Questions tagged [asterisk]

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

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Asterisk/FreePBX auto dial and change caller ID

I have the following file I copy into the auto dial folder: Channel: Local/19541111111@from-internal CallerID: "CallerId" <19543333333> Application: Playback WaitTime: ...
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How to create X number of endpoints?

In pjsip.conf I have each endpoint defined as follows: [my-endpoint](!) type=endpoint context=default disallow=all allow=ulaw transport=simpletrans [1234](my-endpoint) auth=auth1234 aors=1234 [...
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Asterisk ARI - ChannelDestroyed event not firing when using early bridging

I'm using Asterisk 18 with ARI's node library. This specific issue is quite simple: I aim to create a channel, add it to an existing bridge where some participants already are, then dial the channel. ...
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Call each available queue members just once and then leave the queue in Asterisk

I'm using Asterisk 13.30.0 in realtime. I have a queue and this queue's ring strategy is fewestcalls. I want each call in the queue call the available members in queue just once and then leave the ...
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how to make the script work for all files with similar names - vbScript

I am using a vbscript for moving files using a list.txt file. Script first ask me the file extension to be operated on by an input box. and the code for that is ' File Extension type strExt = InputBox(...
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ASterisk 18 - Webrtc etxension call has no audio or video after attending call , sip extemsion to another sip works fine

I have setup a asterisk on openwrt. everything works fine except calls between webrtc extension and sip extension. Asterisk version is 18 here is the output of httpstatus http.conf is as follows [...
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Asterisk: ignore call forwarding in queue

Scenario: there is a simple queue configured for 3 extensions (1, 2, 3). User of the phone 1 enabled call forwarding on his phone (let's say, to extension 10). Problem: if a call comes in, it directly ...
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How to setup Cisco 7945 with two lines from different Asterisk servers?

I have a phone line with 192.168.3.10 Asterisk IP Server, and another phone line with 192.168.3.11 Asterisk IP Server. Now I want to setup 900 from the first server and 500 from the second server on ...
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How to remove caller from asterisk queue when they disconnect?

We have our asterisk queue set up with the below configuration in queues.conf. It works great except for one thing. When a caller in the queue hangs up they are not removed but retain there place in ...
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Asterisk issue with IVR and '1' extension number

I have FreePBX 15 installation. I have successfully added multiple extensions. Numbers include some three digit numbers and '1' extension which is a normal ip-phone like others with exactly the same ...
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Asterisk realtime sip.conf deny/permit

I have a realtime sip.conf using deny/permit rules. For example: in sip.conf [MyPeer] type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ; in realtime i ...
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Asterisk, why my isdn adapter TA can receive but not make calls?

I have a b410p card ISDN BRI, configured with 4 NT ports. On one NT port(2) I connect a ISDN phone(is working), on the other(4) I connect a TA(usb hamlet) connected to a VM with Windows 2000 running(...
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Asterisk: strange thing happen with this dialplan

I have a ISDN phone connect to a Digium b410p card. Is working fine but there are some strange thing with the dialplan, with this simple dialplan [from-dahdi] exten => 1796522,1,Dial(DAHDI/g12/${...
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Asterisk and isdn phone: can "receive" calls, but outgoing calls are not seen by asterisk

I have configured a b410p isdn card with four NT ports. On the 4th NT port i put a TA which is on Windows os, and run an old isdn phone software(rvs phone). I have configured the extensions.conf file ...
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Asterisk and isdn bri card give me error 3302 isdn

I'm trying to configure a b410p card with asterisk. I have attached a phone isdn to the NT port of the card. This is my configuration /etc/dahdi/system.conf # Span 1: B4/0/1 "B4XXP (PCI) Card 0 ...
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Not hear audio from both remote extensions?

hello I have a Freepbx server as my telephone exchange, I am using a VPN with two Mikrotik devices to connect the extensions, all the remote extensions are registered correctly and have a ring, but I ...
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Asterisk - Video Recording on current version (18.6.0)

I’ve setup Asterisk to be able to accept inbound video connections and setup an extension to record as follows: pjsip.conf contains: [video-trunk] type = endpoint transport = transport-tls-nat ...
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Asterisk: how to get current sip.conf settings?

I'm not sure if sip general parameter (such as rtptimeout) is set. There is no mention of it in /etc/asterisk/sip.conf. But may be it is set somewhere else (in include files, for example). How do I ...
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Asterisk: how to end frozen calls

Sometimes my voip provider shuts down service without ending calls which are in progress (these calls are incoming to my server and they are coming to queue). So calls are never gracefully finished, I ...
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Asterisk fails to accept incoming calls

I have configured a local Asterisk server. There are local users and a single trunk for external calls. Users can successfully make local and outgoing calls. However when external calls come to the ...
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Outbound audio issues with Asterisk

I am running Asterisk 16.9.0 on a CentOS server at a client's location. That server connects via a SIP trunk provided by the telco. I configured Zoiper on a couple of PC's there and connecting using ...
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Incoming calls are rejected because there is no 's' extension

I've have configured VOIP provider and voip device. Sip config as follows sip.conf [operator] canreinvite=no context=from-operator host=operator_ip secret=operator_secret type=peer qualify=yes port=...
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Execute ssh via plink on Windows Host with sudo commands

I've Windows and I need access to Asterik (Linux server), via SSH, to execute one command. Now I open Putty, enter IP Server, Port, user and password SSH, execute the sudo command, enter password ...
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Asterisk: ODBC options not available in menuselect

I have done yum install unixODBC and yum install freetds, but when I try make menuselect while installing Asterisk, none of the ODBC options are available. func_odbc and res_odbc show up with an XXX ...
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How to establish connection between android phones using zoiper running Asterisk?

I have this scenario: 2 smartphones and 1 laptop, all in the same network. Both smartphones have Zoiper in order to comunicate with an asterisk server. The laptop is running Mac OS (Host) and inside ...
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Connect a music player in SIP voice call

I need to establish a normal Voip conversation between 2 persons and one of them, using a softphone in Windows, insert audio from a music player which can be controlled by the own user. The voip ...
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How to set the MixMonitor extension that match the RTP codec used?

I have a dialplan where I am using the application MixMonitor. This application take a file path, and the extension is used to define the format of the file (wav, alaw, ulaw, g729, ...). For the ...
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How to use Jitsi as a softphone SIP client?

Wikipedia lists Jitsi features as including: IPv6 support for SIP and XMPP But where and how are SIP credentials configured? So that I can troubleshoot an Alice and Bob SIP call: sip diagram Above is ...
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Asterisk outbound_registration real time issue

I configured an asterisk to run with realtime database. Every thing is working as expected, except the outbound_registration part, which was not running unless using pjsip.conf static file! From ...
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Is it possible to use signal level as signalisation for AEX2400 board in Asterisk?

I want to be able to use signal level on an analog line to be able to generate valid call via a AEX2400 with FXO modules. To understand better my question, lets say I have hooked up a smartphone to ...
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How to receive phone calls using a data only card without using third party online service?

I’m needing a 5G nsa card for my laptop no larger than 42mm×30mm. The only one I found is the MV31‑W from Gemalto (the manufacturer). But there are maybe other options. After a call to the ...
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Router won't forward ranges of ports to Linux server. Is there a software solution?

I need to forward a large range of ports to a Linux machine running Asterisk. The machine is behind a household router. The router can be successful configured to forward single ports to Asterisk, e.g....
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Where Is Asterisk FreePBX Video Call Recording File

I would like to know where can I find (if it does exist) the video file for a call with video. I already have all the things running properly: Call recording enabled for extensions as Force Call ...
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Setting up a computer as a phone station via my router's telephone ports [closed]

First off, I should note that this is my first time working with telephone hard-/software beyond simply using a phone, plugging it in or setting up a router. In addition, I'm aware I could just use ...
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How to make asterisk call file as user account or extension

I am trying to figure out why the callerid works when I call with twinkle through asterisk but not when making a call file. The callerid with the call file is always unknown. Twinkle account is setup ...
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Asterisk 13 & Pocketsphinx Segmentation Fault

I've built a voice recognition system for german digits using asterisk 13, pocketsphinx and a plugin and server (based on astsphinx) to connect asterisk with pocketsphinx. When testing the system with ...
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Asterisk configuration, need to use a different internip for each trunk

I'm currently configuring my asterisk server, that has two trunks, one for the incoming calls, one for the outgoing calls. For each trunk, I would need to define a specific internip. So far, the IP ...
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Asterisk: check original Caller ID on transfer

I am using Asterisk 16.5.0 with usecallmanager patch (which allow me to use Cisco phones features). We have a number of ConfBridge rooms, which are supposed to be called by employees only. Some ...
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How to omit a bad PSTN line in Asterisk

I have an Asterisk/Dahdi server that serves as a PSTN software PBX running on Debian with a Digium 1TE134F card. The setup has been up and running for sometime but we noticed that one of our lines is ...
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Asterisk, IAXModem & HylaFax: No carrier detected

I try to get a HylaFax + IAXModem + Asterisk Setup on a Debian 10 server running. The server is directly connected to the Internet, the firewall is disabled (no rules set up). The only thing that ...
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How to configure instant messaging in freepbx 14 & Asterisk 13

i have a trouble to configure instant messaging using freepbx 14 and asterisk 13 , i want that two sip clients can send and receive messages using their soft phones on smartphone and desktop , can ...
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Sip Trunk Peer issue with 2 asterisk servers

I am getting the unauthorized error while trying to place a call: [Sep 3 16:20:19] Asterisk 13.27.0-vici, Copyright (C) 1999 - 2014, Digium, Inc. and others. [Sep 3 16:20:19] Created by Mark ...
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Random hop in IVR

I have an Asterisk 13.23.1 server with freePBX 10.13.66-1. My problem is this, in one IVR when the caller puts a no valid destination (enter a number not configured in the IVR plan) it should go to ...
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Asterisk: chan_sip.c:4321 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

I have a setup of Asterisk PBX running on a Raspberry Pi 1 (IP 192.168.2.2) and it runs a few local IP phones just fine. I recently purchased a Grandstream HT813 gateway (IP 192.168.2.3) to connect ...
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DAHDI 3.0.0 not assign spans and cannot generate configuration

I'm setting up a new fresh installation of Debian 10, Asterisk 16 and DAHDI 3.0.0. I have specific the Digium TDM410P 4 port analog card (PSTN) and trying to configure it with asterisk and dahdi. ...
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Asterisk 16 LTS & PJSIP; hello world works but no sound coming from endpoints

I have recently set up an Asterisk server with version 16.4.0 LTS. I am completely new to asterisk yet I have managed to set up the server with the service and it runs smoothly among LAN users and it ...
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Supervisor don't see playback recording in freePBX UCP dashboard

My IT Head surely change the default location where call recordings are being saved, as a result Team Leads can no longer view playback button in UCP, they cannot do call listening. I believe that ...
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Asterisk 16.4 pjsip trunk registration

I'm not an asterisk expert, and I'm stuck at this moment. I'm trying to setup an asterisk box with realtime. Most work, and my endpoints are able to make calls between each other. But I need to setup ...
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Asterisk jitter buffer on receive side of B-Leg (device side)

I've been trying to apply a jitter buffer to the B-Leg or device side of calls for some time now without any success. I've set the jitter buffer to something very large just to test, using an echo ...
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Asterisk how to call by names instead of numbers

I haven't found anywhere clear sample how can I setup alphanumeric alias for internal numbers. For example I am calling now to 504@ip. How can I make it possible to reach 504 number by john@ip by ...
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