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aeap-speech-to-text's Introduction

The Asterisk(R) Open Source PBX

        By Mark Spencer <[email protected]> and the Asterisk.org developer community.
        Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.

SECURITY

It is imperative that you read and fully understand the contents of the security information document before you attempt to configure and run an Asterisk server.

See Important Security Considerations for more information.

WHAT IS ASTERISK ?

Asterisk is an Open Source PBX and telephony toolkit. It is, in a sense, middleware between Internet and telephony channels on the bottom, and Internet and telephony applications at the top. However, Asterisk supports more telephony interfaces than just Internet telephony. Asterisk also has a vast amount of support for traditional PSTN telephony, as well.

For more information on the project itself, please visit the Asterisk home page and the official documentation. In addition you'll find lots of information compiled by the Asterisk community at voip-info.org.

There is a book on Asterisk published by O'Reilly under the Creative Commons License. It is available in book stores as well as in a downloadable version on the asteriskdocs.org web site.

SUPPORTED OPERATING SYSTEMS

Linux

The Asterisk Open Source PBX is developed and tested primarily on the GNU/Linux operating system, and is supported on every major GNU/Linux distribution.

Others

Asterisk has also been 'ported' and reportedly runs properly on other operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin, and the BSD variants.

GETTING STARTED

First, be sure you've got supported hardware (but note that you don't need ANY special hardware, not even a sound card) to install and run Asterisk.

Supported telephony hardware includes:

  • All Analog and Digital Interface cards from Sangoma
  • QuickNet Internet PhoneJack and LineJack
  • any full duplex sound card supported by ALSA, OSS, or PortAudio
  • any ISDN card supported by mISDN on Linux
  • The Xorcom Astribank channel bank
  • VoiceTronix OpenLine products

UPGRADING FROM AN EARLIER VERSION

If you are updating from a previous version of Asterisk, make sure you read the UPGRADE.txt file in the source directory. There are some files and configuration options that you will have to change, even though we made every effort possible to maintain backwards compatibility.

In order to discover new features to use, please check the configuration examples in the configs directory of the source code distribution. For a list of new features in this version of Asterisk, see the CHANGES file.

NEW INSTALLATIONS

Ensure that your system contains a compatible compiler and development libraries. Asterisk requires either the GNU Compiler Collection (GCC) version 4.1 or higher, or a compiler that supports the C99 specification and some of the gcc language extensions. In addition, your system needs to have the C library headers available, and the headers and libraries for ncurses.

There are many modules that have additional dependencies. To see what libraries are being looked for, see ./configure --help, or run make menuselect to view the dependencies for specific modules.

On many distributions, these dependencies are installed by packages with names like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.

So, let's proceed:

  1. Read this file.

There are more documents than this one in the doc directory. You may also want to check the configuration files that contain examples and reference guides in the configs directory.

  1. Run ./configure

Execute the configure script to guess values for system-dependent variables used during compilation. If the script indicates that some required components are missing, you can run ./contrib/scripts/install_prereq install to install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need to rerun ./configure.

  1. Run make menuselect [optional]

This is needed if you want to select the modules that will be compiled and to check dependencies for various optional modules.

  1. Run make

Assuming the build completes successfully:

  1. Run make install

If this is your first time working with Asterisk, you may wish to install the sample PBX, with demonstration extensions, etc. If so, run:

  1. Run make samples

Doing so will overwrite any existing configuration files you have installed.

  1. Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
        # asterisk -vvvc

You'll see a bunch of verbose messages fly by your screen as Asterisk initializes (that's the "very very verbose" mode). When it's ready, if you specified the "c" then you'll get a command line console, that looks like this:

        *CLI>

You can type "core show help" at any time to get help with the system. For help with a specific command, type "core show help ". To start the PBX using your sound card, you can type "console dial" to dial the PBX. Then you can use "console answer", "console hangup", and "console dial" to simulate the actions of a telephone. Remember that if you don't have a full duplex sound card (and Asterisk will tell you somewhere in its verbose messages if you do/don't) then it won't work right (not yet).

"man asterisk" at the Unix/Linux command prompt will give you detailed information on how to start and stop Asterisk, as well as all the command line options for starting Asterisk.

Feel free to look over the configuration files in /etc/asterisk, where you will find a lot of information about what you can do with Asterisk.

ABOUT CONFIGURATION FILES

All Asterisk configuration files share a common format. Comments are delimited by ';' (since '#' of course, being a DTMF digit, may occur in many places). A configuration file is divided into sections whose names appear in []'s. Each section typically contains two types of statements, those of the form 'variable = value', and those of the form 'object => parameters'. Internally the use of '=' and '=>' is exactly the same, so they're used only to help make the configuration file easier to understand, and do not affect how it is actually parsed.

Entries of the form 'variable=value' set the value of some parameter in asterisk. For example, in chan_dahdi.conf, one might specify:

	switchtype=national

In order to indicate to Asterisk that the switch they are connecting to is of the type "national". In general, the parameter will apply to instantiations which occur below its specification. For example, if the configuration file read:

	switchtype = national
	channel => 1-4
	channel => 10-12
	switchtype = dms100
	channel => 25-47

The "national" switchtype would be applied to channels one through four and channels 10 through 12, whereas the "dms100" switchtype would apply to channels 25 through 47.

The "object => parameters" instantiates an object with the given parameters. For example, the line "channel => 25-47" creates objects for the channels 25 through 47 of the card, obtaining the settings from the variables specified above.

SPECIAL NOTE ON TIME

Those using SIP phones should be aware that Asterisk is sensitive to large jumps in time. Manually changing the system time using date(1) (or other similar commands) may cause SIP registrations and other internal processes to fail. If your system cannot keep accurate time by itself use NTP to keep the system clock synchronized to "real time". NTP is designed to keep the system clock synchronized by speeding up or slowing down the system clock until it is synchronized to "real time" rather than by jumping the time and causing discontinuities. Most Linux distributions include precompiled versions of NTP. Beware of some time synchronization methods that get the correct real time periodically and then manually set the system clock.

Apparent time changes due to daylight savings time are just that, apparent. The use of daylight savings time in a Linux system is purely a user interface issue and does not affect the operation of the Linux kernel or Asterisk. The system clock on Linux kernels operates on UTC. UTC does not use daylight savings time.

Also note that this issue is separate from the clocking of TDM channels, and is known to at least affect SIP registrations.

FILE DESCRIPTORS

Depending on the size of your system and your configuration, Asterisk can consume a large number of file descriptors. In UNIX, file descriptors are used for more than just files on disk. File descriptors are also used for handling network communication (e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and digital trunk hardware). Asterisk accesses many on-disk files for everything from configuration information to voicemail storage.

Most systems limit the number of file descriptors that Asterisk can have open at one time. This can limit the number of simultaneous calls that your system can handle. For example, if the limit is set at 1024 (a common default value) Asterisk can handle approximately 150 SIP calls simultaneously. To change the number of file descriptors follow the instructions for your system below:

PAM-BASED LINUX SYSTEM

If your system uses PAM (Pluggable Authentication Modules) edit /etc/security/limits.conf. Add these lines to the bottom of the file:

root            soft    nofile          4096
root            hard    nofile          8196
asterisk        soft    nofile          4096
asterisk        hard    nofile          8196

(adjust the numbers to taste). You may need to reboot the system for these changes to take effect.

GENERIC UNIX SYSTEM

If there are no instructions specifically adapted to your system above you can try adding the command ulimit -n 8192 to the script that starts Asterisk.

MORE INFORMATION

See the doc directory for more documentation on various features. Again, please read all the configuration samples that include documentation on the configuration options.

Finally, you may wish to visit the support site and join the mailing list if you're interested in getting more information.

Welcome to the growing worldwide community of Asterisk users!

        Mark Spencer, and the Asterisk.org development community

Asterisk is a trademark of Sangoma Technologies Corporation

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aeap-speech-to-text's Issues

Works, but only transcripts the first 10 seconds

Hi, everything is working fine. I got the whole project working so I can call the extension and say a sentence and then I got the response from Google with the exact text I said. No problem there. BUT, it only lets me speak for around 10 seconds. Even when the sentence is like 5 seconds long, I still have to wait 5 seconds more for the response to arrive. Is this by design? Is this project not able to detect silence? Is it a limitation with AEAP or is some kind of configuration I have to do I havent found yet? Any help will be appreciated.

Cannot read property 'streamingRecognize' of undefined

hello

i trying to test AEAP on Version 18.12.1

but some error when dialplan goes on

like this

ERROR[14264]: res_aeap/aeap.c:204 raise_msg_handler: AEAP (0x7f64cc002660): Cannot read property 'streamingRecognize' of undefined

do you have any idea for fix it ?

Asterisk Crash after script restart

thanks for your source code, there is only one example how to use aeap with google speech api and its yours.
my problem is that when I stop and restart the index script and try to call again, the asterisk crash with segmentation fault.
My Asterisk Version is 20.4.0
my dial plan:

exten => 550,1,NoOp()
        same => n,Answer()
        same => n,SpeechCreate(my-speech-to-text)
        same => n,NoOp(${ERROR})
        same => n,GotoIf($["${ERROR}" = "1"]?fail:next)
        same => n(next),SpeechStart()
        same => n,SpeechBackground(tt-monkeys)
        same => n,Verbose(0,${SPEECH_TEXT(0)})
        same => n,SpeechDestroy()
        same => n(fail),Hangup()

this is my aeap.conf:

[my-speech-to-text]
type=client
codecs=!all,ulaw
url=ws://127.0.0.1:9099/filestream
protocol=speech_to_text

it works perfectly but after I stop index.js and start it again, this is what I got:

       > 0x7fb57000e350 -- Strict RTP learning after remote address set to: 192.168.97.12:40048
    -- Executing [550@tgui-out:1] NoOp("SIP/121-00000003", "") in new stack
    -- Executing [550@tgui-out:2] Answer("SIP/121-00000003", "") in new stack
       > 0x7fb57000e350 -- Strict RTP switching to RTP target address 192.168.97.12:40048 as source
    -- Executing [550@tgui-out:3] SpeechCreate("SIP/121-00000003", "my-speech-to-text") in new stack
[Nov 11 15:18:09] ERROR[2736897]: res_aeap/transaction.c:148 transaction_end: AEAP transaction (0x7fb57001ec20): message 'setup' timed out
    -- Executing [550@tgui-out:4] NoOp("SIP/121-00000003", "1") in new stack
    -- Executing [550@tgui-out:5] GotoIf("SIP/121-00000003", "1?fail:next") in new stack
    -- Goto (tgui-out,550,10)
    -- Executing [550@tgui-out:10] Hangup("SIP/121-00000003", "") in new stack
  == Spawn extension (tgui-out, 550, 10) exited non-zero on 'SIP/121-00000003'
    -- Executing [h@tgui-out:1] Set("SIP/121-00000003", "COUNT=0") in new stack
    -- Executing [h@tgui-out:2] NoOp("SIP/121-00000003", "Hangup Cause: 16") in new stack
    -- Executing [h@tgui-out:3] NoOp("SIP/121-00000003", "") in new stack
    -- Executing [h@tgui-out:4] NoOp("SIP/121-00000003", "Who Hangup: ") in new stack
    -- Executing [h@tgui-out:5] AGI("SIP/121-00000003", "agi://127.0.0.1/tgui.agi?status=end") in new stack
 agi://127.0.0.1/tgui.agi?status=end: No script configured for URL 'agi://127.0.0.1/tgui.agi?status=end' (script 'tgui.agi')
    -- <SIP/121-00000003>AGI Script agi://127.0.0.1/tgui.agi?status=end completed, returning 0
    -- Executing [h@tgui-out:6] Set("SIP/121-00000003", "COUNT=1") in new stack
    -- Executing [h@tgui-out:7] GotoIf("SIP/121-00000003", "0?retry:checkstate") in new stack
    -- Goto (tgui-out,h,10)
    -- Executing [h@tgui-out:10] GotoIf("SIP/121-00000003", "0?reset:success") in new stack
    -- Goto (tgui-out,h,15)
    -- Executing [h@tgui-out:15] NoOp("SIP/121-00000003", "Agi-Success") in new stack
Segmentation fault (core dumped)

this is the dump info:

Stack trace of thread 2737132:
                #0  0x0000559352315ef4 ast_format_get_codec_name (asterisk + 0xf3ef4)
                #1  0x00007fb559cafc44 handle_response_setup (res_speech_aeap.so + 0x2c44)
                #2  0x00007fb559cbd4c9 aeap_receive (res_aeap.so + 0x84c9)
                #3  0x00005593523e917f dummy_start (asterisk + 0x1c717f)
                #4  0x00007fb57a69f802 start_thread (libc.so.6 + 0x9f802)
                #5  0x00007fb57a63f450 __clone3 (libc.so.6 + 0x3f450)

asterisk-16.15.0 support aeap ?

Hi,

I was trying to implement this on asterisk-16.15.0 but it doesn't work call just failed.
This is supported on asterisk-16.15.0?

How to use SpeechProcessingSound() ?

Hi,

I've tried using speechProcessingSound() dialplan like this.

exten => speech,1,Verbose(0, This is Dynamic sub-speech-dynamic-google SERVER:${ARG1}, RESPONSE-TYPE:${ARG2}, NUM:${ARG3} )
same => n,Set(RESPONSE-TYPE=${IF($["${ISNULL(${ARG2})}"="1"]?question:${ARG2})})
same => n,SpeechCreate(dynamic-google-${ARG1})
same => n,Set(SPEECH-ERROR=${ERROR})
same => n,ExecIf($["${ERROR}"="1"]?return())
same => n,SpeechProcessingSound(pls-hold-process-tx)) <----- here but, working not
same => n,SpeechStart()
same => n,SpeechBackground(dynamic-beep,${ARG4})
same => n,NoOp(상태확인: 1 이면 말하는 중, ${SPEECH(spoke)})
same => n,Verbose(0,Text:${SPEECH_TEXT(0)}, Score:${SPEECH_SCORE(0)})
same => n,Set(RESPONSE-${RESPONSE-TYPE}-${ARG3}=${STRREPLACE(SPEECH_TEXT(0)," ","")})
same => n,SpeechDestroy

It doesn't seem to work.

thanks

Can't speak for more than 4 seconds:

GoogleProvider: result: {"text":"test test testing hear me I don't know what else to tell you","score":70}
message: {"response":"set","id":"204bcf88-76a7-4e94-809c-f055e3726432"}

When I call and start speaking after "hello world" - it disconnects me after like 3-4 seconds.... How do I keep it permanent?

Unable to reproduce the readme example

Hi, I'm looking for options to stream asterisk audio to my STT engine and tried to reproduce the example in your readme but faced certain errors. Can you please give me pointers how should I debug and get more information about the cause of error?

    -- Executing [918046******@from-external:1] Goto("PJSIP/airtel-00000001", "zi-tts-demo,s,1") in new stack
    -- Goto (zi-tts-demo,s,1)
    -- Executing [s@zi-tts-demo:1] NoOp("PJSIP/airtel-00000001", "") in new stack
    -- Executing [s@zi-tts-demo:2] Answer("PJSIP/airtel-00000001", "") in new stack
    -- Executing [s@zi-tts-demo:3] Playback("PJSIP/airtel-00000001", "silence/1&hello-world") in new stack
    -- <PJSIP/airtel-00000001> Playing 'silence/1.gsm' (language 'en')
    -- <PJSIP/airtel-00000001> Playing 'hello-world.gsm' (language 'en')
    -- Executing [s@zi-tts-demo:4] SpeechCreate("PJSIP/airtel-00000001", "zi-tts-speech") in new stack
    -- Executing [s@zi-tts-demo:5] SpeechStart("PJSIP/airtel-00000001", "") in new stack
  == Spawn extension (zi-tts-demo, s, 5) exited non-zero on 'PJSIP/airtel-00000001'

I'm running the node application locally and exposed it with an ngrok tunnel ngrok http 9099,

❯ node index.js
Server on port '9099': started listening

I tested the websocket connection by trying to connect to ws://127.0.0.1:9099 ws://localhost:9099 or ws://<ngrok_url> and was able to make successful websocket connection.

On the asterisk server, my configuration files look like the following,

aeap.conf

[zi-tts-speech]
type=client
codecs=!all,ulaw
url=ws://348d-83-110-20-206.ngrok.io
protocol=speech_to_text

extensions.conf

[from-external]
exten => 918046******,1,Goto(zi-tts-demo,s,1)

[zi-tts-demo]
exten => s,1,NoOp()
  same => n,Answer()
  same => n,Playback(silence/1&hello-world)
  same => n,SpeechCreate(zi-tts-speech)
  same => n,SpeechStart()
  same => n,SpeechBackground(hello-world)
  same => n,Verbose(0,${SPEECH_TEXT(0)})
  same => n,SpeechDestroy()

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