Genius SIP Forwarding in AVOXI (Inbound Forwarding)

SIP forwarding allows AVOXI Admins 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), 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: 

Add New SIP URI

  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 one 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 use 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. Your particular phone system uses the public domain name or IP address target. 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 the SIP URIs tab.
  3. Click on the "SIP URI Name" you wish to update. 
  4. Make changes to either the Protocol or URI portion of your configuration on the edit page.
  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 the 'Delete' button next to the SIP URI configuration you want to delete. 



  • 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. Like HTTPS, the secure web certificate is sent when a browser tries to connect using TLS. You can configure and manage the TLS/SRTP within your AVOXI online portal as a customer. Certificates are used for authentication and encryption.  

There is an SSL (TLS) certificate for trunk-production-us1.avoxi.com  used for calls from the user to the AVOXI Genius platform and peer.avoxi.io for calls leaving AVOXI Genius. As an AVOXI Administrator, you must manually load the certificate during the configuration process to ensure the TLS connection is made (example sip.avoxi.com:5061).  

Use the steps below to download the required certificate file from 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
  • Click on the "Download SIP URI TLS Certificate." 


Adding SIP URIs to Routing Rule Targets (Numbers)

Once you have set up your new SIP URI, you can 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 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 to the Numbers page to see that the target of that newly configured numbered is set to "Forward to SIP."


Bulk Update your SIP URIs Routing Rule Targets (Numbers)

With our Bulk Update functionality, users can effortlessly update the Call Forwarding rule target on multiple numbers with only a few clicks. Use the below steps to set a new or existing SIP URI as the target for your forwarding rules. 

  • Navigate to your organization's "Numbers Dashboard" using the numbers icon on the left-hand navigation bar.
  • On the "Active Numbers" tab, select the numbers you wish to update using the check box located to the left of the number.
  • Next, click the "three dots" located to the right of the "+ADD" button.
  • Select, Modify Forwarding Rules. 


  • Use the "Add New Rule" button to create a new routing rule. Alternatively, you can use the three lines next to the "Enabled" icon to drop down and select an existing preferred SIP URI. 


  • Once you are done selecting, click Save.
  • Navigate 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 configured SIP within your  AVOXI account; it is important to configure it on your communications infrastructure/platform.

TRANSPORT PROTOCOLS

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)

CODECS

AVOXI supports the below Codecs 

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

SIGNALING IP ADDRESSES US 

  • 104.196.177.203
  • 104.196.177.56

SIGNALING IP ADDRESSES Hong Kong

SIP Signaling (port 5060/5061 TCP/UDP) 

  • 34.92.127.143
  • 34.92.174.171
  • 34.96.219.253
  • 34.96.245.147

Audio (Random UDP ports)

  • 34.150.67.194
  • 34.150.89.83
  • 34.92.121.200
  • 34.92.127.48
  • 34.92.192.51
  • 34.92.230.37
  • 34.92.230.42
  • 34.92.48.216
  • 34.92.74.10
  • 34.92.83.126
  • 34.96.137.99
  • 35.220.131.218
  • 35.220.179.43
  • 35.220.179.55
  • 35.241.116.163
  • 35.241.87.227

TURN/ICE (port 80/443 TCP/UDP)

  • 34.96.206.113
  • 34.92.130.145


MEDIA 

  • RTP port range for AVOX10000-65535
  • SRTP
    • SSL (TLS) certificates for trunk-production-us1.avoxi.com are used for calls from the user to the AVOXI Genius platform, and peer.avoxi.io 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, sip.avoxi.com:5061)
  • RTP Audio Media Steam IP Addresses
    • 34.74.26.164
    • 34.75.61.225
    • 35.196.163.148
    • 34.74.74.172
    • 35.243.241.228
    • 35.196.156.208
    • 34.73.213.42
    • 34.75.227.126
    • 35.227.42.112
    • 34.75.97.143
    • 34.75.197.236
    • 34.75.49.119
    • 35.231.143.46
    • 35.196.59.183
    • 35.243.199.92
    • 35.196.97.117
    • 34.74.129.35
    • 34.75.42.140
    • 35.231.244.203
    • 35.243.176.215
    • 35.237.221.3
    • 34.75.17.11
    • 34.74.55.73
    • 104.196.165.15
    • 34.73.56.126
    • 35.237.210.244
    • 35.196.177.134
    • 35.237.150.40
    • 35.237.212.40
    • 35.227.70.158
    • 35.190.170.102
    • 35.185.23.34

WARNING: If you do not whitelist the designated IPs, you will receive an error message.

Configuring Third-Party Platforms 

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

Forwarding to SIP FAQs

  • Will all inbound calls be diverted to all the signed and active SIP devices and programs now? 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 using a SIP device, will the call be redirected - if so, to where? It depends on the timeout destination and forwarding rules configuration in Genius.


  • If the call is busy, will it 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 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 needs – 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. 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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Updated:

August 4th, 2022

Author:

Louise Ross

Updated By:

Max Germain

KB ID:

863432

Page Views:

6181

Tags:

forwarding to sip, sip forwarding

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