SIP
Connect SIP PBX and SIP Trunking flows to Voximplant Voice AI scenarios
For the complete documentation index, see llms.txt.
Use SIP when you need to connect PBX infrastructure, enterprise SBCs, cloud telephony providers, or existing SIP trunks.
SIP Trunking is the common production pattern for routing calls between your telephony core and Voximplant-managed Voice AI scenarios.
Voximplant supports SIP calling in three distinct models:
If your SIP infrastructure uses a fixed public IP, add it to the SIP allow-list in Voximplant security settings. For firewall configuration, use the official Voximplant SIP/media IP sources below:
Use this when your PBX/trunk forwards calls by dialed number.
In your Voximplant application, create a routing rule whose pattern matches the inbound destination number format you expect - i.e. ^\\+1[2-9]\\d{9}$.
Attach your inbound Voice AI scenario to that rule. See Configure Voximplant - Create a scenario for details.
Route calls to:
number@application.account.voximplant.com
where number matches the routing-rule pattern, application is your Voximplant application name, and account is your Voximplant account name.
Alternatively, you can also use this format app_id#account_id#destination@sip.voximplant.com where destination must be a number
and app_id and account_id are the numeric ids found in your application settings.
Use this when each SIP endpoint should authenticate as a distinct user and route by username or a pattern based on usernames.
Create one Voximplant user per SIP endpoint (agent seat, extension, or softphone identity).
In manage.voximplant.com, go to Applications, open your app, then open Users.
Create one user per endpoint (for example: support01, support02) and set a strong password.
Keep this username/password pair because SIP clients will use it for registration.
Create routing rules that match the target username or alias and attach the relevant scenario. See Configure Voximplant - Create a scenario for details.
Login to this account on your SIP software using
user@application.account.voximplant.com
and the Voximplant user password.
You can specify UDP/TCP transport, or specify TLS transport to secure your connection.
Some SIP infrastructure may require that you specify the password and options in the address: sip:username:password@domain:5061;transport=tls
If you automate provisioning, you can also create users via the Management API: AddUser.
Use this when Voximplant should register to external SIP infrastructure (such as a PBX) and behave like a phone connected to that infrastructure.
In your PBX or SIP provider, create the extension / ID Voximplant will use.
In Voximplant settings, create a SIP registration with that extension’s domain, username, and password.
Attach the SIP registration to your application. This is available from the SIP Registrations tab in the left pane.
When you do this, you will need to select the desired routing rule that handles calls to this registration.
Note that you do not need to configure the rule pattern in this case — the rule works for all the calls to this SIP registration.
Open SIP Registrations in Control Panel and verify the registration shows as connected/active before testing calls.
You should see s status of Success if the registration is active.
You can also verify status via Management API:
GetSipRegistrations.
Please note that you pay for SIP registrations monthly according to our price list. So if you create a SIP registration and delete it immediately, your account is charged for the SIP registration monthly fee.
When a SIP registration status changes from Successful to Failed and vice versa, you get an email notification to the contact email addresses and the HTTP callback invokes. You can learn more about HTTP callbacks in this article.
Use VoxEngine.callSIP() for outbound SIP dialing.
Before placing outbound calls through a SIP registration, confirm the registration is active in Control Panel (SIP Registrations) or via
GetSipRegistrations.
To place an outbound SIP call through a registration, pass regId in the third argument of VoxEngine.callSIP(...).
You can find regId in the SIP registration list in Control Panel, or in
GetSipRegistrations and
CreateSipRegistration responses.
Usually you only need to specify the destination username/number of the called party.
For programmable outbound launching, start the routing rule from GUI or call startScenarios and pass custom data (for example destination/caller ID) to the scenario.
SIP Voice AI connector example:
