Asterisk PBX Support

Connect your open source PBX to our SIP trunks.

IPComms began connecting our SIP trunks to Asterisk® PBXs in 2002. And not to brag, but since then, we've successfully provided over 30,000 SIP/IAX trunks to almost every version of Asterisk on the market.

So, whether you're a forum-surfing, wiki-reading, ISO-burning open source PBX newbie or a full-fledged, Digium® certified, card-carrying member of every Asteriskusers group on the Web... we're sure to be the SIP Trunking service provider you want in your SIP.conf file

Our USA-based support staff is here to help you get your Asterisk PBX connected to our IPComms' SIP trunks. In fact, we'll prove it to you!  Signup for our Free SIP Trunk Trial and experience our extremely high-quality service and technical support for yourself.

Setting up your IAX Trunk inside PBX in a Flash

Setting up your IAX Trunk inside PBX in a Flash

This article will help you setup an IAX2 trunk in your PBX in a Flash system and connect it with IPComms SIP trunks..

  • Setting up a IAX Trunk is very similar to a SIP Trunk, the biggest difference in registration is the Register String. The IAX trunk contains more information than a SIP Trunk. Trunk information can be copied over just like setting up the SIP Trunks
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PBX in a Flash SIP Trunk Configuration & Security

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.

 

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What ports should I forward on my Router to make SIP work?

What ports should I forward on my Router to make SIP work?

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

 

PBX in a Flash (Resetting the root password)

Resetting a root Password

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.

    • Reboot your server
    • When you see the GRUB loader quickly press a key to disrupt the normal booting process
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FreePBX EndPoint Manager

EndPoint Manager

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.

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Setting up your IAX Trunk inside PBX in a Flash/FreePBX

Setting up your IAX Trunk inside PBX in a Flash

  • Setting up a IAX Trunk is very similar to a SIP Trunk, the biggest difference in registration is the Register String. The IAX trunk contains more information than a SIP Trunk. Trunk information can be copied over just like setting up the SIP Trunks
  • Make sure to set the registration string as; username:password@domain
  • If you would like to see if trunks are registered you can go to the FreePBX System Status and look at IP Trunk Registrations.
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Trunk Configuration with PBX in a Flash

Today we will be configuring a Trunk for service with IPComms, to begin we simply copy and paste the information from your registration.

  1. After entering your Trunk Configuration information, click Save Changes.
  2. To check any information you have entered simply go to FreePBX System Status, from here you will be able to see any IP Phones and IP Trunks that are online, as well as some other information about your PBX.
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Configuring Inbound Routes with PBX in a Flash / FreePBX

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.

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Connecting SIP trunks with IP Authentication (Asterisk/FreePBX)

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.

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IPComms SIP Trunk Registration (Asterisk/FreePBX)

IPComms SIP Trunk Registration (Asterisk/FreePBX)

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.

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11 Steps to Secure your PBX

11 Steps to Secure your PBX

 

Don't be a victim of telecom theft

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. 

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