Genius SIP Trunking in AVOXI (Outbound Termination / Making Calls)
Easily integrate your current in-house PBX or third-party platform (example: MS Teams) with your AVOXI Genius online platform and instantly enable outbound calling to any destination worldwide. Available to all AVOXI Genius users at monthly recurring charge of $19.99, SIP Trunking is a popular solution for businesses looking to cut local and long-distance calling costs without having to replace an existing system.
Use the below quick links to view, manage and configure SIP Trunking within your AVOXI Genius online portal.
Viewing SIP Trunk(s)
- To get started, log into the AVOXI Genius platform at https://genius.avoxi.com
- Select the "SIP Trunks" section on the left-hand navigation bar.
- Click the "SIP Trunks" tab located next to SIP URI's
On the SIP Trunks home screen, you can view:
- Trunks: A list of all your existing trunks that you can manage or delete.
- Authentication: Authentication type and information of configured trunks
- Direction: The direction of the trunk, i.e., "Termination."
- Outbound Rates: All new Sip Trunks configured via AVOXI online platform will be billed the monthly recurring charge of $19.99. Outbound calls will be charged per minute for usage according to your outbound rates. Use the link to download and view the application rates.
Adding a SIP Trunk
Use the below step-by-step guide to add and configure your SIP Trunks
- To get started navigate to the SIP Trunk section located on the left-hand navigation bar.
- Select the "SIP Trunks" tab
- Click the "Add SIP Trunk" button
Name your SIP Trunk
Give your SIP Trunk a friendly name using letters, spaces, and numbers. This is the name that will appear under the "Usage" section in the "Account ID" column of your monthly invoice.
- Special characters are not accepted
- Max name length allowed length is 116 characters
- Select how you would like to terminate your outgoing traffic (for example to MS Teams or Regular SIP Trunk).
Method of authentication to ensure outbound calls through the SIP Trunk can be uniquely authorized.
- Enter your Username and Password
Ensure that the username matches the From header user or the To header user
- Enter one or more required IP Address(es)
- If this is a shared platform, please provide additional information for authentication, such as the SIP Header or username/secret on top of the IP address.
Not everyone can support Basic Authentication from a shared platform. If that's the case for you, then you can use SIP Header Authentication. This means you can add a special header and value to the message to prove the message is yours.
- Header Name: Add special header name for SIP invite
- Header Value: Add the special header value for the SIP invite
- Protocol: Select your ideal transport protocol
SIP Invite Message Examples: Consider the following SIP message as an example
INVITE sip:+email@example.com SIP/2.0
CSEQ: 1 INVITE
From: “Alice” <sip:+ firstname.lastname@example.org>
To: Bob <sip:+email@example.com>
Date: Wed, 14 Jul 2021 17:58:47 GMT
P-Asserted-Identity: “Alice” <sip:+firstname.lastname@example.org>
Contact: “+14049375037” <sip:+email@example.com;transport=udp>
User-Agent: Awesome Customer PB
*Note: Please ensure you send the '+' and country code in the From and To fields
Header Authentication: Below is a list of headers, we do not check for header authentication
When ready, select the "Add New SIP Trunk" button to complete.
Now that you have added your SIP Trunk(s) to your AVOXI Genius account, it is important to configure it on your communications infrastructure/platform.
We have a Genius US POP and a Genius HK POP. Here are the DNS Host names along w/ IP Addresses.
- DNS Host name: trunk-production-us1.avoxi.com
- IP Addresses: 220.127.116.11 and 18.104.22.168
Hong Kong POP:
- DNS Host name: trunk-production-hk1.avoxi.com
- IP Addresses: 22.214.171.124 and 126.96.36.199
AVOXI supports UDP, TCP, and TLS(SRTP) version 1.2 (follow here to learn more). Open the following ports to ensure proper connectivity:
- 5060 UDP
- 5060 TCP
- 5061 TLS
AVOXI supports the below Codecs
RTP port range for AVOXI Genius
Port 10000-65535 UDP for both US and Hong Kong Media.
- 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).
- As an AVOXI Administrator you are required to download and manually load the certificate during the configuration process to ensure the TLS connection are made (example sip.avoxi.com:5061).
- Whitelist IP - Media
Configuring Third-Party Platforms
Use the below quick links to view the relevant configuration guide:
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 Genius 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."
Calls will only record if the call recording feature has been enabled at an organizational level. As an Administrator, you can easily manage your business call recording settings within your online platform. Use the following self-help guide to turn your recording on/off and manage retention settings. When deleting a SIP Trunk, this will not delete the call recordings.
View our The Beginner's Guide to SIP Trunking and SBC Solutions to learn more about SIP Trunking.
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