Premium SIPStation SIP Trunking encrypts SIP and RTP with TLS and SRTP between your PBX site and Sangoma's Data Centers. This feature is presently under BETA testing. If you would like to be part of the testing, please complete this short survey first → https://www.surveymonkey.com/r/Y2JCDS9
Prior to using this page as a guide, our technical staff have to enable the feature in our back office systems.
Locating your SIP Username and Password
- Log into your SIPStation.com account.
- Click on My Account in the green navigation menu.
- You will find your SIP Username and SIP Password under the "Account Configuration" section. Take note of these, as you will need to use them when you set up trunks in Switchvox.
Setting up your Trunks in the Switchvox GUI
- Log into your Switchvox server.
- Navigate to System Setup → VOIP Providers.
- Under "VOIP Providers," click on the Create SIP Provider button.
- Click on the Provider Information tab and fill out the following information:
- SIP Provider Name: SIPStation1
- Your Account ID: Enter your SIP Username from SIPStation
- Your Password: Enter your SIP Password from SIPStation
- Hostname/IP address: trunk1.freepbx.com
- Callback Extension: Should be a default extension on your system as per Switchvox Requirements
- DTMF Mode: RFC2833
- Click on the Peer Settings tab and fill out the following information:
- Host Type: Provider
- Host is a Switchvox PBX: No
- Treat system's users like local users: No
- Apply Incoming Call Rules to Provider: Verify this is set to Yes
- Click on the Caller ID Settings tab and fill out the following information:
- Supports Changing Caller ID: Yes
- Caller-ID method: From Header
- Click on the Connections Settings tab and fill out the following information:
- SIP Port: 5060
- SIP Expiry: 120
- Always Trust this Provider: Yes
- Qualify Host: Yes
- Click on the Call Settings tab and fill out the following information:
- ULAW: Yes
- ALAW: Yes
- When done, click the Save SIP Provider button at the bottom of the page to save all your settings.
- Important: Setup is not yet complete. You will need to register to a second trunk.
- Since SIPStation service is set up with two redundant SIP Servers, you need to register to both trunk1.freepbx.com and trunk2.freepbx.com. Inbound calls can come from from either server at any time, and you can send outbound calls to either server.
- Once you have set up trunk1.freepbx.com by following the steps above, please set up a second trunk to trunk2.freepbx.com.
- Use the same information as before, except name the trunk "SIPStation2" and enter "trunk2.freepbx.com" as the Hostname under the Provider Information section.
- After you have set up the second trunk, you can continue to the steps below.
Verifying the Trunks are Registered in the Switchvox GUI
- In the Switchvox GUI, navigate to Server → Connection Status
- Verify your two newly created trunks are registered. (Remember, you need to set up trunks to both trunk1.freepbx.com and trunk2.freepbx.com as described earlier.)
Creating Outgoing Call Rules in Switchvox
- Navigate to Setup → Call Routing → Outgoing Calls
- Create dial pattern rules. Make sure any of the routes that you want to use with SIPStation are set up to use both of your SIPStation trunks, as shown in the next step below.
- Configure your routes to use both of your SIPStation trunks. Set one SIPStation trunk as a primary and another as a failover.
Creating Incoming Call Rules in Switchvox
- Navigate to Setup → Call Routing → Incoming Calls
- Set up inbound rules for any of the DIDs you have purchased on your SIPStation account. Since calls be be delivered from both trunks, make sure the Incoming Provider rule is set to match Any Provider.