This wiki explains how to use the FreePBX SIPSTATION Module in conjunction with SIPStation SIP Trunk service. The SIPSTATION Module helps you configure your trunks, routes, and DIDs in just a few clicks. This wiki assumes you have already set up a SIPStation account as described in our wiki "Creating a SIPStation Account."
The SIPSTATION Module, when combined with a SIPStation SIP Trunk account, provides a quick and easy method for getting a SIP Trunk online and ready to go fast. In general, there is not much to adjust here besides pointing your new DIDs or adding an Area Code for 7-digit dialing.
Table of Contents
Downloading or Installing the SIPStation Module
- The preferred solution to install the module is to go to Module Admin in your GUI and install directly from there. This will assure you have the most up-to-date version of the module that is compatible with your version of FreePBX.
Using the SIPStation Module
Adding a SIPStation FreePBX Module Keycode
- Navigate to your SIPSTATION Module, where you should see a screen like this:
- Log into the SIPStation Store from within the module. If you have multiple registrations, the system will ask you to choose one.
- Click "My Account."
- click on the Account Configuration tab
- You will see your FreePBX Module Keycode. You are going to need to copy it, but it is very long, so we recommend that you simply click the "Copy to Clipboard" button to copy the information to your computer's clipboard.
- Scroll back up the page in your SIPSTATION Module and look for the Account Key field. Paste your account key into this field.
- Click "Add Key" to connect your SIPStation service to FreePBX.
- After you have entered your key, your screen might look like this:
- Click the "Apply Config" button at the top of the module.
- Now, your screen should look like this:
Note that you would normally see the entire external IP Address.
Managing an Account Key, Refreshing Account Info, & Running a Firewall Test
After you associate a SIPStation account key with your SIPSTATION module, you will see several new buttons:
- Remove Key- Removes the account key but leaves the trunks intact.
- Remove Key & Delete Trunks- Removes the account key and removes the trunks.
- Refresh Asterisk Account Info- Updates the module to reflect changes you've made in your SIPStation account. For example, if you purchase a new DID, you will would want to come back to the SIPSTATION module and update the account information by clicking this button followed by the "Apply Config" button.
- Run Firewall Test- Running the firewall test will tell you if you have RTP media port forwarding set up correctly on an external firewall. Note that it is recommended that you do set up SIP RTP port forwarding for the full sip media port range, as it will provide more reliable audio for SIP Trunking. You can read more about port forwarding in our wiki "Configuring your PBX or device with SIPStation Service."
- Your firewall test should come back with a "PASS" status
- Your firewall test should come back with a "PASS" status
The System Status section displays current settings and allows you to check for any warnings.
Note that you will see your actual Ip information as the pictures below are just an example.
- Asterisk Registration Status- Displays the Asterisk Registration status.
- Your Contact IP- This is the contact IP as seen on the gateway. Warnings are provided if errors are detected. This should be your external IP as seen on the WAN side of your router. If it is not, or does not match your Network IP, you should configure your NAT Settings in the Asterisk SIP Settings module or in sip_nat.conf (if not using that module).
- Your Network IP- This is the network IP as seen on the gateway. Warnings are provided if errors are detected. This should be your external IP as seen on the WAN side of your router. If it is not, or it does not match your Network IP, you should configure your NAT Settings in the Asterisk SIP Settings module or in sip_nat.conf (if not using that module.)
- SIP Ping- This is the round-trip signaling delay to the SIP server as determined by the Asterisk ‘qualify’ command. This is signaling delay only. The voice connections (RTP media streams) are routed from your system to the closest POP (Point Of Presence) where the call enters the PSTN. This assures the optimal minimum latency, but can’t be reported because it is dependent on each call.
- Codec Priorities- This is the Codec Priority Asterisk reports for these trunks. This is filtered to show codecs supported by the gateways. The Codecs can be edited on the trunk page to make changes to priority or available codecs.
Firewall Test- Shows the status of your firewall test intended to assist with RTP media port forwarding configuration for external firewall/router devices. When a firewall test is initiated, a free port is chosen from the defined SIP RTP media range on the PBX and an API call is made to the FreePBX mirror server using http/https. The mirror server sends back a sample SIP media payload to the chosen RTP port. If the payload is received successfully, the test passes. This test is not definitive pass/fail. It can assist with config, but may not catch all configuration edge cases for all devices.
The Account Settings area shows you your current settings and allows you to make changes in certain areas. If you make any changes, click the "Update Account Info" button at the bottom of this section, followed by the "Apply Config" button at the top of the module.
- Global Failover Number- The number to which to route incoming calls in case your PBX can't be reached. Please see our "Failover Options" wiki for more information.
- Global Failover IP/FQDN- The IP address or Fully Qualified Domain Name (FQDN) to which to route incoming calls in case your PBX can't be reached. Please see our "Failover Options" wiki for more information.
- International Calling- Shows whether international calling is enabled. Please see our "International Outbound Calling" wiki for more information.
- Outbound Fax- Shows whether outbound faxing is enabled. Please see our "Outbound T38 Faxing" wiki for more information.
- SMS Support- Shows whether SMS (text messaging) is enabled.
- Edit E911 Information- This displays and allows you to edit your Master E911 address information on file. It is vitally important that this information is correct. Please review our e911 wiki.
Route and Trunk Configuration
The Route and Trunk Configuration section shows you which trunk(s) each route is allowed to use, and gives you the opportunity to enable 7-digit dialing. If you make any changes, click the "Update Route/Trunk Configurations" button, followed by the "Apply Config."
- Area Code- You may enter an Area Code if you wish your trunks to allow 7-digit dialing. You must ensure 7-digit dialing is enabled for your Outbound Routes. More information on this topic is available in our "Outbound Routes Module User Guide" wiki.
- Routes- By using the gw1 and gw2 toggles, you may set which outbound routes use these trunks.
DID Configuration is found at the bottom of the module. Here you will see a list of your DIDs, a description (if desired), and where the DID is routed to. If a DID is routed to an extension, you can choose whether you would like the extension CID to be set as the DID. You can also set failover numbers for each DID, change the Emergency CID, and see the information for the most recent inbound call. The color of the DIDs relates to their E911 status, which is explained in our "E911 Service" wiki. If you make any changes here, click the "Update DID Configurations" button followed by the "Apply Config" button.
Our SIPStation wiki contains answers to frequently asked questions related to the service. Visit "SIPStation and FAXStation" to navigate around various SIPStation-related wikis.
For SIPStation-related technical support, please open a ticket in our online Support Center. The process is described in our "Support" wiki.
SIPstation module receives data from push2.schmoozecom.com and you need to allow traffic from there.