IAX to SIP Migration Process

Step 1 – Asses your current system setup

The majority of systems/PBXs that have been configured for IAX will also have the ability to utilise SIP. However, we recommend checking this before making any plans to migrate your service.

Step 2 – Configuring your Network for SIP calls

As IAX only requires port 4569 to be open, the network which contains your system will need to allow for additional ports and IPs.

VoIP Gateway IP Ranges  
77.240.61.165/27 

77.240.56.34/27 

Proxy Domains and IP Addresses 

voiptalk.org (77.240.48.94)

draytel.org (185.158.58.194) 

If your router allows you to, we strongly recommend whitelisting our NAT Proxy domains and the domain you use to register against our gateway. 

NAT Proxy Domains and IP Addresses 

nat.voiptalk.org (77.240.48.201)

nat.draytel.org (185.158.58.201)

SIP and RTP traffic will traverse through these ports:

SIP Ports (5060 - 5065), UDP Protocol

RTP Ports (10000 - 40000), UDP Protocol

We strongly recommend configuring your network to allow traffic on these ports, but only from the IP ranges mentioned above.

Step 3 - Testing connectivity and registrations via a Soft Phone

Once your network is allowing our IPs and the port ranges, we recommend connecting a softphone to your SIP trunk to check that your network can successfully allow calls out and receive calls in as well.

MicroSIP

Desktop: https://help.telappliant.com/softphones/register-microsip-with-a-voiptalk-sip-trunk

Zoiper

Desktop: https://help.telappliant.com/zoiper2/zoiper-desktop-setup-for-voiptalk

In your chosen Soft Phone’s configuration menu, there should be an option to define a SIP Server, SIP Registrar or SIP Domain value. Set the value to: voiptalk.org.

For SIP Authentication, set your SIP User ID or similar to your VoIPtalk ID (eg 84411076) and set the Password to what has been set on the VoIPTalk Portal.

If you have an Outbound Proxy  or SIP Proxy setting, set this to: nat.voiptalk.org:5065. If this setting allows you to define the port in a separate field, set the IP Address to: nat.voiptalk.org and the port number to: 5065.

Save all settings just made on the soft phone and check within the application that the registration to your trunk was successful.

Step 4 – Apply SIP configuration to the live system

Upon a successful test via the Softphone. The same SIP configuration can now be applied to your system which was previously configured for IAX. 

You will need the same SIP domain and proxy:

• voiptalk.org & nat.voiptalk.org

as well as the SIP User ID and password:

• on the VoIPTalk portal. Instructions to find your SIP trunk’s password is detailed in the section below.

Please see below for details on obtaining your SIP trunk details through the VoIPTalk portal.

Step 5 - Obtaining your Trunk details via the VoIPTalk portal

Login to your VoIPTalk Account - https://www.voiptalk.org/

You will then be taken to your voiptalk account dashboard which will display your Trunk IDs. Click on the Trunk ID to open up a new window (make sure to allow pop ups on your browser if prompted), this window will display the Trunk ID, as well as the password, and will also allow you to change the password if required.


 

Step 6 - Updating your Number Routing

On your voiptalk portal, you'll need to update the routing of each DID on your account to change the routing from IAX to SIP. This is required for your inbound calls to work over the SIP protocol. You can do this by following the below steps

1. Click on Incoming Numbers
   
   
    
2. Click the dropdown “Choose Incoming Number” and select the number you wish to update the routing of. Click “Display”
   
    
3. Click the dropdown “Select a Configuration Option” and select the appropriate routing choice (Described Below). Click “Select Option”
   
    
4. Fill out the fields of your selected Routing option and Click Route/Divert/Save Changes etc.
   
    
5. Perform the above steps for any additional numbers which require rerouting

Please find information on the routing options below. The dialstring of the current routing can be seen in the "Current Status" Field

• Configure Caller ID: This option allows you to select the current number as the default Caller ID for all calls placed outbound from the trunk.
   
• Route to a voiptalk ID: This will set calls to be sent to any devices registered to the trunk. Calls sent to devices in this manner will be addressed to the trunk, i.e. the URI will be configured like “Request-URI: TrunkID@voiptalk.org”.
   
• Route to a PBX by IP or DNS: Select “SIP-Compatible PBX”. This allows you to send calls to a specific IP/Domain. This is recomended if you want to use IP Authenthication. Calls sent in this manner will have their URI configured like so “Request-URI: Number@XX.XX.XX.XX”.
   
• Route to SIP PBX: This option configures calls to be sent to a SIP PBX. Calls sent in this manner will have their URI configured like so “Request-URI: DDI-TrunkID-Number@voiptalk.org” which inbound number matching rules on the PBX will need to take into account.
   
• Divert Number: Allows you to set a PSTN number as a divert destination.
   
• Route to Voffice: See “Route to SIP PBX”
  .
• Route to External Provider: If you'd like to route your traffic to an external SIP provider or Domain, you can use this option to do so. Enter the username of the external domain you wish to send calls to in the "VoIP ID" field and the External domain in the "VoIP Provider" field. Calls sent in this manner will have their URI configured like so “Request-URI: VoIPID@VoIPProvider".

The following options can be disregarded as they are in the process of being discontinued.

• Route to my IAXy
• Advanced Divert