP), or RTPTap via Sangoma cards). I have done this many times to catch intermittent problems that are continuously reported by users but cannot be readily reproduced. I just ask that the user log the time of the call and what they experienced, then I can listen to the recording, ascertain all the critical info that users leave off trouble reports, and figure out the commonalities. Obviously, all due notice/permission and/or legal disclosures should be made/given before recording anything. It is great for troubleshooting (and yes, calls do get crossed and all kinds of other strangness in Asterisk, you know, what you write off as user error :-) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMD timing issues
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
tested it with no luck. This is what I get:
Channel SIP/gafachi-081c81a8 was answered.
-- Executing [EMAIL PROTECTED]:1] Set(SIP/gafachi-081c81a8,
CALLERID(number)=66) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/gafachi-081c81a8,
CALLERID(name)=Robocop) in new stack
-- Executing [EMAIL PROTECTED]:3] AMD(SIP/gafachi-081c81a8,
) in new stack
-- AMD: SIP/gafachi-081c81a8 55 (null) (Fmt: 4)
-- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Channel [SIP/gafachi-081c81a8]. Too long...
-- AMD: Channel [SIP/gafachi-081c81a8]. Too long...
-- Executing [EMAIL PROTECTED]:4] GotoIf(SIP/gafachi-081c81a8,
0?human:machine) in new stack
-- Goto (robocop2,15155515509,7)
-- Executing [EMAIL PROTECTED]:7]
WaitForSilence(SIP/gafachi-081c81a8, 4000|2) in new stack
-- Waiting 2 time(s) for 4000 ms silence with 0 timeout
-- Executing [EMAIL PROTECTED]:8]
Playback(SIP/gafachi-081c81a8, mr-roboto-short) in new stack
-- SIP/gafachi-081c81a8 Playing 'mr-roboto-short' (language 'en')
== Spawn extension (robocop2, 15155515509, 8) exited non-zero on
'SIP/gafachi-081c81a8'
[Mar 19 23:45:42] NOTICE[12477]: pbx_spool.c:351 attempt_thread: Call
completed
to SIP/[EMAIL PROTECTED]
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: BYE
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: REGISTER
I think the problem is right here:
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Channel [SIP/gafachi-081c81a8]. Too long...
-- AMD: Channel [SIP/gafachi-081c81a8]. Too long...
It does this part without waiting the full 5000 ms it should before
biffing out - as far as I can tell it isn't waiting at all. I haven't
done anything with ztdummy - is that my problem? It seems like it could
be a timing issue, but I couldn't find any mention of ztdummy being
needed for AMD. I have seen some stuff that talks about frames just not
being sent over VOIP when there is silence, so maybe it has something to
do with that? I have updated to the latest build of app_amd.c and still
have had no luck, although originally things would just hang completely
on AMD so it was a step in the right direction. :-)
Here's my dialplan:
[robocop2]
exten = _1NXXNXX,1,Set(CALLERID(number)=66)
exten = _1NXXNXX,n,Set(CALLERID(name)=Robocop)
exten = _1NXXNXX,n,AMD
exten = _1NXXNXX,n,GotoIf($[${AMDSTATUS}=HUMAN]?human:machine)
exten = _1NXXNXX,n(human),Playback(rick-roll-short)
exten = _1NXXNXX,n,Hangup
exten = _1NXXNXX,n(machine),WaitForSilence(5000)
exten = _1NXXNXX,n,Playback(mr-roboto-short)
exten = _1NXXNXX,n,Hangup
I'm running all this on fedora core 4 with a 2.6.16 kernel. Happy to
provide any more information that would be useful in helping me solve
this problem. It is driving me nuts!
--
Drew Miller
Iowa Democratic Party
Information Technology Director
Office: (515) 974-1682
Cell: (515) 451-4509
AIM: ItsDrewMiller
MSN: [EMAIL PROTECTED]
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[asterisk-users] AMD on a SIP trunk...
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 [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CIDTEMP=49875calllogId=135514 016566275538) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],2, SIP/juarez-60/6275538|25|C) in new stack -- Called juarez-60/6275538 -- SIP/juarez-60-0892f740 is making progress passing it to Local/[EMAIL PROTECTED],2 -- SIP/juarez-60-0892f740 answered Local/[EMAIL PROTECTED],2 -- Executing [EMAIL PROTECTED]:1] Answer(Local/[EMAIL PROTECTED],1, ) in new stack -- Executing [EMAIL PROTECTED]:2] AMD(Local/[EMAIL PROTECTED],1, ) in new stack -- AMD: Local/[EMAIL PROTECTED],1 016566275538 (null) (Fmt: 64) -- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] AMD just stops and it takes over a minute until the line is dropped. The same number dialed through Zap works without a hitch. What could be the reason? If I dial the same number without AMD I can talk to the other person so I know the SIP line is fine. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD on a SIP trunk...
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 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 [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CIDTEMP=49875calllogId=135514 016566275538) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],2, SIP/juarez-60/6275538|25|C) in new stack -- Called juarez-60/6275538 -- SIP/juarez-60-0892f740 is making progress passing it to Local/[EMAIL PROTECTED],2 -- SIP/juarez-60-0892f740 answered Local/[EMAIL PROTECTED],2 -- Executing [EMAIL PROTECTED]:1] Answer(Local/[EMAIL PROTECTED],1, ) in new stack -- Executing [EMAIL PROTECTED]:2] AMD(Local/[EMAIL PROTECTED],1, ) in new stack -- AMD: Local/[EMAIL PROTECTED],1 016566275538 (null) (Fmt: 64) -- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] AMD just stops and it takes over a minute until the line is dropped. The same number dialed through Zap works without a hitch. What could be the reason? If I dial the same number without AMD I can talk to the other person so I know the SIP line is fine. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to production. If anyone is using this successfully in a production environment I would really appreciate any posts of settings you are using. My settings are as follows:AMD(3500|1500|300|5000|120|50|5|256)Thank you. Alan. Want to be your own boss? Learn how on Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMD Machine Detect
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 not so well in others. Im currently experimenting with the dialplan app BackgroundDetect. For voice broadcasting apps, BackgroundDetect has the advantage of playing the message to the caller while simultaneously listening for a live caller or an answering machine. This gets rid of the annoying pause that the caller hears after saying, Hello. Heres where I got the idea: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundDetect See the section Basic Answering Machine Detection. HtH, MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al 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 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 proving to be very difficult tweaking the settings to get it to work reliable enough to go to production. If anyone is using this successfully in a production environment I would really appreciate any posts of settings you are using. My settings are as follows: AMD(3500|1500|300|5000|120|50|5|256) Thank you. Alan. Want to be your own boss? Learn how on Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMD Machine Detect
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 message twice, figuring that if it missed it the first time around then it would record it the second time. Even though this method was more reliable it is, in my eyes, not as professional as leaving the message only once and directly when the voicemail recording starts.How far did you get with background detect? When I'm in front of the server I'll reply with my settings but I'm interested to hear if you've had much success with it?Thanks, Al.Michael Collins [EMAIL PROTECTED] wrote: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 not so well in others.Im currently experimenting with the dialplan app BackgroundDetect. For voice broadcasting apps, BackgroundDetect has the advantage of playing the message to the caller while simultaneously listening for a live caller or an answering machine. This gets rid of the annoying pause that the caller hears after saying, Hello. Heres where I got the idea: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundDetectSee the section Basic Answering Machine Detection.HtH, MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al LougherSent: Wednesday, June 21, 2006 9:24 AMTo: asterisk-users@lists.digium.comSubject: [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 and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to production. If anyone is using this successfully in a production environment I would really appreciate any posts of settings you are using. My settings are as follows:AMD(3500|1500|300|5000|120|50|5|256)Thank you.Alan. Want to be your own boss? Learn how on Yahoo! Small Business. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
