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Flowroute

Change FreePBX 13 to use alternate SIP port 5160

This article describes how to change your Asterisk-based system to use Flowroute's alternate SIP port 5160 for all SIP signaling. Making this change is effective in avoiding ISP or internet backbone VoIP issues that occur on the standard SIP port 5060 that most third-party VoIP systems use. This article assumes you're comfortable with editing Asterisk configuration files or your PBX's admin web interface, changing the local SIP port on your desk phones and/or softphones, and that you have some computer networking understanding. 

IMPORTANT:

  • Changing the port might require taking your system offline. 

  • Because you might need to change the local SIP port on each user's phone to port 5160, a considerable amount of downtime to complete this task might be required. 

We recommend only making these changes during off-hours, or when you can allow for a sufficient amount of downtime.

Before you begin

  • You will need to command line to your Asterisk system's config files or admin web interface access to your Flowroute Trunk peer details.

To change your SIP port to 5160:

  1. Do one of the following: 

    • Go to /etc/asterisk/, or 

    • Use your PBX web interface to edit your Flowroute Trunk. 

  2. Set the port to 5160 through one of the following methods: :

    • In /etc/asterisk/, open sip.conf with a text editor; or

    • In the PBX web interface, edit the Trunk Peer Details in your system's web interface by adding the following information: 

      port=5160
      bindport=5160  

      If you're using SIP registration, add 5160 to the end of your registration string so it resembles the following:

      register => 55555555:GcMe7555cKnw@sip.flowroute.com:5160 

      IMPORTANT: SIP Registration for New PoPs

      If you are taking advantage of our new Points of Presence (PoPs), make sure to update the domain portion, sip.flowroute.com, in this format: {your_preferred_pop}.sip.flowroute.com where {your_preferred_pop} might be "us-west-wa" for example. Your new domain will then be us-west-wa.sip.flowroute.com. See Interconnection with the New PoPs for technical specifications.

  3. Restart Asterisk or your reboot your PBX. 

    NOTE: Consult your PBX documentation on how to perform this. Many PBXs can restart Asterisk by running the following commands:
    service restart asterisk 
    /etc/init.d/asterisk restart

If your desk phones do not work after the changes, you might need to change the local SIP port on each one to 5160.  Please consult with your phone manufacturer's documentation if you're unsure how to do this.

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