Tone Detection
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ic3m0nst3r
Jan 19, 2009, 4:18:44 AM1/19/09
to Adhearsion
This is actually related to the Predictive Thingy... This will be a
total no0b question. hehehe. Is there anyway that adhearsion detects
the tones, like answering machines, fax tones, voicemails etc. In some
point i know it can be part of asterisk mainly, haven't checked
lately, but if so, that asterisk is the one to rely on the Tone
Detection/Call Detection, does adhearsion return codes for this
feature...
I think in some ways I make sense.
Thanks!
total no0b question. hehehe. Is there anyway that adhearsion detects
the tones, like answering machines, fax tones, voicemails etc. In some
point i know it can be part of asterisk mainly, haven't checked
lately, but if so, that asterisk is the one to rely on the Tone
Detection/Call Detection, does adhearsion return codes for this
feature...
I think in some ways I make sense.
Thanks!
JasonGoecke
Jan 21, 2009, 8:23:27 PM1/21/09
to Adhearsion
This is a function of Asterisk and the underlying network in use.
Generally the progress messages such as answer supervision, busy, SIT
tones, etc are all handled at the signaling level. This would be SIP/
IAX2 or in the case of TDM the D-Channel via ISDN. You would need to
do the combination of a Originate command and then monitor the events
via events.rb to get these details to determine the final status of
the call attempt.
In terms of answering machine detection this once again is a function
of Asterisk and there are a number of approaches, one of which may be
found here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundDetect
In this case once you have Answer Supervision you would have the call
go to a context which calls an AGI to the Adhearsion dialplan. The
dialplan could then execute that command, see here:
http://api.adhearsion.com/Adhearsion/VoIP/Asterisk/Commands.html#execute-instance_method
Generally the progress messages such as answer supervision, busy, SIT
tones, etc are all handled at the signaling level. This would be SIP/
IAX2 or in the case of TDM the D-Channel via ISDN. You would need to
do the combination of a Originate command and then monitor the events
via events.rb to get these details to determine the final status of
the call attempt.
In terms of answering machine detection this once again is a function
of Asterisk and there are a number of approaches, one of which may be
found here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundDetect
In this case once you have Answer Supervision you would have the call
go to a context which calls an AGI to the Adhearsion dialplan. The
dialplan could then execute that command, see here:
http://api.adhearsion.com/Adhearsion/VoIP/Asterisk/Commands.html#execute-instance_method
VICIDIAL
Jan 21, 2009, 8:39:32 PM1/21/09
to adhea...@googlegroups.com
Audio SIT tones are not detected by anything in Asterisk currently.
The closest thing would be AMD, WaitForSilence, NVfaxdetect,
BackGroundDetect, etc... but those only work AFTER receiving an Answer
signal on a call which is not usually the case with SIT tone calls.
Unfortunately PRI/SIP termination codes in the USA are not consistent
across carriers.
MATT---
--
MATT---
The astGUIclient/VICIDIAL project is sponsored by
Binfone Telecom - http://www.binfone.com
The closest thing would be AMD, WaitForSilence, NVfaxdetect,
BackGroundDetect, etc... but those only work AFTER receiving an Answer
signal on a call which is not usually the case with SIT tone calls.
Unfortunately PRI/SIP termination codes in the USA are not consistent
across carriers.
MATT---
MATT---
The astGUIclient/VICIDIAL project is sponsored by
Binfone Telecom - http://www.binfone.com
ic3m0nst3r
Jan 21, 2009, 8:58:11 PM1/21/09
to Adhearsion
Thanks guys. I'll explore this one also.
@jason --- have you tested this one?
@matt --- Is vicidial using this right now?
Thanks
On Jan 22, 9:39 am, VICIDIAL <vicid...@gmail.com> wrote:
> Audio SIT tones are not detected by anything in Asterisk currently.
> The closest thing would be AMD, WaitForSilence, NVfaxdetect,
> BackGroundDetect, etc... but those only work AFTER receiving an Answer
> signal on a call which is not usually the case with SIT tone calls.
> Unfortunately PRI/SIP termination codes in the USA are not consistent
> across carriers.
>
> MATT---
>
>
>
> On Wed, Jan 21, 2009 at 8:23 PM, JasonGoecke <jsgoe...@gmail.com> wrote:
>
> > This is a function of Asterisk and the underlying network in use.
> > Generally the progress messages such as answer supervision, busy, SIT
> > tones, etc are all handled at the signaling level. This would be SIP/
> > IAX2 or in the case of TDM the D-Channel via ISDN. You would need to
> > do the combination of a Originate command and then monitor the events
> > via events.rb to get these details to determine the final status of
> > the call attempt.
>
> > In terms of answering machine detection this once again is a function
> > of Asterisk and there are a number of approaches, one of which may be
> > found here:
>
> >http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundD...
@jason --- have you tested this one?
@matt --- Is vicidial using this right now?
Thanks
On Jan 22, 9:39 am, VICIDIAL <vicid...@gmail.com> wrote:
> Audio SIT tones are not detected by anything in Asterisk currently.
> The closest thing would be AMD, WaitForSilence, NVfaxdetect,
> BackGroundDetect, etc... but those only work AFTER receiving an Answer
> signal on a call which is not usually the case with SIT tone calls.
> Unfortunately PRI/SIP termination codes in the USA are not consistent
> across carriers.
>
> MATT---
>
>
>
> On Wed, Jan 21, 2009 at 8:23 PM, JasonGoecke <jsgoe...@gmail.com> wrote:
>
> > This is a function of Asterisk and the underlying network in use.
> > Generally the progress messages such as answer supervision, busy, SIT
> > tones, etc are all handled at the signaling level. This would be SIP/
> > IAX2 or in the case of TDM the D-Channel via ISDN. You would need to
> > do the combination of a Originate command and then monitor the events
> > via events.rb to get these details to determine the final status of
> > the call attempt.
>
> > In terms of answering machine detection this once again is a function
> > of Asterisk and there are a number of approaches, one of which may be
> > found here:
>
>
> > In this case once you have Answer Supervision you would have the call
> > go to a context which calls an AGI to the Adhearsion dialplan. The
> > dialplan could then execute that command, see here:
>
> >http://api.adhearsion.com/Adhearsion/VoIP/Asterisk/Commands.html#exec...
> > In this case once you have Answer Supervision you would have the call
> > go to a context which calls an AGI to the Adhearsion dialplan. The
> > dialplan could then execute that command, see here:
>
JasonGoecke
Jan 21, 2009, 9:06:26 PM1/21/09
to Adhearsion
Correct on Audio SIT Tones. You still do get SIP/ISDN messages that
the number is either temporarily or permanently unavailable depending
on the trunks you are using, as you are correct that you do not get
answer supervision in that case. It is important that anytime you
deploy a new system that you test thoroughly with your provider if you
would like to have a more reliable system in terms of detecting all
appropriate cause codes. Having said this, some of the carriers in
between your system and the person you are trying to reach may present
problems of their own. If only standards were consistent...
In some cases I have found that folks actually want the SIT tones
transferred to a human in some cases so that they may record the new
number when it is available. What would be more interesting here is
using something like Automatic Speech Recognition to capture these
details when a SIT is detected.
On Jan 21, 5:39 pm, VICIDIAL <vicid...@gmail.com> wrote:
> Audio SIT tones are not detected by anything in Asterisk currently.
> The closest thing would be AMD, WaitForSilence, NVfaxdetect,
> BackGroundDetect, etc... but those only work AFTER receiving an Answer
> signal on a call which is not usually the case with SIT tone calls.
> Unfortunately PRI/SIP termination codes in the USA are not consistent
> across carriers.
>
> MATT---
>
>
>
> On Wed, Jan 21, 2009 at 8:23 PM, JasonGoecke <jsgoe...@gmail.com> wrote:
>
> > This is a function of Asterisk and the underlying network in use.
> > Generally the progress messages such as answer supervision, busy, SIT
> > tones, etc are all handled at the signaling level. This would be SIP/
> > IAX2 or in the case of TDM the D-Channel via ISDN. You would need to
> > do the combination of a Originate command and then monitor the events
> > via events.rb to get these details to determine the final status of
> > the call attempt.
>
> > In terms of answering machine detection this once again is a function
> > of Asterisk and there are a number of approaches, one of which may be
> > found here:
>
> >http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundD...
the number is either temporarily or permanently unavailable depending
on the trunks you are using, as you are correct that you do not get
answer supervision in that case. It is important that anytime you
deploy a new system that you test thoroughly with your provider if you
would like to have a more reliable system in terms of detecting all
appropriate cause codes. Having said this, some of the carriers in
between your system and the person you are trying to reach may present
problems of their own. If only standards were consistent...
In some cases I have found that folks actually want the SIT tones
transferred to a human in some cases so that they may record the new
number when it is available. What would be more interesting here is
using something like Automatic Speech Recognition to capture these
details when a SIT is detected.
On Jan 21, 5:39 pm, VICIDIAL <vicid...@gmail.com> wrote:
> Audio SIT tones are not detected by anything in Asterisk currently.
> The closest thing would be AMD, WaitForSilence, NVfaxdetect,
> BackGroundDetect, etc... but those only work AFTER receiving an Answer
> signal on a call which is not usually the case with SIT tone calls.
> Unfortunately PRI/SIP termination codes in the USA are not consistent
> across carriers.
>
> MATT---
>
>
>
> On Wed, Jan 21, 2009 at 8:23 PM, JasonGoecke <jsgoe...@gmail.com> wrote:
>
> > This is a function of Asterisk and the underlying network in use.
> > Generally the progress messages such as answer supervision, busy, SIT
> > tones, etc are all handled at the signaling level. This would be SIP/
> > IAX2 or in the case of TDM the D-Channel via ISDN. You would need to
> > do the combination of a Originate command and then monitor the events
> > via events.rb to get these details to determine the final status of
> > the call attempt.
>
> > In terms of answering machine detection this once again is a function
> > of Asterisk and there are a number of approaches, one of which may be
> > found here:
>
>
> > In this case once you have Answer Supervision you would have the call
> > go to a context which calls an AGI to the Adhearsion dialplan. The
> > dialplan could then execute that command, see here:
>
> >http://api.adhearsion.com/Adhearsion/VoIP/Asterisk/Commands.html#exec...
> > In this case once you have Answer Supervision you would have the call
> > go to a context which calls an AGI to the Adhearsion dialplan. The
> > dialplan could then execute that command, see here:
>
VICIDIAL
Jan 21, 2009, 9:08:12 PM1/21/09
to adhea...@googlegroups.com
VICIDIAL can use app_amd if you choose it to be used, but the only
comprehensive call progress analyzer(SIT tones, AMD, Fax, etc...)
we've successfully used is a proprietary add-on, the Sangoma
CPA(formerly ParaXip).
MATT---
comprehensive call progress analyzer(SIT tones, AMD, Fax, etc...)
we've successfully used is a proprietary add-on, the Sangoma
CPA(formerly ParaXip).
MATT---
JasonGoecke
Jan 22, 2009, 12:11:18 AM1/22/09
to Adhearsion
Good find, I was not aware of that app and its compatibility with
Sangoma. Will definitely be digging into that one.
Sangoma. Will definitely be digging into that one.
VICIDIAL
Jan 22, 2009, 12:18:02 AM1/22/09
to adhea...@googlegroups.com
Hello,
The Sangoma CPA is actually an entirely SIP based application and does
not require any Sangoma hardware at all. The one downside of it, it
has to be run on Windows and on an Intel-based server only.
MATT---
The Sangoma CPA is actually an entirely SIP based application and does
not require any Sangoma hardware at all. The one downside of it, it
has to be run on Windows and on an Intel-based server only.
MATT---
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