Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

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Asterisk 13 “ __set_address_from_contact: Invalid contact uri” error

I'm installing Asterisk 13 on our production Ubuntu 14.0.4 box and the process seemed to go well. I previously installed it on my personal server as well. It works perfectly on my personal server, ...
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Asterisk : SIP/2.0 401 Unauthorized : same phone , different results

First of all I have already taken a look at similar questions such as : http://serverfault.com/questions/392979/asterisk-sip-2-0-401-unauthorized or Asterisk 401 Unauthorized when trying to register ...
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1k views

Unable to connect to Asterisk via Telnet

I'm trying to connect to asterisk via telnet. In order for PHPAGI to use certain commands, it has to be able to do the same. So I'm cutting out the middle man and testing it manually. I've ...
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2answers
951 views

FreePBX How to play an announcement for misdialled calls

Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately. Some of our users who ...
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27 views

Load too high in Virtual Machine [closed]

I have a problem with a virtual machine are configured in the same way, are asterisk switches that receive the same number of calls, and for some reason the load Constantly rising to reach 14.0 and ...
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1answer
70 views

Using Asterisk for IVR calls without internet connectivity

I am thinking about using Asterisk as an IVR for a project in Burundi, in a place where we often lose internet connection. I'd like the Asterisk server to deliver IVR calls through, say, an Android ...
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12 views

Asterisk Cancel Transfer (atxfer)

I'm using the Asterisk 13 AMI to start a atxfer. It works so far. But how can I cancel the transfer action? Example: Bob calls Alice. Then after some time talking, Alice starts an automated transfer ...
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58 views

Make Asterisk to serve another subnet

I have server with Linux Ubuntu 12.04. It has two subnets - 192.168.0.1(eth0) and 192.168.1.1 (eth1). There is an Asterisk server (IP PBX) connected to first subnet, lets say it has IP address ...
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1k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
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162 views

Connect Cisco 7975 VOIP phone to two separate Asterisk PBX's

I have a Cisco 7975 registered on an Asterisk SIP server, and I want to register a second line on a second (different IP) Asterisk SIP server. I tried to do this with firmware version 8.5.3 but it ...
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33 views

How to have dynamic extensions in Asterisk

I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this). With my old provider I could have inside my ...
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1k views

How can I limit Asterisk's Chanspy feature to certain channels?

I'm using a standard config of Asterisk 1.4 with FreePBX using the chanspy feature code. I want to restrict what channels it is possible to spy on, ideally doing this in a .conf file. How would I do ...
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92 views

How to match numbers based on end of number?

I am setting up an Asterisk VoIP server (using FreePBX) and I need to identify all numbers that end with a particular pattern. This is required for routing purposes, that is, to find out whether they ...
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1answer
59 views

Fail2ban critical exception - does not block ips

I'm trying to get fail2ban running on CentOS7 (no SELinux) and it uses firewalld. My goal is to set it to ban Asterisk password fails. Setup was default yum install fail2ban Config-wise I ONLY added ...
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27 views

How to know what number was dialled on incoming call

The scenario In the UK. I have one (BT openreach) phone line coming into a Cisco SPA232D. That phone line is ringable on a number, let's say 01235AAAAAA. I have a virtual number, 01235BBBBBB, which ...
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1answer
35 views

Dynamic Extension creation in Asterisk

I am looking for a way to configure asterisk to allow for dynamic creation of extensions whenever a register command is received. Put security aside for the moment as this is for a development ...
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83 views

Linux based asterisk server, SIP softphone able to hear audio but not send audio to others

Running Ubuntu 14.04 LTS and Asterisk 11.7.0. The server is set up for SIP phones to use our SIP trunk via voip.ms/didlogic (two different services because we like international rates on didlogic, but ...
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1k views

Asterisk: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)

Strange issue started with our Asterisk install yesterday. When dialling out via ISDN BRI the call fails and the following error is logged: Unable to create channel of type 'DAHDI' (cause 34 - ...
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5k views

Asterisk cached an old dial plan? rejected because extension not found in context?

I'm trying to fix my asterisk server which has been quite stable up until recently. Here's my problem: a) if i run a sip reload and/or dialplan reload from asterisk terminal, it sometimes breaks the ...
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35 views

Asterisk test ISDN card

I have an AVM ISDN controller C4 plugged into an asterisk machine. It shows up in the PCI device list as: Digital Equipment Corporation StrongARM DC21285 (rev 04) How can I test this card to see ...
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2answers
2k views

Asterisk IVR setup: disallow direct dial for some extensions?

We've got PBX to configure. It's based on Elastix 2.4.0, freePBX 2.8.1 and Asterisk 1.8.20.0. Currently we have a setup for inbound calls like that: Inbound call > Announcement (30 sec) > IVR ...
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2answers
10k views

Asterisk SIP digest authentication username mismatch

I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue ...
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20 views

Make an authenticated SIP call via URI

I'm trying to redirect a phone number to a SIP address that's only accessible when authenticated. Given the following, how would the SIP URI look like (if possible) ? SIP Server: sip.example.com Auth ...
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68 views

Asterisk call should be with peer, but gets trapped and stays local

I'm working with Asterisk and Vicidial, trying to place outbound calls through a SIP "trunking" provider. The calls never connect, instead I hear either demo-instruct.gsm or invalid.gsm playing from ...
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23 views

how do I route Asterisk dials from serverA to serverB?

I'm trying to make an Asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but am having a conceptual problem with ...
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64 views

The benefits of a proxy server before Asterisk

We are busy to develop an app to let users call eachother using webRTC. On this moment we have audio working, but video not. Now I hear that a proxy server can be the solution for this problem. But it ...
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31 views

Asterisk Auto dial out issues

I'm trying to make auto dial out in asterisk. I have created call file in /tmp directory and moved the file to var/spool/asterisk/outgoing/. It's not working. Please help me to resolve this issue. ...
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32 views

Asterisk one way audio detection

I have some calls have one way audio. I tried to write ${CHANNEL(rtpqos,audio,all) to CDR and think what if themssrc=0 than no audio, but it's not true. Is any way to detect one way audio? ...
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72 views

OpenLDAP & Asterisk - no global superior knowledge

I'm struggling to setup OpenLDAP to allow Asterisk to store information about users. I've tried to follow several howto's, but unfortunately I'm not able to set it up correctly. It seems like most ...
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1answer
45 views

Asterisk logging to Graylog2

Does anyone have any experience with logging Asterisk servers to Graylog2? I'd like to be able to send certain verbosity levels to Graylog2, but I don't have a clue how to go about it. The goal is ...
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47 views

Asterisk WaitForSilence NEVER detects silence

I am trying to use my dialplan to play recordings with WaitForSilence to make sure it wait until the person is done speaking or the message is left on voicemail. However, it doesn't seem to wait for 5 ...
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952 views

Limit simultaneous call per user in Asterisk

I want to limit simultaneous calls per extensions in Asterisk for security reasons. For example when a user is on the call no body else would be able to make a call by that extension. How can I ...
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63 views

Asterisk new PJSIP driver security option

I have a working asterisk environment, but I get a lot of unwanted traffic, like sip scanners of people who even try to call as a guest. I'm using res_pjsip, the configuration is stored in ...
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47 views

Asterisk realtime include dialplans

I noticed that when creating a dialplan in asterisk on realtime, I cannot inlcude contexts. So to be clear, I want to do the following: [context1] switch => Realtime [context2] include => ...
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27 views

Asterisk Dialplan Siptrunk dial user input

I have an SIP Trunk on my Asterisk box. So my Asterisk box has one public number and 100 internal numbers. external number: 0030XXXXXXXX internal numbers: 2000, 2001, 2002, 2003 ... and so on I ...
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13 views

Attaching Asterisk console to process

I want to open control console on currently running Asterisk instance. When am I supposed to use option -r and when -c in order to connect to Asterisk process? What is a main difference?
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1answer
197 views

Asterisk 911 dialing rule

Our company is looking to replace our outdated PBX with VoIP. One function we really like about our current PBX is that it alerts the secretary when a line dials 911 and it shows the extension where ...
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1answer
60 views

why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?

When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection: <-------------> [Mar 23 20:01:34] ...
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50 views

Asterisk using PORT NUMBER in FROM FIELD of SIP calls

I have an annoying situation with Asterisk 13. I setup a SIP trunk that works OK to receive calls. However, I can't make calls even though my SIP provider is working (I can place calls with a regular ...
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2answers
80 views

routing for iax-protocol does not work on pfsense

I have problems with a pfsense-box routing iax-traffic (UDP port 4569). Our company has a central office and eight branch-offices. At the branch-offices we use pfsense as firewall and VPN endpoints ...
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1answer
136 views

Asterisk - very high Load Average with no reason

There is a problem with Asterisk server showing high Load Average I can not explain. Ny setup is: Asterisk 13.1.0 Linux 3.13.0-24 (Ubuntu Server) Dual socket (Xeon E5-2620) server, HT enabled - 24 ...
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109 views

Asterisk Firewall access Rules

I have a questions regarding the firewall rules for Asterisk VoIP server. I have a restricted access to UDP/TCP port 5060, which seems to be blocking calls. I want to make sure that my VoIP Server ...
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1answer
104 views

FreePBX UK Dial Patterns Route Through Specific Trunk

I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly. I've got Freepbx installed and working properly - no issue there. I want to create a dial ...
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307 views

WebRTC on standalone asterisk - no audio

After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows: Im not getting audio from WebRTC to WebRTC clients. I work in a LAN ...
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1answer
138 views

Asterisk: execute shell command on server when call is accepted on SIP extension

I'm trying to configure asterisk to execute shell command for incoming calls - but only when the call is accepted. I've managed to setup extensions.conf so that command is executed when new call comes ...
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290 views

Can't dial through SIP trunk: FreePBX/Asterisk

I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration: type=friend host=201.217.31.10 callerid=mynumber ...
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932 views

Asterisk Intrusion Prevention

Let me start of by saying that I'm a noob, and what I've figured out so far has only been by stumbling my way through it. I have Googled around and the solution may be out there already, but it was ...
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155 views

What is the use of echo canceler in ISDN boards if all my endpoints have echo canceling?

Most digital telephony boards (BRI/PRI) have optional hardware echo canceler presented as alternative to the cpu intensive software echo canceling available in Asterisk and FreeSWITCH. I'm ...
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173 views

Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf file. ...
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113 views

Asterisk Sip Server and “Screen Sharing” function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...