2,758 questions
-1
votes
0
answers
20
views
How to fork real-time stereo audio? [closed]
Several articles mention software modules that coordinate the forking of real-time audio streams from Asterisk: MixMonitor, app_audiofork, etc.
QUESTION: is there a definitive way to fork real-time, ...
0
votes
0
answers
20
views
Asterisk Date Readback Issue
We are using asterisk say data service for readback of date. We have integrated asterisk with bpmn-js, and use context to execute commands for say date in case of readback.
Following is the code which ...
-1
votes
1
answer
98
views
Remove "Anonymous" text from From header - SIP invite
I need to remove "Anonymous" text from below From header in INVITE request. Done following configuration but no luck.
Used Asterisk 18.23.1 built by root @ pilrh-noc-dynamicconivr-app01 on a ...
2
votes
1
answer
373
views
AsterNET.ARI how to implement a simple call between internal numbers?
I need to call from number 401 to 402 using AsterNET.ARI Asterisk ARI (401 and 402 are two internal numbers that are connected to softphones)
my extensions.conf
exten => 1000,1, NoOp()
same => ...
0
votes
1
answer
296
views
Make multiple calls asterisk ari python
I wrote a script on python 3 that makes only one call, if the caller answered, then we play the melody and hang up. I use ari asterisk. How can this be done for only a few numbers?
#!/usr/bin/env ...
-2
votes
1
answer
67
views
Amı Originate Calls Diffrent Context From Sended Context
I'm using ami-io library with Node.js. I'm sending originate call with a context for redirect answered calls to ivr. There is 2 ivrs for diffrent purpose. Also I'm setting the lead_id to channel for ...
0
votes
2
answers
120
views
Problem Getting RECOG_INSTANCE() Value When Using MRCPRecog with Asterisk-Java AGI
I am running some operations over Asterisk-Java AGI (Asterisk Gateway Interface) using MRCPRecog. There is a problem I am facing with the results from MRCPRecog.
The result returned by MRCPRecog is as ...
0
votes
1
answer
120
views
How do i write a dial plan that plays a audio file takes user input and plays another audio file?
I am using Asterisk, Laravel and PAMI for this, and I
wanted make something that sends out a call that does the following:
Plays a recording:"press 1 or 2"
waits for user input
plays a ...
0
votes
3
answers
396
views
Using Cut with multiple delimeter
I have a small problem. I have a variableCTI_VARIABLES with this content:
cti_CallDirection=1@;cti_ContractNumber=00202417@;cti_TransferVDN=48195@;cti_IvrButton=1
Each different variable within this ...
0
votes
1
answer
207
views
asterisk dialplan, how to use includes in contexts right
In an asterisk dialplan, say one had:
[ext-miscdests]
include => ext-miscdests-custom
exten => 5,1,Noop(MiscDest: [some destination])
exten => 5,n,Goto(from-internal,[some number],1)
and ...
0
votes
2
answers
170
views
Cannot listen to an asterisk manager event : QueueMemberPause
I am using nest js and asterisk manager for a call-center/CRM software. What i need to do is find out if an extension is on a break by emmiting its status on real time.
this.ami.on('...
-1
votes
1
answer
577
views
Asterisk AMI call originate with python for a group of contact at a time
I want to generate call to a group of contacts(suppose i have 1000 contacts in a group) from database at a time. Here for example i have 30 concurrent channels, i want that the code should originate ...
-1
votes
1
answer
201
views
dialplan show the unicode char
I have s simple code:
exten => p111,1, NoOp(********** test unicode char ************);
same => n,Answer();
same => n,Set(myVariable="امین رستمی")
same => n,NoOp(******** #2 ${...
-1
votes
1
answer
229
views
Prepending a 1 to outbound Caller ID FreePBX
I use FreePBX as an SBC for an older switch we use. The older switch can only send 10 digits as outbound caller id, while our SIP provider requires 11 digits for STIR attestation.
I have tried adding ...
-1
votes
1
answer
194
views
Multiple call logs being stored from cdr event in Asterisk while transfering call
While transferring a call, the lifespan of a call throws 5 different cdr events. One from queue and other acts as an inbound call.
this.ami.on('cdr', async (evt) => {
if (evt.source == '//...
0
votes
1
answer
309
views
Transcribing a GSM audio file in real-time using Google Cloud Speech-to-Text fails during an active Asterisk recording
I'm working on a project where I want to transcribe audio from a GSM file in real-time as it's being recorded by Asterisk. While I can transcribe already existing files perfectly using Google Cloud's ...
0
votes
1
answer
216
views
Asterisk PAMI QueueStatus Action does not return the queue name, how can I get the queue name with asterisk ami action
The Asterisk PAMI QueueStatus Action returns QueueParams Event, that reutrns information about the queue including [Queue, Max, Strategy, Calls, HoldTime, etc..]. But it does not return the queue name/...
-1
votes
1
answer
229
views
Asterisk AGI fails to execute cURL
I'm using Bicom PBXware. I have complex cURL with headers to send using -H which means I can not use Asterisk CURL app. So I created bash script which works properly:
/opt/pbxware/pw/var/lib/...
0
votes
2
answers
354
views
Detecting when Asterisk rejects an incoming call via AMI
We have a number of Asterisk servers; occasionally, human error means someone messes up the dialplan and incoming calls are rejected:
NOTICE[27927][C-00000188]: chan_sip.c:26826 handle_request_invite:...
1
vote
1
answer
565
views
asterisk fastagi pyst2 python3
I'm making a fastagi app with python3 and pyst2 library. The code:
#!/usr/bin/python3
# -*- coding: utf-8 -*-
######################################################
import asterisk, sys
import ...
0
votes
1
answer
107
views
Asterisk strftime %B
I want to record calls with mixmimonitor and save it with this format of the filename - %d-%B-%Y-%H:%M:%S.wav
%B - is the full month name (July for example) - it works, but I get the name of the month ...
-1
votes
1
answer
177
views
sip.js connects to asterisk service, long connection problem
When I use sip.js 0.20.0 to connect to asterisk, when I leave the browser tab for a long time, I cannot receive incoming calls, but I can make calls. How can I keep connected?
I think it should be ...
0
votes
1
answer
298
views
asterisk mute all channels on a confbridge by single command
I am working on a solution with asterisk, that needs to enter all caller channels in a confbridge and then there is admin user that can mute and unmute any channel at runtime by it's choice.
All of ...
-1
votes
2
answers
226
views
Why is text2wav not working in phpagi demo
I'm just trying to run demo in PHPAGI folder available here: https://sourceforge.net/projects/phpagi/files/phpagi/2.20/
However, the text2wav is not working. When the script has to run text2wav (in ...
-1
votes
1
answer
111
views
Guava issue when starting Asterisk-Java
I'm trying to compile a project and it uses guava. The project is Asterisk-Java (https://github.com/asterisk-java/asterisk-java) and when I'm trying to run the .jar with the command java -cp asterisk-...
0
votes
2
answers
2k
views
Python asterisk ami
Im trying to implement a simple call generator using asterisk ami interface.
I used astersik-ami package, there was no issue connecting and sending command to asterisk but i have hard time to make the ...
0
votes
1
answer
297
views
Asterisk undefined symbol with new module
Having this module as below:
#include <time.h>
#include "asterisk/module.h"
#include "asterisk/logger.h"
#include "asterisk/res_resetqueuestats.h"
static unsigned ...
0
votes
1
answer
952
views
PJSIP How to forward all the SIP headers as received in incoming call
I am running an Asterisk 20.03 SIP server. I receive multiple X Headers in the incoming SIP INVITE, and I want to send all these SIP headers when I forward this INVITE to the connected SIP client.
I ...
-1
votes
1
answer
484
views
Creating your Context in extensions_custom.conf | Asterisk
Salam! I'm trying to write my own context, but as a result, the curl request is not sent. I add files to extensions_custom.conf
Tell me, please, what's the matter?
[from-internal-custom]
include => ...
-1
votes
1
answer
125
views
Outgoing SIP call fails
I am trying to initiate a call using Python through the AMI. Code I am using to initiate call:
import socket
ami_host = 'localhost'
ami_port = 5038
ami_username = 'user'
ami_password = 'pass'
...
-1
votes
2
answers
636
views
Asterisk AMI Originate - add hangup_handler for first leg
We use Asterisk AMI Originate for outbound calls (our software), so after successfully connecting to the first leg (external), we can use dialplan to Dial the second leg (internal) and handle the ...
-1
votes
1
answer
590
views
Extract data elements from CURL result in Asterisk 11 (IssabelPBX)
I am trying to consume data from an API to build an IVR in asterisk 11. The CURL call works fine and I get back a JSON array of the data. However, I want to extract the value from specific elements to ...
0
votes
1
answer
123
views
When handling a SIP request, how do I run a long-run running function before sending a response?
I am writing an Asterisk module. When a certain SIP request is received, I want to wait until a certain condition is triggered, and only after it finish (max 5 seconds), should Asterisk's core ...
0
votes
1
answer
390
views
SIP Sorcery (C#) : How the server can reject call & Receiver can detect that server has rejected a call
SIPUserAgent objSIPUserAgent = new SIPUserAgent();
WindowsAudioEndPoint objWindowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession objVoIPMediaSession = new ...
2
votes
1
answer
751
views
how to get variable value from asterisk with asternet
I use the asterNet for manage event IN asterisk.
I need to get value of variable (result) in c# from dialplan query.
exten => test,1, NoOp(************ test ****************);
same => n,Answer();...
1
vote
2
answers
1k
views
How can I get number of active calls from Asterisk Manager Interface Event
I've made connection to asterisk, and managed to get number of active and inactive peers from event PeerStatus, but now i need to get number of active calls and channels and display them. I've tried ...
0
votes
1
answer
385
views
wav file from fluent-ffmpeg not working for asterisk if returned as stream nodejs
I am using aws polly for text to speech and generating mp3 out of it, to get the certain wav as asterisk needs wav with specific paramters, I am using fluent-ffmpeg npm package.
I generated the file ...
0
votes
1
answer
328
views
How to add custom information to CEL records for a call?
With a CDR, is easy to add a custom field.
[exit]
exten = s,1,NoOp()
same => n,Set(CDR(my_custom_field)=${SOME_VARIABLE})
same => n,Dial(PJSIP/phone1&DAHDI/17,30,tTkK)
same => n,Hangup()
...
0
votes
2
answers
279
views
Asterisk, Goto+Hungup
There is an extension, for which I need a random sound file to be played, and once it's played the call should be hung up. I have the below dialplan which plays, for example, the 2nd playback and then ...
0
votes
1
answer
330
views
How to set dial options in Asterisk callfile
I'm already making calls from file, but I want to put TtM(MyMacro) dial options. The call file haves these properties:
$callFileOptions = "Channel: SIP/Algar_AMD/$phoneNumber
\...
0
votes
0
answers
485
views
Blind Call Transfer in Asterisk | ari-client (Node.js)
Precise overview about the flow:
I am calling my initial extension(100) using Zoiper through Twilio sip domain using Zoiper as soft phone which actually direct the call to my asterisk server.
Now when ...
0
votes
1
answer
710
views
Making a call from one extension(100) to another extension(200) through ari-client (Node.js)
Precise overview about the flow:
I am calling my initial extension(100) using Zoiper through Twilio sip domain which actually direct the call to my asterisk server.
Now when my call is in asterisk ...
0
votes
0
answers
205
views
Add PJSIP header on inbound channel
I am trying to add PJSIP header in asterisk in node.js but headers are not setting.
await new Promise((resolve, reject) => {
incoming.setChannelVar(
{variable: 'PJSIP_HEADER(add,X-...
0
votes
1
answer
900
views
run a macro on dial cmd Asterisk 18
i want run a macro on dial cmd,
this is my macro `
[macro-subrao]
exten => s,1,AMD()
exten => s,n,Set(RAO=${AMDSTATUS})
`
this is my dialplan
`
exten => _+33.,n,Set(FILENAME=${EXTEN}_${...
0
votes
0
answers
2k
views
How to stream back audio using External Media Channel using ARI in asterisk
Working with AsterNET.ARI I am able to expose audio stream from asterisk, using ExternalMedia.
Scenario:
Call is started and Stasis App is invoked
Create ExternalMediaChannel that sends stream to a ...
1
vote
1
answer
850
views
Asterisk ARI Caller id is always Anonymous
I'm trying to make a outgoing call using asterisk and stasis.
From my app, i send a simple post to /ari/channels/create, like this:
params = {
endpoint: `PJSIP/${trunkPrefix}${numberOriginal}@${...
0
votes
2
answers
412
views
variables in extensions_custom.conf
I am using Queues to handle incoming calls which works pretty well.
What I would like to do now is to get the info which extensions has answered a call in "sub-queue-answer" context of ...
0
votes
1
answer
714
views
Laravel - listening for specific Asterisk events using Marcelog Pami
I'm building application in Laravel 8. In application I need to receive event from Asterisk. To do that I downloaded Pami package.
In app\Listeners folder I created file AmiTestListener.php for ...
0
votes
1
answer
2k
views
Asterisk extension, conditional execution?
I have an this extension in file /etc/asterisk/extensions_custom.conf:
exten => _XXXX,1,NoOp("-- from internal custom --")
exten => _XXXX,n,Set(CURL_RESULT=${CURL(https://your....
-1
votes
2
answers
579
views
Asterisk Freepbx - hide external number on user's display after forwarding
For some reason I can not find the necessary information on the Internet.
Subscriber A has set forwarding to an external number (output to the city goes through a sip trunk). I want this external ...