Jonathan H
2018-Jan-20 23:57 UTC
[asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote:> I have seen this take over 2 seconds before on a sluggish machine.Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause.> you'd need to pipe that to a Google Speech API tunnel. > That's probably not something you can hack away at with simple > Asterisk dialplan applications.Funnily enough, I had just found an old reply from last year to another similar question:> ---------- Forwarded message ---------- > From: Matt Riddell > Date: 22 September 2017 at 16:01 > Subject: Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition? so many questions! > At least in older versions you can use EAGI to get a handle to the audio stream.So I had a look and found this: https://stackoverflow.com/questions/34026698/asterisk-write-plugin-to-catch-voice-stream And read this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_EAGI There's a few knowledge gaps, but I think with a few days reading and the great help here, we might have a solution :) This is all very helpful - if anyone else feels like wading in, please do. Many thanks!
Dmitriy Serov
2018-Jan-21 08:30 UTC
[asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
Hello.
A little sub from my dialplan:
[sub-Read]
exten => s,1,NoOp(Read)
?same => n,Set(LOCAL(tmp_record_file)=/tmp/asterisk-in/${EPOCH})
?same => n,Monitor(wav16,${tmp_record_file},o)
?same => n,Read(tmp_ext,${ARG2},${ARG3},${ARG4},${ARG5},${ARG6})
?same => n,StopMonitor()
?same => n,NoOp(ReadStatus=${READSTATUS})
?same => n,Gotoif($[ ${LEN(${tmp_ext})} > 0 ]?end)
?same =>
n,AGI(agi-ruvoip-net.php,speeddial-voice,${ARG1},${tmp_record_file}-in.wav16)
?same => n,NoOp(Voice recognition result: "${agi_result}")
?same => n,Gotoif($[ "${agi_result}" != "found" ]?end)
?same => n,Return(${agi_call_exten})
?same => n(end),return(${tmp_ext})
21.01.2018 2:57, Jonathan H ?????:> On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote:
>
>> I have seen this take over 2 seconds before on a sluggish machine.
> Thanks - my host uses SSD and everything seems pretty quick, but I'll
> give it a 1 second pause.
>
>> you'd need to pipe that to a Google Speech API tunnel.
>> That's probably not something you can hack away at with simple
>> Asterisk dialplan applications.
> Funnily enough, I had just found an old reply from last year to
> another similar question:
>
>> ---------- Forwarded message ----------
>> From: Matt Riddell
>> Date: 22 September 2017 at 16:01
>> Subject: Re: [asterisk-users] Asterisk 15, Jack, streams, speech
recognition? so many questions!
>> At least in older versions you can use EAGI to get a handle to the
audio stream.
> So I had a look and found this:
>
>
https://stackoverflow.com/questions/34026698/asterisk-write-plugin-to-catch-voice-stream
>
> And read this:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_EAGI
>
> There's a few knowledge gaps, but I think with a few days reading and
> the great help here, we might have a solution :)
>
> This is all very helpful - if anyone else feels like wading in, please do.
>
> Many thanks!
>
Jonathan H
2018-Jan-23 14:51 UTC
[asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
Hi Dmitry and Tim (and everyone else with input into this thread) Just wanted to thank you all; with your guidance, I've managed to bolt something very clean and efficient together using Dmity and Tim's templates, piped into the ding-dong npm node package, which calls Google Speech API node package. What I particularly like about DingDong is that it's well documented insofar as it's so simple, it barely needs documentation! Once I've finished experimenting in a day or two, I'll stick a Gist up with everything together. Once again, many, many thanks. Refs: https://github.com/antirek/ding-dong https://github.com/googleapis/nodejs-speech On 21 January 2018 at 08:30, Dmitriy Serov <serov.d.p at gmail.com> wrote:> Hello. > > A little sub from my dialplan: > > [sub-Read] > exten => s,1,NoOp(Read) > same => n,Set(LOCAL(tmp_record_file)=/tmp/asterisk-in/${EPOCH}) > same => n,Monitor(wav16,${tmp_record_file},o) > same => n,Read(tmp_ext,${ARG2},${ARG3},${ARG4},${ARG5},${ARG6}) > same => n,StopMonitor() > same => n,NoOp(ReadStatus=${READSTATUS}) > same => n,Gotoif($[ ${LEN(${tmp_ext})} > 0 ]?end) > same => n,AGI(agi-ruvoip-net.php,speeddial-voice,${ARG1},${tmp_recor > d_file}-in.wav16) > same => n,NoOp(Voice recognition result: "${agi_result}") > same => n,Gotoif($[ "${agi_result}" != "found" ]?end) > same => n,Return(${agi_call_exten}) > same => n(end),return(${tmp_ext}) > > > 21.01.2018 2:57, Jonathan H ?????: > > On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: >> >> I have seen this take over 2 seconds before on a sluggish machine. >>> >> Thanks - my host uses SSD and everything seems pretty quick, but I'll >> give it a 1 second pause. >> >> you'd need to pipe that to a Google Speech API tunnel. >>> That's probably not something you can hack away at with simple >>> Asterisk dialplan applications. >>> >> Funnily enough, I had just found an old reply from last year to >> another similar question: >> >> ---------- Forwarded message ---------- >>> From: Matt Riddell >>> Date: 22 September 2017 at 16:01 >>> Subject: Re: [asterisk-users] Asterisk 15, Jack, streams, speech >>> recognition? so many questions! >>> At least in older versions you can use EAGI to get a handle to the audio >>> stream. >>> >> So I had a look and found this: >> >> https://stackoverflow.com/questions/34026698/asterisk-write- >> plugin-to-catch-voice-stream >> >> And read this: >> >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_EAGI >> >> There's a few knowledge gaps, but I think with a few days reading and >> the great help here, we might have a solution :) >> >> This is all very helpful - if anyone else feels like wading in, please do. >> >> Many thanks! >> >> > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180123/15a3cd7e/attachment.html>