SIP Forwarding in AVOXI (Inbound Forwarding)

SIP forwarding gives AVOXI Admins the ability to forward incoming calls to your AVOXI virtual number to a SIP address or PBX.  Just like a toll-free number can be forwarded to a landline or mobile telephone, it can also be forwarded to VoIP (Voice over Internet Protocol), or also referred to as SIP.  Forwarding to SIP works the same as call forwarding in that you can forward calls based on a specific action or preset rules. These rules can be configured or updated at any time via your AVOXI online portal.  

SIP TRUNKING:  Use the following SIP Trunking Configuration Guide to configure SIP for outbound termination.

Use the below guides and quick links  to configure the SIP account all calls should be diverted too: 


  1. First, log into your AVOXI platform
  2. Navigate to the SIP Trunks section located on the left-hand navigation bar
  3. Click on the SIP URIs tab
  4. Click on the ‘+ Add’ button to the right of the filter magnifying glass
  5. The SIP URI configuration tool consists of 4 configurable fields and 1 auto-generated preview field.
    1. Name:  Meaningful name of the new SIP URI configuration.
    2. SIP Connection: How you would like to route your incoming call traffic.
    3. Protocol: A transport protocol for your SIP URI configuration. AVOXI supports UDP, TCP, and TLS(SRTP), for more information about the differences between these three protocols, please refer here!  
    4. URI: This is the configurable URI you will be using to set up your forwarding. There are three components. 
      1. The caller ID format is expected from the dialed AVOXI number, the three format options are e.164 with or without a ‘+’ and a custom DINS field.
      2. The public domain name or IP address target is used by your particular phone system. In this example, Twilio’s public domain is used. For more information about the diverse targets. 
      3. Last, is the port to which your URI is listening, the default is set to 5060 but based on your solution, it may be different. 
    5. Preview: This is a preview of your full SIP URI
    6. Click ‘Add New SIP URI’ to save your configuration. 
    7. Select ‘Cancel’ to cancel out of the configuration tool.

Update an Existing SIP URI

A reminder that editing an existing SIP URI configuration will change the SIP URI settings for all applicable numbers. If you still wish to proceed please follow the below-detailed instruction.

  1. Navigate to the SIP Trunks section located on the left-hand navigation bar.
  2. Select on the SIP URIs tab.
  3. Click on the "SIP URI Name" you wish to update. 
  4. On the edit page, make your changes to either the Protocol or URI portion of your configuration.
  5. Review the "Preview" to ensure this is displaying as required.
  6. Once satisfied select the "Save" button to complete your configuration update.

Deleting an Existing SIP URI

  • On the SIP URI tab on the SIP Trunks page, click on the ‘Delete’ button next to the SIP URI configuration you would like to delete. 

  • After selecting the ‘Delete’, it is vital your review the confirmation message to ensure how this will impact additional settings such as Virtual Attendant, Forwarding Rules, and Teams. 

  • When you are 100% sure, select the ‘I understand, delete this SIP URI’ to remove it from the list.

Downloading SIP URI TLS Certificate

Transport Layer Security (TLS) certificates, also known as Secure Sockets Layer (SSL), work the same as HTTPS and are crucial for securing internet browser connections through data encryptions.  Similar to HTTPS when a browser tries to connect using TLS the secure web certificate is sent. As a customer, you can configure and manage the TLS/SRTP within your AVOXI online portal. Certificates are used for authentication and encryption.  

There is an SSL (TLS) certificate for  used for calls coming from the user to the AVOXI Genius platform, and for calls leaving AVOXI Genius.  As an AVOXI Administrator you are required to manually load the certificate during the configuration process to ensure the TLS connection are made (example  

Use the below steps to download the required certificate file from within your AVOXI online application.

  • Navigate to the SIP Trunks section located on the left-hand navigation bar.
  • Click on the SIP URIs tab.
  • Select the three buttons located to the right of the "+Add" button
  • Simply click on the "Download SIP URI TLS Certificate

Adding SIP URIs to Routing Rule Targets

Once you have set up your new SIP URI, you can now set them as targets for your forwarding rules. Not only can you set SIP URIs for call forwarding rules, but also for Teams, Virtual Attendants, and timeout targets.  

For the below example, the SIP URI is being set as a Call Forwarding rule target:

  • Navigate to the number where you want to set your configured SIP URI 
  • Click the number and then the ‘Forwarding’ tab 
  • Use the three lines located next to the "Enabled" icon to dropdown and select SIP and your preferred SIP URI.

  • Click the edit button on the rule and navigate to the ‘Forward Calls To

  • Once you are done selecting, click Save
  • Navigate back to the Numbers page to see that the target of that newly configured numbered is set to "Forward to SIP"

Configuration Details

Now you have added your SIP Trunking to your AVOXI account, it is important to configure it on your communications infrastructure/platform.


AVOXI supports UDP, TCP, and TLS(SRTP version1.2 (follow here to learn more).  Open the following ports to ensure proper connectivity:

  • 5060 UDP
  • 5060 TCP
  • 5061 TLS(SRTP)


AVOXI supports the below Codecs 

  • G.711u(ulaw), 
  • G.711a(alaw), 
  • G.729




  • RTP port range for AVOX10000-65535
  • SRTP
    • SSL (TLS) certificates for are used for calls coming from the user to the AVOXI Genius platform, and for calls leaving AVOXI Genius.  As an AVOXI Administrator, you are required to manually load the certificate during the configuration process to ensure the TLS connection are made (for example
  • RTP Audio Media Steam IP Addresses

WARNING: If you do not whitelist the designated IP's you will receive an error message.

Configuring Third-Party Platforms 

Use the below quick links to view the relevant configuration guide:

Forwarding to SIP FAQ's

  • Will all inbound calls be diverted to all the signed and active SIP devices and programs at that moment?  Each Avoxi number is configured separately but honors all current forwarding rules.

  • WIll the outbound portion of an international call be cheaper when forwarding to SIPYes,  we do not charge to terminate rates to SIP

  • If nobody answers the call by means of a SIP device will the call be redirected - if so to where? It depends on the timeout destination and forwarding rules configuration set up in Genius.

  • If it is busy the call will the call be diverted to voicemail where a caller can leave a message?  Yes, only if setup in Genius either through timeout destination or rules configuration.

  • What is the difference between UDP, TCP, and TLS(SRTP)?  By default, VoIP calls forwarded over AVOXI's network have been done so through a UDP (User Datagram Protocol) connection. This is the industry standard and the connection type most commonly recommended for VoIP users. However - due to personal preference, familiarity, local restrictions, or business need – some customers would like to have calls forwarded using a TCP (Transmission Control Protocol).  Learn more here! 

  • Does AVOXI support TLS?  Yes, if configured, both TLS and SRTP are used to encrypt calls between you and AVOXI. TLS/SRTP can be used for both inbound and outbound voice services (Originating and Terminating).  TLS, or transport layer security, protocol is the top and most powerful layer responsible for securing SIP voice and media messages. This protocol uses cryptographic encryption to provide end-to-end security. TLS is best for encryption, authentication, data integrity, and secure SIP trunking in general.  The Secure Real-time Transport Protocol (SRTP) is a security framework that extends the Real-time Transport Protocol (RTP). It’s mainly intended to be used in VoIP communications to secure the actual media – the little 'packets' of data that run over the highway set up by the signaling. 

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November 3rd, 2021


Louise Ross

Updated By:

Louise Ross



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