Questions tagged [asterisk]
Asterisk is a communications server software and an open source framework for building communications applications.
195
questions
3
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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....
- 43
0
votes
0
answers
17
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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 ...
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0
votes
1
answer
54
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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 ...
- 303
0
votes
1
answer
63
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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 ...
- 303
0
votes
1
answer
132
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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 ...
- 3
1
vote
0
answers
103
views
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>&...
- 306
0
votes
0
answers
45
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How to configure directory lists for fleet of SIP phones (Cisco 7960 and Aastra 480i) connected to Asterisk?
So I work for a medium sized company running a FreePBX/Asterisk PBX server on-prem with a fleet of mostly Cisco 7960 and Aastra 480i deskphones. Right now, our Asterisk server also runs a TFTP ...
0
votes
1
answer
320
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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 ...
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1
vote
1
answer
233
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
[...
- 119
0
votes
1
answer
382
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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. ...
0
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0
answers
107
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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 ...
0
votes
1
answer
232
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 ...
- 311
0
votes
1
answer
205
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 ...
- 101
0
votes
1
answer
472
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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 ...
- 103
0
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1
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357
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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 ...
- 347
0
votes
2
answers
278
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 ...
Gökhan AKSOY
1
vote
1
answer
41
views
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(...
- 306
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1
answer
28
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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/${...
- 306
0
votes
1
answer
204
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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 ...
- 306
0
votes
1
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100
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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 ...
- 306
0
votes
1
answer
571
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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
...
- 101
1
vote
1
answer
670
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 ...
- 159
0
votes
1
answer
154
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 ...
- 159
0
votes
1
answer
742
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 ...
- 101
0
votes
1
answer
647
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 ...
- 303
0
votes
1
answer
1k
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=...
0
votes
2
answers
999
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 ...
0
votes
1
answer
341
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 ...
- 121
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1
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121
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 ...
1
vote
1
answer
662
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 ...
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votes
1
answer
264
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 ...
- 281
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votes
1
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2k
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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 ...
1
vote
0
answers
383
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 ...
0
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0
answers
46
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 ...
- 281
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1
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754
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 ...
- 1,799
0
votes
3
answers
280
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....
- 63
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1
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435
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 ...
- 101
1
vote
0
answers
37
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 ...
- 111
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votes
1
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691
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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 ...
- 53
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1
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134
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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 ...
0
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1
answer
320
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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 ...
0
votes
1
answer
595
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 ...
- 111
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votes
1
answer
86
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 ...
- 103
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1
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563
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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 ...
- 3
0
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1
answer
2k
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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 ...
0
votes
1
answer
294
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 ...
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0
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1
answer
105
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 ...
- 1
1
vote
0
answers
3k
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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 ...
- 11
0
votes
1
answer
733
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. ...
2
votes
2
answers
1k
views
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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