If you own or operate an Asterisk PBX, trust us, security will be a priority for you... either now or later! If you only do one thing to secure your PBX, take this next piece of advice seriously! What ever you do, no matter how tempting it may be, Never, Never, Never...
This article will help you setup an IAX2 trunk in your PBX in a Flash system and connect it with IPComms SIP trunks..
Start this tutorial after you have completed PBX in a Flash Setup. After Installation, you will need to obtain your IP Address. Once the IP Address has been typed in you will be able to see PBX in a Flash with the Icons: Voicemail & Recordings, Flash Operator Panel, and MeetMe Conference for users, and FreePBX Administration, Linux Webmin, and Menu Configuration for the Admin user.
SIP uses TCP and UDP protocols to carry its call control information (not the payload) and is usually carried on ports 5060 and 5061. The actual payload is transmitted using the RTP protocol (Real-time Transport Protocol) which is specifically designed to carry payloads that are time-sensitive information such as voice and video.
RTP has a broad range of ports assigned 16384 - 32767. However different SIP vendors use different ports they may or may not fall within this range.
Here are the ports needed for SIP to work.
• Call control: Ports 5060 and 5061
• RTP audio: Ports 16384 - 32767
The causes of one-way audio in IP Telephony can be varied, but the root of the problem usually involves IP routing issues. This article takes a look at some of the most common scenarios and solutions that have been experienced by our technicians.
How to reset a root password in PIAF and generic RHEL(Red Hat Enterprise Linux) based systems.
Having the ability to reset your PIAF password in-case of a lock-out is very vital when it's necessary to keep an open communication. Resetting a password may take a few minutes.
EndPoint Manager is a module within FreePBX®, that can be used to install and provision IP phones as well as manage firmware updates. This is a very useful tool that works with the most of the major brands. As an example we will setup a Cisco phone, to begin select Install on Cisco. Next, you will see available models for that brand, select Enable for your current model. Next, go to the Advanced Settings and set the IP Address of the PBX, and set the directory where phones will update the firmware from.
Today we will be configuring a Trunk for service with IPComms, to begin we simply copy and paste the information from your registration.
Inbound routes are very important if you want to have numbers routed to a specific destination(s). With this current setup, if you are calling 6784601475 (DID Number) and you are calling from 7702180222 (CallerID Number) the call will come in as it is setup below with music on hold, signal ringing, and a 3-second pause before it goes to the destination set below (Marcus Cell). If your provider does not provide inbound Caller ID, the Caller ID (CID) Superfecta may be a work around.
The current build was done on Ubuntu 12.04.3 LTS. This should world on Debian Wheezy and Higher.
This is a vanilla install of Asterisk 13, with no Web Interface or extra features.
IPComms allows two types of SIP trunking when connecting to our network. Our default registration method and by far the most common, is basic SIP Registration. This method uses a SIP username and password with a registration string to connect to our SIP network. The second methog, which is less common, but useful in many scenarios, is SIP IP Authentication.
This article will cover registering your Asterisk PBX to IPComms using SIP IP Authentication.
The first step in making and receiving phone calls using the IPComms SIP trunking network is registering your SIP device to our network using SIP registration. This article will cover registering your Asterisk PBX to IPComms using SIP IP Authentication.
Auto Attendants allow your calls to be automatically answered for you 24/7, giving you time to focus on your business without having to manage incoming call routing.
Recordings give you the opportunity to play pre-recorded messages, to your callers. Once you have a recording made you can use the recordings in different area’s of MyOffice PBX. In order to create a custom Auto Attendant (IVR), we must first create a custom recordings. There are two ways to create a recording. You can record one via your phone or upload a pre-recorded ".wav" file.
In this example we'll show you how group extensions together into groups and define a rule for delivering calls to extensions within that group.
Outbound call routing is automatically setup by IPComms. There is no need to make any outbound routing changes.
Next, we'll show you how to route incoming calls. Setting up an inbound destination determines where an incoming call will go (e.g.extension, IVR Menu, Ring Group, external phone number).
MyOffice PBX works with most SIP-based phones and other VoIP devices. In this example, we'll use the free softphone from ZoIPer (Windows version) to register to one of our newly created extensions.
The account settings for the currently logged on user. Status, Language, Time Zone and Password
If you are reading this, you're probably like most of us... after many hours, or even several days of downloading software, setting up servers, configuring trunks and cracking open firewall ports, you finally achieve success - your PBX is working, and calls are passing. So, you wipe the sweat from your forehead, push away your ergonomic mesh-backed office chair (with lumbar support) and walk away pleased - not giving a second thought to security. Until one day, you log into your PBX and see the skull-and-boned call sign of a hacker that has decided to pay you’re perfectly running PBX a visit.