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.
Use the below guides and quick links to configure the SIP account all calls should be diverted too:
Add New SIP URI
- First, log into your AVOXI platform, and navigate to the SIP Trunks section located on the left-hand navigation bar:
- Click on the SIP URIs tab:
- Click on the ‘+ Add’ button to the right of the filter magnifying glass.
The SIP URI configuration tool consists of 3 configurable fields and 1 auto-generated preview field.
- Name: Meaningful name of the new SIP URI configuration
- Protocol: A transport protocol for your SIP URI configuration. AVOXI supports UDP, TCP, and TLS, for more information about the differences between these three protocols, please refer here!
URI: This is the configurable URI you will be using to set up your forwarding. There are three components.
- 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.
- 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.
- 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.
- Preview: This is a preview of your full SIP URI
- Click ‘Add New SIP URI’ to save your configuration. Click ‘Cancel’ to cancel out of the configuration tool.
How to edit an existing SIP URI
- Navigate to the SIP URI tab on the SIP Trunks page and click on the existing SIP URI configuration you would like to edit
- Once you are on the edit page, make your changes to either the rule, protocol, or URI portion of your configuration and click save once satisfied
How to delete an existing SIP URI
- First, navigate to the SIP URI tab on the SIP Trunks page and click on the ‘Delete’ button next to the SIP URI configuration you would like to delete.
- When you click ‘Delete’, you will get a confirmation message verifying your choice.
- Click ‘I understand, delete this SIP URI’ to remove it from the list.
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
- Click the edit button on the rule and navigate to the ‘Forward Calls To’ dropdown and select SIP and your preferred SIP URI
- 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
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 version1.3 (follow here to learn more). Open the following ports to ensure proper connectivity:
- 5060 UDP
- 5060 TCP
- 5061 TLS
AVOXI supports the below Codecs
SIGNALING IP ADDRESSES
- RTP port range for AVOXI 10000-65535
- SRTP- encrypted media is coming soon
RTP Audio Media Steam IP Addresses
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 SIP? Yes, 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 and TCP? 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, but only on the AVOXI Genius platform. TLS only allows SIP entities to authenticate servers to which they are adjacent. Establishing a TLS connection authenticates both transport endpoints but does not authenticate the SIP messages flowing through the link.
- What does AVOXI Genius not support? We do not support SRTP (encrypted media). We do however support encrypted signaling via TLS.