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

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Grouping extensions in Asterisk

We have an asterisk based phone system with multiple clients connected to it. At the moment, all the extension numbers fall in the range 100 - 9999. But, now we have an issue where a particular ...
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3answers
975 views

Convert a WAV (Asterisk recordings) to MP3

I have a large collection of WAVs. They are: RIFF, little-endian WAVE audio GSM 6.10 Mono 8000Hz effective bitrate of 13.6 I need to convert these to MP3. Current I am forced to send the file ...
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1answer
161 views

How to add a user who is out of state as if they were just another extension AT our office? [closed]

We want to add a person who is in another state to our phone system as though they were simply in another cubicle here at the office. So, they can dial "302" and get the office assistant's extension ...
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1answer
382 views

Asterisk IVR to Extension slow

I have a IVR set to pickup our PSTN lines and tell the user to enter the extension OR to press 1 for a representative. If the person enters an extension it transfers within 1 second , but if they ...
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1answer
79 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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555 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
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1answer
249 views

Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...
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6k views

Transfering call asterisk to different context

I have a Small and basic PBX, and with two contexts wich basicly are sales and supervisor both have different roles and privileges. I notice that it is possible to transfer call from the same context ...
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2k views

It is possible to set up an Asterisk PBX running on EC2 and have reliable call quality?

There is an EC2 kernel with a 1000Hz clock, which I have successfully used to set up Asterisk etc already, but I am wondering if other issues (e.g. transit from London to Dublin to London- our current ...
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1answer
2k views

How can I link two Trixbox servers to make free calls between them?

We have a Trixbox server at the office, and I have one at my home too. They have been working fine independently of each other for a while, but I figured if they could talk to each other over the ...
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262 views

Asterisk server for unlimited cell phone calling?

Does anyone know if Asterisk server can do the following? I have 10 freelance programmers each working from a different city, each with their own cell phone. I want to have one business phone number ...
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115 views

Recommended FXO Gateway for UK analogue lines

Can anybody recommend any FXO gateway devices to connect analogue telephone lines to an Asterisk VoIP system. Requirements: Minimum of 4 ports. Enterprise grade - quality is more important than ...
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621 views

Is a short delay before I can hear the person I'm calling a “known issue” with Asterisk?

Whenever I call someone using my workplace's Asterisk VOIP server there's a 2-3 second delay before I can hear them so I always miss them saying "hello". This happens with any SIP client running on ...
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1answer
3k views

Can I monitor Asterisk usage with Cacti?

I want to be able to see how many minutes of use each user in my asterisk uses. Right now, I am using a POTS interface to the outside wold, but I want to know how much the system is used, so I can ...
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1answer
207 views

Asterisk - Trigger when an extension becomes reachable/unreachable

We have people in our office that use softphones to connect to our Asterisk system. The softphones are online when they are in the office, and offline when they are not. So the Reachable/UNREACHABLE ...
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1answer
5k views

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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1answer
370 views

Sending faxes from Asterisk (email to fax) — is any feedback possible?

I am about to make some email to fax gateway --- some python program will make Asterisk call files and put them into some queue (files or db based). Then another program should take "call tasks" from ...
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1k views

Outgoing calls from vtiger to an asterisk extension

I've successfully installed vtiger and asterisk on Ubuntu, and I managed to configure them such that I get an incoming and an outgoing popup in vtiger. The problem is that when I click on an extension ...
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688 views

IM and File Transfers Integrated With VoIP

I have an Asterisk box serving an office of people. I'd like to provide instant messaging and file sharing capabilities alongside the voice and video capabilities provided by Asterisk (ala Windows ...
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407 views

Share phones but not extensions in Asterisk

I'm setting up a new Asterisk PBX, and I'm on the process of buying IP phones. I plan to add phones as needed, as we can't afford to buy them all at once. However, I do want to create a single ...
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2answers
878 views

Sound disappeared on IVR in Asterisk

I'm setting up an IVR in asterisk. It was working fine but I installed some new hardware and the sound has disappeared. I can see this in the console: Feb 22 22:24:00 DEBUG[10555]: pbx.c:1697 ...
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311 views

Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience ...
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1k views

Trixbox won't accept incoming calls on IAX trunk

any trixbox/asterisk experts out there? I have a trixbox installation that refuses to accept incoming calls via IAX2 from a certain VoIP provider. Initially, I thought it was a firewall problem but I ...
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6answers
762 views

Server Fax Farm - Need suggestions, or advice

We're Looking at creating a large fax farm via T.38 (Fax over Voip - hundreds of incoming and outgoing faxes) on linux servers, anyone have any suggestions on what is available? All my searches return ...
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105 views

where is the hold music file located on Asterisk?

would anyone know what the file name is (and the location) of the hold music file is in Asterisk?
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185 views

Is this Asterisk Application possible?

I have no experience with Asterisk and am looking to replicate an application I have running on the Voxeo platform. The workflow is this: Make an outbound call. If an answering service is detected, ...
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720 views

Migrate asterisk config files to database configuration

Does anyone have documentation or tooling to help migrate from a config file based asterisk deployment to a database deployment?
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149 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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1answer
110 views

Asterisk and FXO issue over DDNS

I've been experimenting with asteriskNow (freePBX/asterisk 11) as a sip server with DDNS. Everything seems to be working fine using externhost and such; calls to remote extensions to and from asterisk ...
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1answer
148 views

How to know if graceful restart / stop was executed?

Is there a way to know if restart gracefully or stop gracefully has been executed on asterisk while still running and waiting for calls to end then, restart/stop? When this mode is executed, Asterisk ...
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915 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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2answers
767 views

Asterisk issue with ConfBridge

I'm receiving an error when I try to call an extension that leads to ConfBridge() app in my dialplan Here is the extensions.conf exten => 601,1,Noop() same => ...
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1answer
111 views

Installing FreePBX or Asiterisk-GUI on working server

We have server with Ububuntu and asterisk 1.8 on it without any web-interfaces. Our managers ask me to make them web-interface for looking over config, users, calls and so on. I've installed ...
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295 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
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296 views

Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...
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1answer
273 views

Asterisk: Forcing a sip peer to connect via ipv6?

I've got an asterisk server that connects to an upstream provider over a WAN. The upstream provider supports both IPv4 and IPv6 connectivity, and the asterisk server is behind a NAT. When asterisk ...
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1answer
402 views

Routing Asterisk traffic evading a VPN

I have two offices set in two distant locations, connected through a IPSec VPN (using pfSense). On one office, I have an Asterisk PBX where all the extensions connect, and the Asterisk PBX is ...
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288 views

Troubleshooting a failing SIP connection

I have a linux Asterisk server that is behind a firewall. The firewall has three interfaces : eth0 is the LAN, where Asterisk resides, eth2 is the default gateway to the internet (via an ADSL modem ...
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1answer
1k views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
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1answer
784 views

How to play and record wav in Asterisk

Please, how could I do these things on asterisk: 1) Externally (with some API) make a call from asterisk and play wav and hangup the call 2) How should I configure the extension/route/whatever on the ...
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2answers
1k views

Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk

We have Trixbox CE 2.8 and Asterisk 1.6. I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls. I also want to remove the +1 for long distance so ...
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1answer
1k views

GUI for Asterisk 1.8.8 on Debian 6 [closed]

Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
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2answers
436 views

Asterisk and sip station codecs

we have a small instance running on ec2 of voxilla/FreePBX-Asterisk-1.6.0.5-i386.manifest which has been updated as far as i can see We signed up for sipstation (because it removes half the ...
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4k views

Asterisk - inbound calls from SIP DID trunk “rejected because extension not found”

I have a DID with DIDforSale that's pointed at my Asterisk server. When I call it from my landline phone I get the AT&T disconnected line recording. The Asterisk CLI shows the error message: [Oct ...
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2answers
1k views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
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1answer
457 views

SIP ( Asterisk ) CoS solutions with no VLAN's?

I have a Asterisk box actually two running pure asterisk. No GUI i.e. FreePBX, TL etc. The asterisk box is connected to a an EX4200 VC Stack, the EX4200's then connect to our headquarters via a Qwest ...
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1answer
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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1answer
129 views

Is there a minimum CPU requirement for a single call using G729?

I am just trying this out on an AWS micro VM, which gives out roughly the equivalent of a 600Mhz processor. Currently call quality is poor, but I am trying to ascertain if this is due to a lack of ...
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384 views

TrixBox Message Waiting Indicator in Device and User Mode

My trixbox installation supports hot-desking, so that users can take any desk and log into the phone. To do this, I placed FreePBX into Device-and-User mode, which allows the devices (phones) and ...
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1answer
1k views

Asterisk failing at startup after upgrading to asterisk18

I was using asterisk16 and asterisk16-skypeforasterisk, which was working fine. I have recently upgraded to asterisk18 and asterisk18-skypeforasterisk, after that I am receiving the following error ...