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Questions tagged [asterisk]

Asterisk is a communications server software and an open source framework for building communications applications.

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How to connect Asterisk with SIP service [closed]

I am new to the SIP service with Asterik, can you please help me to understand how the SIP service will be connected to the Asterisk server. I used to work only with PRI service and Asterisk server. I ...
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11 views

Zoiper SIP to Digital Ocean Asterisk: Nothing happens on any test call

So i have a Zoiper app on cell phone and I'm trying to make a test call on a new Asterisk server with Digital Ocean. It connects and registers just fine, and seems to at least acknowledge I'm trying ...
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39 views

Is it possible to decrease the transmit/receive volume in Asterisk below normal?

I am playing around with changing the volume in Asterisk. There are a few applications where I would benefit from this. Most of them are to increase the volume but there are also some scenarios in ...
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1answer
94 views

asterisk: error while loading shared libraries: libasteriskssl.so.1

I am using Ubuntu 14.04 and followed the instructions to install asterisk from this link Ubuntu 14.04 Asterisk server installation All worked well until I run asterisk and get a shared lib error. ...
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42 views

Asterisk sip.conf in MySQL database

I'm able to include a phone inside a database for realtime usage. So, this code (from /etc/asterisk/sip.conf): [phone] type=friend username=phone secret=12345 host=dynamic disallow=all allow=g729 ...
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107 views

How to setup Asterisk NAT on docker for Windows

I need to setup a docker container running an Asterisk PBX. The problem is that I receive the wrong ip in sdp replies. In fact Asterisk is using the container internal IP. The second problem is that ...
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60 views

Asterisk not adding CDR record in DB for call to unregistered peer

I configured Asterisk CDR backend to PostgreSQL, and everything is working fine for calls between two registered peers. The problem is when a user calls an unregistered peer. I want Asterisk to ...
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66 views

On a PBX (Asterisk) determine which phones are connected

I am using Asterisk as my PBX. On my sip.conf I used to have this option enabled: [general] ... qualify=yes more info about qualify can be found in HERE. Thanks to that option I was able to tell if ...
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49 views

Asterisk CDR events

I'm getting asterisk cdr events from smal nodejs app. Calls are made using originate. Through one call, asterisk could check several communication provider, and after last i am getting this: [2018-09-...
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2answers
122 views

Converting a 2 channel stereo g729 file to individual single channel pcm encoded file

Whenever I try to convert a G729 encoded two channel file into mono channel pcm encoded wav files I get the following error Input #0, wav, from 'g729stereo.wav': Duration: 00:00:07.07, bitrate: 16 ...
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2answers
236 views

Asterisk NOT FOUND answer on OPTION request

Remote server send me OPTION package, but my asterisk server send "404 NOT FOUND" response. Is there an option in sip.conf or something else? sip.conf [remote-server] type=friend host=10....
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39 views

How to playback an mp3 on all active Asterisk calls?

I have a Asterisk PBX and a SIP trunk connected to it. Due to some finding with my SIP trunk provider, I have to change the SIP trunk credential on nightly bases. However, I do not want to simply ...
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1answer
302 views

Asterisk Trying to play back audio file “File /path/to/file.wav does not exist in any format”

i'm trying to rig an application to send out premade calls. I'm trying to get asterisk to play back wav/mp3/gsm files (I compiled it with the proper extensions and they're loaded) but when ever I ...
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0answers
69 views

Setting CallerID on outgoing ISDN calls? [closed]

Does anybody have any experience of settings the CallerID on outgoing ISDN calls? We are trying to set the caller id to one of the available DDI's on our circuit but without success. They worked on ...
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35 views

Understanding Yealink VOIP phone configuration

I know how to connect my VOIP phone to asterisk. My asterisk server sometimes goes down so I will like to understand better how can I use a failover server. This guide: http://download.support....
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109 views

How to add new CID Superfecta source in FreePBX?

I don't find information about how to add a new CID Superfecta source (my own custom programmed one) in FreePBX. Documentation doesn't say and there is no obvious method to do it in the FreePBX ...
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424 views

How do you fix “No Trust List Installed” error on a Cisco 7942G VoIP Phone?

I have a 7942G. I'm trying to connect it to Asterisk. Looking at Settings -> Status -> Status Messages, I see this: Error Updating Locale SEP############.cnf.xml No Trust List Installed I'm only ...
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1answer
73 views

How do I make music interactive using Asterisk?

I recently started my own computer repair business & use FreePBX/Asterisk for my phone system. Just wondering if it's possible to make my music on hold interactive. I would like to let callers ...
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1answer
128 views

how can two sip user talk to each other by asterisk

I am using asterisk 13 on centos 6 , now I have add two users in pjsip.conf as below [transport-udp] type=transport protocol=udp ;udp,tcp,tls,ws,wss bind=0.0.0.0 [6001] type=endpoint transport=...
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1answer
137 views

How to retrieve and save Asterisk phonebook entries via ARI?

How can I retrieve/update/add Asterisk phonebook entries via ARI? I'm looking at ARI documentation https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ARI, but I don't find anything about the ...
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1answer
157 views

Recipe for target ‘asterisk’ failed

Ubuntu14 I am trying to configure asterisk in my PC. I have downloaded the file asterisk-15-current.tar.gz unzipped and started installing. Ran ./configure and it was successful. on Make command, ...
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1answer
334 views

how to integrate mobile call to a chatbot using asterisk

I am developing a voice based chatbot which can call a person and have the conversation. Currently, the chatbot uses my pc mic and speaker for input and output audio. I am trying to integrate this ...
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135 views

Disable call recording/spychan after attended transfer in Asterisk

I'd like to know how to disable call recording or spychan after an attended transfer. I'm not using the attended transfer feature. Here's the context: An incoming call is waiting in a queue. When a ...
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1answer
188 views

Asterisk securing passwords

I am running asterisk on the raspberry pi 3; therefore, I want to protect the passwords. Someone can remove the SD card and the passwords will be in plain text! I know I can encrypt the entire OS but ...
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222 views

Minimal Asterisk installation on Debian

I'm trying to install Asterisk using Debian 9's package on my home-server (argh... unprofessional user, I'm sorry). But well, the dependencies are really astonishing, asterisk-modules needs a lot of ...
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28 views

2 of my softphones cannot register to Asterisk

I just installed Asterisk 15.2.2 on a CentOs 7 VM. I installed Ekiga and I'm trying to connect Albert with registrar localhost:5065. I have a second VM (Ubuntu) with Ekiga too. I'm trying to connect ...
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1answer
349 views

Asterisk call through browser

Currently, we are using two version of IPPBX (VOIP) System in our office, which are CUCM v11 and Briker 1.2. We want to call directly through web browser instead of using a soft phone (or any desktop ...
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1answer
168 views

Internal calls issues asterisk vicidial

I have just installed new Vicidial v7.0.3 on an i7 12 GB Ram pc I have completed all steps in the installation including OS-install, network configuration, Firewall set to internal on one of the ...
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2answers
376 views

How do I send Asterisk recording emails with a post-call script executed from FreePBX?

The script I have designated FreePBX to call, when called directly from the CLI sends the email flawlessly. The emails are not triggered after a call takes place, however. Background context: https://...
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1answer
289 views

twinkle configuration for Sangoma

I'm able to run AsteriskNOW virtualized on my local system. Also, I'm using Anveo on two different SIP hard phones. Anveo works fine, indicating that the network handles SIP fine. Now, how do I ...
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1answer
1k views

Asterisk initiate call through sip notify message

I have a Yealink ip phone on my desk and I want to remote control it so that I can place the calls from my computer instead than by typing the numbers directly on the phone. This link explains how to ...
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1answer
50 views

Pre-Bridge Handlers Asterisk playback issue

Hi i am using pre bridge handler U to play an audio when call is answered by the called party however caller is also able to hear the playback is there anyway possible so that only called party can ...
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1answer
1k views

Asterisk record all calls even when transferred

I do not know how to record ALL calls. I am having trouble when transferring the calls. Anyways this is my configuration: sip.conf: [general] tcpenable=yes udpenable=yes ... etc [peer1] ...
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1answer
39 views

IPSec doesn't hide IAX2 informations

I am trying to setup a secure truncked IAX2 connexion with IPSec between routers: But when I make sniffing attack with wireshark, all call's informations are showen ! IPsec is well configured and ...
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1answer
513 views

Asterisk exit queue and reenter queue without losing position

I want to give my customers an option to press 1 if they wish to listen to no music on hold. Sometimes is annoying to keep listening to the same song lol. Anywyas this is what I have: queues.conf: [...
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1answer
939 views

Asterisk/FreePBX behind pfSense – no audio in/out

I have following setup: ISP modem in bridge mode -> pfSense firewall -> HP2920 switch -> asterisk | VoIP phones I finally got inbound and outbound calls working but I hear no audio in/out. ...
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1answer
425 views

Asterisk (VOIP) port forwarding on TP-LINK

I have local Asterisk server on Ubuntu 16.04. Its address is 192.168.0.101:5060 and I would like to be allowed to talk with other users when I am outside local network. I tried this but it doesn't ...
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1answer
205 views

(ringinuse disabled) (Invalid) has taken no calls yet

I had a freepbx server in place. I setup a queue with static agents and ring all strategy. Now, suddenly no body in the queue will ring once the call routed to the queue. I check the queue and found ...
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3answers
239 views

What is the difference between cp -R foo/ and cp -R foo/*

I noticed that whenever I use a command cp -R foo/* I copy everything except files like .env or probably .gitignore The solution is to use cp -R foo/ as it copes the whole directory, but this does ...
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1answer
248 views

I am having error in Asterisk

I am working with Asterisk 13 on OpenWRT(Linux distro) and having following error. -- SIP/sipphone-00000000 is making progress passing it to DAHDI/1-1 [Aug 7 22:40:44] ERROR[13139][C-00000000]: ...
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1answer
88 views

Keeps detecting SELinux as enabled even though I disabled it (When trying to install FreePBX)

I was following this guide to install FreePBX on CentOS 6.9 (minimaslist), everything worked, until the last step to install FreePBX: ./install -n from /usr/src/freepbx, I keep getting the follwing ...
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236 views

How to fix Asterisk directory path

I have asterisk 11 installed on Centos. Everything was working fine till yesterday. I was installing mysql and using for connection. I have it working and my cdr are now nicely posted into mysql db. ...
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741 views

Can't hear audio in incoming chan_mobile calls

I have set up RasPBX on a Raspberry Pi 1 using chan_mobile to forward calls to and from a mobile phone to a SIP client using the both the instructions for plain Asterisk here and via FreePBX here. In ...
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1answer
130 views

Default port for calls with Asterisk

I have AsteriskNOW 11.9.0 with FreePBX 2.11.0.43. I moved Asterisk to a non-standard port number, but now I can call extensions only if I specify the port number explicitly in a callee's number, for ...
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1answer
410 views

Asterisk ERROR: stasis_app_exec: Stasis app 'channel-dump' not registered

I'm testing Asterisk ARI and got this dialplan on extension 1000 exten => 1000,1,NoOp() same => n,Answer() same => n,Stasis(channel-dump) same => n,Wait(100) same => ...
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1answer
251 views

Connecting Teamspeak to Asterisk conference

I'm trying to connect a TeamSpeak 3 channel to an Asterisk conference (using ConfBridge) on FreePBX. Not much out there about how to do it. Both the PBX and TS3 servers are running on the same CentOS ...
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1answer
132 views

Config to pass to Asterisk so it rings all extensions starting from “4” when a certain number is dialed

Which magic configuration line in which file should I put so that when a certain number is dialed, all extensions that match some wildcard ring, and a call is transferred to the phone which picks up ...
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1answer
164 views

Connect two analog modems across gsm carrier

I have a cisco router, and I need an emergency backdoor to it from a remote site in case the internet connection is down (for any reason). I've connected a good old analog modem to cisco console port ...
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0answers
83 views

Centos Server with Multiple NICS Static ROUTS

I have a centos server running asterisk with multiple nic cards. i have eth0 setup as my main nic connected to my LAN and eth4 connected to my sip trunk provider my sip provider requires my to log ...
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2answers
86 views

VOIP virtual numbers

I recently purchased a virtual number which is similiar to local landline number provided by my local provider. how voip provider can provide virtual numbers over VOIP? Do they have contract with ...