Questions tagged [asterisk]

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

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Connect Asterisk to Exchange shared mailbox via IMAP

We have been tasked with connecting Asterisk VM to our exchange environment so that Asterisk can drop a .wav file of the received v/m into a shared mailbox. We have been given the settings needed by ...
NullCool's user avatar
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2 answers
33 views

What command to analyze the traffic in Freeswitch

What is the command in freeswitch to analyse the trafic, our vendor don't want to inform what command is it, they share the screenshot like below:
DevDon's user avatar
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Powershell - Get-ChildItem asterisk wildcard in path - strange behaviour when used parameter -File

I found one strange behaviour when I'm using Get-ChildItem cmdlet Folder structure: d:\test\fooA\foo\1-9.txt files d:\test\fooB\foo\1-9.txt files d:\test\fooC\foo\1-9.txt files d:\test\fooD\foo\1-9....
Jan Krejci's user avatar
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24 views

Zanzibar Open Ivr doest not accept sip invite?

I'm sending a bottle at the sea with this question because this project is very old and no one use this now. So if noone can answear, I'll delete this post. I'm trying to integrated asterisk with ...
Florian's user avatar
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1 answer
241 views

Asterisk not playing Audio in BackGround or MP3Player APP

Hi Everyone in Asterisk 18 Cooked on a Centos 7 Server when I want to play a wav file, mono pcm s16 le 8000 hz lets called MyWavFile.wav I try to play it ether way whit application BackGround or ...
denn0n's user avatar
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1 answer
107 views

Can Asterisk use GSM PCIe card for Dial out or Dial in?

I want to know if it is possible to use asterisk to make calls through a PCIe Mobile Network card or a USB dongle I'm Looking it's to use a GSM line dial out and dial in and only requires one GSM ...
denn0n's user avatar
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1 answer
190 views

Asterisk confbridge, timeout only while waiting marked

I have just configured an asterisk server for my home doorbell. The goal is to allow my doorbell to join a conference room but wait for the first user to also join in order to start talking. I managed ...
Suxsem's user avatar
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About asterisk dialplan for external number

Suppose this simple dialplan [office-phones] exten => 1001,1,Dial(PJSIP/alice) exten => 1002,1,Dial(PJSIP/michael) exten => _XXXXX.,1,Set(CALLERID(all)="Name Surname <123456789>&...
elbarna's user avatar
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1 answer
552 views

Jitsi as a VoIP SIP softphone with asterisk - calls don't work

I have an asterisk server and I am trying to use Jitsi Desktop on Debian 11 as a SIP softphone (like Jami and Linphone) to connect to it and make calls. Jitsi can register and connect to the asterisk ...
Mos's user avatar
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2 answers
378 views

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 [...
Woodsy's user avatar
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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. ...
Mahmood Yousef's user avatar
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0 answers
135 views

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 ...
MehmetHikmet's user avatar
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1 answer
276 views

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 ...
o.v's user avatar
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1 answer
325 views

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 ...
amir22's user avatar
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1 answer
601 views

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 ...
gatorreina's user avatar
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1 answer
404 views

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 ...
Disa's user avatar
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2 answers
346 views

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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1 vote
1 answer
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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(...
elbarna's user avatar
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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/${...
elbarna's user avatar
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1 answer
220 views

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 ...
elbarna's user avatar
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0 votes
1 answer
114 views

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 ...
elbarna's user avatar
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0 votes
1 answer
741 views

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 ...
Alcanzar's user avatar
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1 vote
1 answer
977 views

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 ...
Roman's user avatar
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1 answer
203 views

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 ...
Roman's user avatar
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1 answer
991 views

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 ...
Eugene's user avatar
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0 votes
1 answer
829 views

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 ...
Hussain Akbar's user avatar
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1 answer
2k views

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=...
user9066046's user avatar
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2 answers
1k views

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 ...
Matias Wajnman's user avatar
0 votes
1 answer
399 views

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 ...
Matt's user avatar
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0 votes
1 answer
147 views

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 ...
Francisco Monteiro's user avatar
1 vote
1 answer
800 views

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 ...
Joel Guzmán's user avatar
0 votes
1 answer
291 views

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 ...
iXô's user avatar
  • 301
0 votes
1 answer
3k views

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 ...
Nicholas Saunders's user avatar
1 vote
0 answers
438 views

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 ...
Mohamed Lotfy's user avatar
0 votes
0 answers
49 views

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 ...
iXô's user avatar
  • 301
-1 votes
1 answer
958 views

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 ...
user2284570's user avatar
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0 votes
3 answers
385 views

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....
sebinho's user avatar
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0 votes
1 answer
658 views

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 ...
Andrew Paes's user avatar
1 vote
0 answers
42 views

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 ...
PixelMaster's user avatar
0 votes
1 answer
784 views

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 ...
Petepf's user avatar
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0 votes
1 answer
144 views

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 ...
christian's user avatar
0 votes
1 answer
376 views

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 ...
Arthur Deslauriers's user avatar
0 votes
1 answer
656 views

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 ...
kab00m's user avatar
  • 111
0 votes
1 answer
98 views

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 ...
gatorreina's user avatar
0 votes
1 answer
622 views

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 ...
nd_'s user avatar
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0 votes
1 answer
2k views

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 ...
Ranoo Hashem's user avatar
0 votes
1 answer
338 views

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 ...
Zingo's user avatar
  • 1
0 votes
1 answer
121 views

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 ...
joaquin's user avatar
1 vote
0 answers
4k views

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 ...
somebody0's user avatar
0 votes
1 answer
854 views

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. ...
Geørge Tømzaridis's user avatar