How to connect two Asterisk PBXs using a SIP Peer/User Trunk Pairing

Thanks to: https://www.telecomworld101.com/ 

Session Initiation Protocol (SIP)) is a signalling protocol used for setting up and tearing down Voice over Internet Protocol (VOIP) calls. Voice over Internet Protocol makes the transition from traditional conference calls to conference calling via the world wide web. A SIP call uses two protocols: SIP and RTP. Real Time Protocol (RTP) is used for the transfer of the actual voice data. If you want to find out more about SIP visit Wikipedia's SIP page.

The first step in setting up an SIP trunk is to draw a picture of what you need to do. Here's an example of a simple PBX to PBX connection that will be using a User/Peer pairing to form a SIP trunk. The two PBXs are name 106 and 111 after their IP host address. They could very well be named Montreal and New York. PBX 106 has all their extensions starting with 3xxx while PBX 111 has all of their extensions starting with 2xxx. This will be handy when making outbound routes.

The SIP trunks are drawn as arrows pointing to their PBX peer and named based on their destination which seems like a good practise. 111-Peer is going from PBX 106 to PBX 111 and 106-Peer is going from PBX 111 to PBX 106. I've left an area on both sides for configuration information.

Note: It is good practise to indicate the protocol used in the naming of trunks, users and peers (ex. 106-SIPpeer). This is very important if you are using multiple protocols for trunking (IAX, SIP and T1). Adding the protocol to the trunk names will create a unique entry and prevent unintentional confusion in the dialplans between trunking protocols!

On the PBX 106 side, I will need to configure an outbound trunk called 111-peer which will connect to the opposite PBX using the account 106-user. PBX 111 will need to create a user account called 106-user on the inbound trunk. It makes sense to call the user account 106-user because that's who is going to register to PBX 111.

Additionally, on PBX 106, we will need an user account called 111-user so that PBX 111's outbound trunk 106-peer can register to. On PBX 111's outbound trunk 106-peer, we tell it to use user 111-user.

The trunk names and usernames can be called anything you like. I tried to use names that would help explain what is happening.

I've made up a SIP trunk using Peer/User pairing configuration tool in an Excel spreadsheet that creates both PBX 106 and PBX 111's trunk configuration. It is easy and fast to do and takes all the guess work out of it. You enter in the IP address (or domain name) of each PBX, the names for the two trunks, the names for the two users and the user passwords. It spits out the configurations for both PBXs in the same format that you see in the FreePBX Add Trunk menu.

The following pages go through the steps to configure the PBXs using the FreePBX interface. We will configure the trunks one side at a time starting with PBX 106. Once both PBXs have their SIP trunks up, we will configure the outbound routes.

1. Configuring the SIP Trunks
   1. Configuring PBX 106 SIP Trunk
   2. Configuring PBX 111 SIP Trunk
   3. Testing the SIP trunks

2. Configuring the Outbound Routes

   We will be configuring the outbound route for dialing directly to the extension of the peer PBX.

   • Creating PBX 106's Outbound Route
   • Creating PBX 111's Outbound Route

3. Now you should be able to dial through each PBX to its peer from any SIP, IAX2 or POTS extension. You can check the status of the phones online and trunks online through FreePBX Statistics window

   

   In creating the trunks, there was no limit put on the maximum number of channels that can use the trunk. For the above FreePBX Statistics window, I had 4 phones (channels) connected in 2 connections (external calls) across the SIP trunk. There are two IP trunks shown here as one is an IAX2 trunk and the other the newly created SIP trunk.

   Two channels were IP phones, one was an IAX2 S100i POTS to IAX2 adapter and one FXS pots phone. All worked beautifully! You don't have to configure any protocol translations - the PBX does it all for you.

1. Configuring the SIP Trunks

Configuring PBX 106 SIP trunk 

We are going to create a SIP trunk called 111-peer that will connect to PBX 111. At PBX 111 we will be connecting to PBX 111's sip trunk called 106-peer. Confusing? Yes sir! Select Add Trunk from the FreePBX main setup menu Select Add SIP Trunk Nothing to do here, so go to the Outgoing Settings section Above are the default values which we will change to Here's the explanation of the changes: Trunk Name: 111-peer - you can name this anything you like, we're going to PBX 111 so 111-peer sounds like a good name host=192.168.1.111 - IP address or domain name of the peer PBX you want to connect to username=106-user - this is the name of PBX 111's user account to authenticate to. fromuser=106-user - this is used during authentication during the SIP invite secret=106-password - this is the password that is used to authenticate the 111-peer SIP trunk to PBX 111. type=peer - this indicates that this trunk is the peer. qualify=yes - this line is optional. It periodically pings its peer to keep the connection alive. Here's the default settings for the Incoming Settings. Configure the user account for PBX 111 in this section: 111-user - This creates the account that PBX 111 will use on PBX 106 secret=111-password - This is the password for 111-user account type=user - This is a user account in the user/peer pairing context=from-trunk - This account is accessed by a trunk Press submit, update and reload. Now go to Configuring PBX 111 SIP trunk Configuring PBX 111 SIP trunk We are going to create a SIP trunk called 106-peer that will connect to PBX 106. At PBX 106 we will be connecting to PBX 106's sip trunk called 111-peer. Select Add Trunk from the FreePBX main setup menu Select Add SIP Trunk Nothing to do here, so go to the Outgoing Settings section We will change these default values to: Here's the explanation of the changes: Trunk Name: 106-peer - you can name this anything you like, we're going to PBX 106 so 106-peer sounds like a good name host=192.168.1.106 - IP address or domain name of the peer PBX you want to connect to username=111-user - this is the name of the SIP trunk coming from PBX 106 fromuser=111-user - this is required by the SIP invite authentication process secret=111-password - this is the password that is used to authenticate the 111-user account type=peer - sets the trunk as a peer in the user/peer pairing qualify=yes - this line is optional. Periodically pings to keep the connection alive. Here's the default settings for the Incoming Settings. Configure the user account in this section: 106-user - This creates the account that PBX 106 will use on PBX 111 secret=106-password - This is the password for 106-user account type=user - This is a user account in the user/peer pairing context=from-trunk - This account is accessed by a trunk Press submit, update and reload. Now go to Testing a SIP trunk Testing a SIP trunk  Testing PBX 106 At this point, we are ready to verify that the SIP trunk is alive. The "qualify=yes" line sends an option packet every 60 seconds to see if the destination is alive. If the destination does not respond within 2 seconds for 7 tries in a row, it will be marked as unreachable. At the asterisk CLI for PBX 106, I've typed the command 'sip show peers": 111-peer/106-peer means: 111-peer - this is the trunk name 106-peer - this is the username 192.168.1.111 - This is the address of the PBX that we are trunking to Dyn - Is it dynamic port addressing or not. I believe that dynamic is for SIP phone extensions and blank is for SIP trunks. I've also read that is could be for dynamic IP addressing. I'll verify this in the lab. Nat - N - We haven't enabled Network Address Translation (NAT) for this trunk ACL -Not sure what this means at this time Port - This is the port that SIP is operating on. It should be 5060 for a SIP trunk. The other ports are dynamically assigned for SIP extensions. Status - If you have the line "qualify=yes", the status will be "OK" with a time in brackets. If you don't add the line, then it will be indicated as "unmonitored" which makes me uncomfortable. Regardless the trunk will still work The result is that we are connecting on trunk 111-peer using user 106-peer to the PBX at 192.168.1.111 using port 5060, not NAT and things are OK with a 1 mS ping time. Even if it displays "OK", you may still have SIP authentication issues and the dreaded "All circuits are busy now. Please try your call later." If you are receiving this error message, consult the Troubleshooting SIP webpages. Testing PBX 111 At the asterisk CLI for PBX 111, I've typed the same command 'sip show peers": This verifies that we are connecting on trunk 106-peer using user 111-peer to the PBX at 192.168.1.106 using port 5060, not NAT and things are OK with a 1 mS ping time. Even if it displays "OK", you may still have SIP authentication issues and the dreaded "All circuits are busy now. Please try your call later.". If you are receiving this error message, consult the Troubleshooting SIP webpages. Now it's time to create the Outbound Routes for PBX 106 2. Configuring the Outbound Routes We will be configuring the outbound route for dialing directly to the extension of the peer PBX. Configuring PBX 106's Outbound SIP Trunk This example will configure an Outbound Route so that PBX 106 extensions can dial PBX 111 extensions directly. You start by selecting Outbound Routes Direct Dial PBX 111's Extensions This outbound rule allows PBX 106 extensions (3xxx) to directly dial PBX 111's extensions (2xxx). For example, to dial PBX 111's extension 2001. You dial 2001. Three things to configure: Route Name: 111-dial-2xxx - Can be anything, be descriptive so you remember what it is 6 months from now when it stops working! Dial Patterns: 2xxx - This says any 4 digit extension starting with 2 will be forwarded to the designated trunk. Trunk Sequence: SIP/111-peer - This is the trunk that we configured that goes to PBX 111 Now it's time to create the Outbound Routes for PBX 111 Configuring PBX 111's Outbound SIP Trunk  Configuring PBX 111's Outbound SIP Trunk This example will configure an Outbound Route so that PBX 111 extensions can dial PBX 106 extensions directly. You start by selecting Outbound Routes Direct Dial PBX 106's Extensions This outbound rule allows PBX 111 extensions (2xxx) to directly dial PBX 106's extensions (3xxx). For example, to dial PBX 106's extension 3001. You dial 3001. Three things to configure: Route Name: 106-dial-3xxx - Can be anything, be descriptive so you remember what it is 6 months from now when it stops working! Dial Patterns: 3xxx - This says any 4 digit extension starting with 3 will be forwarded to the designated trunk. Trunk Sequence: SIP/106-peer - This is the trunk that we configured that goes to PBX 106 Now return to Step 3 of How to connect two Asterisk PBXs using a SIP Trunk to test your new configuration by dialing directly from one PBX to the other's extensions!