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 and SIP behind NAT

I am trying to Setup an Asterisk-Server to accept calls from a client in an other Network. The Server and the client are behind an NAT. I have already activated STUN on the client, but I am still ...
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14 views

Can't make Asterisk because pjsip.h is not found

I was following this tutorial to install Asterisk 13 on CentOS 7: http://jonathanmanning.com/2015/06/15/how-to-install-and-compile-asterisk-13-on-centos-7/ At one of the last steps, after I run "make"...
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Asterisk: How to disable specific dahdi channel without reloading module?

My POTS line #2 (which is channel 2-1 on Dahdi group 0) is having noise issues, while the phone service provider is fixing that problem I would like to disable this channel temporarily. Is there a ...
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elastix cdr stop working

CDR was working before 19 march. Unfortunately i dont remember what kind of changes i made to configuration, but this exactly not changes to CDR config. elastix 2.4.0 asterisk 11.7.0 mysql 5.0.95 ...
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Frequent brief CPU spikes on asterisk server

We have frequently where the load average on our asterisk server suddenly spikes (80 - 100 +) and the idle goes down to single digits. We have the problem only on the servers running CentOS 7. We ...
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1k 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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1k views

Call duration limit & least cost routing

We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little ...
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10k 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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Asterisk AGI randomly returning 4 and failing to execute PHP AGI

I've got a PHP script that I execute 2-3 times throughout a single calls life time, if it is inbound it will execute once for the queue, once for start of call and a last time for end of call. For ...
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2k views

How do I detect call forwarding in Asterisk?

Basically, I want to do the same thing that Google Voice does. I forward my calls to a DID number that rings my Asterisk server via IAX2, which, if it detects the call has been forwarded, sends it to ...
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1k views

Asterisk / Elastix Address Book and Call Recording

I am trying to build a small asterisk based PBX using Elastix. It'll be having 4 FXO (2 nos.no ISDN, normal analog POTS, 2 nos. GSM connections using a GSM Terminal) and 4 FXS (2 IP Phones and 2 ...
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Outbound Rule not working AsteriskNow

I am trying to set up outbound routes for UK numbers (both landline and mobile). UK numbers follow this convention: 0[12378]000000000 However, I cannot make outbound calls on my AteriskNow server. I ...
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3k 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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19 views

Call from '201' to extension '202' rejected because extension not found in context 'test'

I have this code on my extensions.conf [test] exten => 20,1,Answer() exten => 20,n,Playback(hello-world) exten => 20,n,Hangup() and this is my sip.conf [201] user=201 secret=123456 type=...
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20 views

Asterisk automatic installation

I recently started a project which I need to automate the installation of Asterisk servers for my customers in AWS infrastructure. Customers already own AWS infrastructure. I was thinking the ...
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25 views

ISDN BRI vs SIP Trunks over EDIA

I am installing Asterisk in a small office with just 4 lines and 8 Extensions. I have two choices from my local telco (Fairpoint): 1) Old School ISDN BRI lines which I would connect to Asterisk ...
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24 views

Asterisk's “dialplan reload” and “help” commands are not found

I've installed Asterisk 13(current) on CentOS7(without DAHDI/LIBPRI). It seems to work. I was following instructional videos, but at some point these use dialplan reload and help commands. These ...
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53 views

Four node asterisk cluster

I have a question regarding Asterisk servers. I have a client that have two locations connected with fiber optic. On both locations they have PRI ISDN-s. On one location there are 4 PRI ISDN, and on ...
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313 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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68 views

send fax with Elastix and Hylafax

I need help getting around "Unknown problem" while I am trying to send faxes using Hylafax and Elastix. I have an elastix 4.0 inside a VM, calling outside and inside the box is working great, using ...
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33 views

Asterisk: How to enable the HTTPS mini server?

I am attempting to enable the HTTPS mini server in Asterisk with little luck. Here is my http.conf: [general] enabled=yes bindaddr=192.168.254.26 bindport=8088 tlsenable=yes tlsbindaddr=192.168.254....
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41 views

Asterisk/FreePBX: how to reject all incoming calls after few seconds?

I have 3 dumb GSM relays that control the gates. If a person whose number is in device's address book calls, gate opens. Call gets rejected (in my country calls that didn't connect or lasted less than ...
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504 views

Asterisk congestion reasons

After moving Asterisk from one server to another, I noticed that ASR dropped by 15-20 %. Also I noticed following warning in logs file [Dec 8 15:52:00] WARNING[4771][C-00000031] pbx.c: Maximum ...
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20 views

Redirect Call Back Out To PSTN

I'm wondering if it's possible on a VoIP system (specifically Asterisk) over SIP trunks to redirect a call back out to the PSTN to no longer be routed through the PBX. What I'd like to do is see ...
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51 views

Phone won't register with Asterisk server

I'm trying to register a Grandstream phone on our asterisk server. We've done this tens of times, have documented how to do this, and recently it doesn't work anymore... I've tried with several phones ...
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65 views

SELinux preventing Asterisk from connecting to IMAP server for voicemail

I am running Asterisk on Fedora 23 and am using the voicemail IMAP module to store voicemails to Gmail. When SELinux is in enforcing mode it prevents Asterisk from connecting to Gmail. Here is the ...
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74 views

Phone rings, but no call, and no log entries in Asterisk

We have one phone (123) that rings from time to time, displaying what seems like an internal number (6001), but it's not, because we don't use that number and nothing in that range. When you pick up ...
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20 views

Adding command to elastix dialplan

TlDr: Is it possible to add a function (forkCDR) before Elastix Dial() an extension? We had a working asterisk, but due to the administration is now transferred to a person with lesser competence, we ...
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35 views

Dependency issue installing Asterisk 1.8 on Amazon Linux Server

I'm trying to install Asterisk 1.8 on an Amazon Linux Server (2015.09). sudo yum install asterisk This results in the following errors: Error: Package: libresample-0.1.3-12.el6.x86_64 (epel) ...
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31 views

Whitelist phones for DISA in asterisk

What would be the best practice to allow only a whitelisted list of numbers to access a DISA system in asterisk? I have a public number that lot of people use but if a certain number calls, it allow ...
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64 views

Asterisk - Send and and recieve SMS to mobile phones

I have an Asterisk that has several employees connected to it. each person has their own DID and they can make/receive calls to/from PSTN - No problem there I am looking for a viable option, to ...
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37 views

Change FreePBX Queue settings programatically?

Is there is anyway to change the settings(such as 'timeout') of a queue added by FreePBX?. I found No such options in Asterisk Manager (AMI) and CLI.
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3k 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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7k views

Skype Connect as SIP/Trunk for Asterisk

First off: I'm not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an ...
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547 views

SIP.js connection to Asterisk 11.20 over WSS not working

I have successfully setup sip.js using a standard non secure ws:// to an asterisk 11 server using firefox 43. I can make and receive calls to another ff browser/hardphone. But this does not work with ...
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226 views

Asterisk internal call not routing correctly. Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' for endpoint 'anonymous'

I am trying to figure out why my FXO adapter has suddenly stopped working, it has been a while since it was first configured, I had only changed an internal call timeout setting on the FXO adapter and ...
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227 views

Elastix 2.5 - Change settings via WEB GUI does NOT take effect

My PBX system is currently running Elastix 2.5, which has been installed & configured about three months ago. For maintaining purpose, I have used etckeeper to tracking changes in /etc folder, ...
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40 views

Asterisk One Way Audio or No Inbound Calls

I have an Elastix server running in Amazon Web services, but I can't seem to resolve an issue that is causing either one way audio or no inbound calls to ring. If I set my SIP extensions to "Nat = ...
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23 views

Postfix not attaching file from script

I'm trying to send an asterisk voice recording out via email using the script found here but instead of sending the file, it just puts the path to the file at the top of the email sent. I believe all ...
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11k views

Asterisk SIP/2.0 401 Unauthorized

I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well. The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of ...
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33 views

Asterisk support for rfc 3327 & rfc 6140

Does asterisk support rfc 3327 and rfc 6140 if so which version of asterisk supports these rfc's. Is there any open source sip server registrar that supports the above mentioned rfc's
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169 views

Asterisk hacking defence: “Call setup” in tcpdump from denied IP number

I am running Asterisk 11.7 on Ubuntu for my small business. These last days someone could access the server and make all phone extensions ring in regular intervals. I would like to understand how that ...
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37 views

Can't route traffic from Cisco to Asterisk over port 5060

My VOiP provider has informed me that I must route incoming calls from my Cisco router to my Asterisk server to port 5060. I can ping my Asterisk server from my Cisco router fine, there's a direct ...
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48 views

Asterisk server with two connections [closed]

I need that configuartion It works, but with only one connection, with eth0 Or with only mobile broadband I searched for it, but found only forwardind/routing solutions. How i can see asterisk ...
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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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SIPML5 connection to Asterisk 13 over wss

I have successfully setup sipml5 using a standard non secure ws:// to an asterisk 13 server, can make and receive calls using demo at https://www.doubango.org/sipml5/call.htm. However, I am having ...
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43 views

How to connect extensions of remote VOIP over remote STUN server?

Image of Architecture I am trying to setup I have configured the Asterisk server in a docker image and it's running, I have also configured the STUN/TURN server and it's also running on the same ...
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26 views

OPENVPN ipp.txt file assigining duplicate ip addresses

I am having an issue with duplicate entries for hosts in the ipp.txt file where the same host CN has been added multiple times with different ip addresses. For some or other reason some of my openvpn ...
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Having trouble with one way adio on an asterisk sip phone

I have an asterisk server (Elastix on CentOS 7) that I am currently running in Amazon Web Services. The server works great and clients can connect without issue using a stun server. However I have ...
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80 views

DTMF Tone issue with Elastix

I am very new to Elastix and have just gone through the setup of a new system on Elastix 2.5.0-8, I have Elastix in the middle of 2 sip trunks, one is the service provider and the other is a 3rd Party ...