Why AGI script terminating?

Asterisk Asterisk APIs
1 
test-VirtualBox*CLI> module load res_speech.so
Loaded res_speech.so
test-VirtualBox*CLI> module load res_agi.so
Loaded res_agi.so
test-VirtualBox*CLI> agi set debug on
AGI Debugging Enabled
test-VirtualBox*CLI> core set verbose 4
Console verbose was OFF and is now 4.
  == Setting global variable 'SIPDOMAIN' to '192.168.1.5'
    -- Executing [1234@Long-Distance:1] Answer("PJSIP/1101-00000000", "") in new stack
       > 0x7f81a00210c0 -- Strict RTP learning after remote address set to: 103.39.128.127:45606
       > 0x7f81a00210c0 -- Strict RTP qualifying stream type: audio
       > 0x7f81a00210c0 -- Strict RTP switching source address to 192.168.1.3:7078
    -- Executing [1234@Long-Distance:2] AGI("PJSIP/1101-00000000", "speech-recog.agi,en-US") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/speech-recog.agi
<PJSIP/1101-00000000>AGI Tx >> agi_request: speech-recog.agi
<PJSIP/1101-00000000>AGI Tx >> agi_channel: PJSIP/1101-00000000
<PJSIP/1101-00000000>AGI Tx >> agi_language: en
<PJSIP/1101-00000000>AGI Tx >> agi_type: PJSIP
<PJSIP/1101-00000000>AGI Tx >> agi_uniqueid: 1611306068.0
<PJSIP/1101-00000000>AGI Tx >> agi_version: 16.15.1
<PJSIP/1101-00000000>AGI Tx >> agi_callerid: 1101
<PJSIP/1101-00000000>AGI Tx >> agi_calleridname: Maria Berny
<PJSIP/1101-00000000>AGI Tx >> agi_callingpres: 0
<PJSIP/1101-00000000>AGI Tx >> agi_callingani2: 0
<PJSIP/1101-00000000>AGI Tx >> agi_callington: 0
<PJSIP/1101-00000000>AGI Tx >> agi_callingtns: 0
<PJSIP/1101-00000000>AGI Tx >> agi_dnid: 1234
<PJSIP/1101-00000000>AGI Tx >> agi_rdnis: unknown
<PJSIP/1101-00000000>AGI Tx >> agi_context: Long-Distance
<PJSIP/1101-00000000>AGI Tx >> agi_extension: 1234
<PJSIP/1101-00000000>AGI Tx >> agi_priority: 2
<PJSIP/1101-00000000>AGI Tx >> agi_enhanced: 0.0
<PJSIP/1101-00000000>AGI Tx >> agi_accountcode: 
<PJSIP/1101-00000000>AGI Tx >> agi_threadid: 140196669904640
<PJSIP/1101-00000000>AGI Tx >> agi_arg_1: en-US
<PJSIP/1101-00000000>AGI Tx >> 
<PJSIP/1101-00000000>AGI Rx << SET VARIABLE "confidence" "-1"
<PJSIP/1101-00000000>AGI Tx >> 200 result=1
<PJSIP/1101-00000000>AGI Rx << SET VARIABLE "utterance" "-1"
<PJSIP/1101-00000000>AGI Tx >> 200 result=1
<PJSIP/1101-00000000>AGI Rx << CHANNEL STATUS
<PJSIP/1101-00000000>AGI Tx >> 200 result=6
<PJSIP/1101-00000000>AGI Rx << GET FULL VARIABLE ${CHANNEL(audionativeformat)}
[Jan 22 14:31:08] ERROR[7489][C-00000001]: pbx_functions.c:651 ast_func_read2: Function CHANNEL not registered
<PJSIP/1101-00000000>AGI Tx >> 200 result=1 ()
<PJSIP/1101-00000000>AGI Rx << RECORD FILE /tmp/stt_tIZFBW sln "#" "5" BEEP "s=2"
    -- <PJSIP/1101-00000000> Playing 'beep.gsm' (language 'en')
[Jan 22 14:31:09] WARNING[7489][C-00000001]: file.c:1504 ast_writefile: No such format 'sln'
<PJSIP/1101-00000000>AGI Tx >> 200 result=-1 (writefile)
    -- <PJSIP/1101-00000000>AGI Script speech-recog.agi completed, returning -1
    -- Executing [1234@Long-Distance:3] Verbose("PJSIP/1101-00000000", "1,The text you just said is: -1") in new stack
 The text you just said is: -1
    -- Executing [1234@Long-Distance:4] Verbose("PJSIP/1101-00000000", "1,The probability to be right is: -1") in new stack
 The probability to be right is: -1
    -- Executing [1234@Long-Distance:5] Hangup("PJSIP/1101-00000000", "") in new stack
  == Spawn extension (Long-Distance, 1234, 5) exited non-zero on 'PJSIP/1101-00000000'

Edit:
I am trying to call agi script but it terminates and give an error like above what am I doing wrong?

My PJSIP.conf:
pjsip1.txt (944 Bytes)
My Extensions.conf:
extensions1.txt (322 Bytes)

2

Can anybody tell what is the error?

3

Not without the contents of the script.

4

speech-recog.txt (8.6 KB)

https://zaf.github.io/asterisk-speech-recog

Edit:
Official author of the above script is Lefteris Zafiris
hope this will help
Note: I can’t upload .agi file that’s why I changed the extension of a file to .txt.

Edit
Sorry for my half explanation and do not provide Name of the author.

5

As it says, Asterisk doesn’t have a format called sln.

I thought you wanted real time speech to text, but you seem to be recording the audio.

As a general point, you are using language features that write to standard error, but you don’t seem to have redirected standard error to a file, so finding the output of those may be difficult.

It’s interesting that you don’t use a class library for AGI. Whilst I have reservations about them, as they seem to add little value and obscure what is really happening, most people seem to use them.

6

yeah david but I made below step
./configure
make menuselect
make install

and in menuselect I select sln media type and install that.

And yes I am tried to real time speech to text but due to some leak of knowledge I first record that and then send to api for speech to text.

But David can you please tell me what is this,

Thank you

7

I don’t understand that, as I thought that that function was implemented in the Asterisk core. Maybe it is treating it as a request for a global variable, because you haen’t provided a channel name. Unfortunately the wiki, for GET FULL VARIABLE, appears to be wrong in multiple ways, but I don’t have time to look at the code to find out exactly how it does work.

8

It is provided by the func_channel module.

@Yash3120 Are you selectively loading modules? If so, then you are using functionality from modules you haven’t loaded (such as func_channel and format_sln).

9

thanks error gone yeah I load module manually

10

Yash3120 is not the author of the AGI, Lefteris Zafiris (zaf@fastmail.com) is.

11

Yes, you can see I given link of official GitHub link in above one the thread.

12

And sorry for my inconvenience I did not mention his full name, I just provide github link.

I apologies for that.

13

If you load it manually, it will ‘go away’ again when you restart Asterisk.

You should take a look at your ‘modules.conf’ file.

14

I note this, yeah okay I have enable at auto load so I don’t have to load everytime.

Thanks :+1:

15

That’s an interesting perspective.

An AGI library does obscure the foundations of the protocol:

  1. Asterisk ‘owns’ STDIN/STDOUT and anything you do to those streams is not going to end well.
  2. The protocol is ‘single line oriented, request and response’ meaning you issue a single line request, and you get a single line response (IIRC).
  3. You have to read the ‘AGI environment’ Asterisk gives you before you can do anything – even if you don’t need anything from it.

But using a library makes the feature available to a class of programmers who don’t wake up in the morning excited that they can start the day with a deep dive into the Asterisk source code.

I fall somewhere inbetween :slight_smile:

I know the protocol pretty well (at least I used to :slight_smile: ) and I use a library because it improves the quality of my code and reduces the quantity of code I have to write.

Typing:

client_id = agi_get_integer_variable("CLIENT-ID");

beats typing the approximately 100 lines of C code that implement it.

16

The no SLN format can be part of the problem