A SIP trunk is a virtual connection that delivers inbound voice calls from the public telephone network (PSTN) to an IP-based phone system over the internet.
CLI Format – Select how the Caller ID (CLI) is formatted for incoming calls:
Option
Description
raw
Passes the caller ID unchanged (default).
E.164
Converts the CLI to E.164 format (Country Code + Area Code + Number).
local
Converts the CLI to local format (Area Code + Number).
CLI Prefix – Prepend a custom prefix to the incoming CLI for identification or routing purposes.
CLI Number List – Assign a Number List to allow or reject incoming calls based on full number matches, prefix matches, or length restrictions. For more information, see the Number List documentation page.
Warning
CLI format conversion may not work correctly for calls originating from outside the country of the DID.
Number Lists work by matching the inbound CLI. If you modify the CLI format and CLI prefix, you may need to adjust your numbers in the Number List accordingly.
The routing method determines how DIDWW delivers calls to your endpoint.
Use Static SIP URI when your SIP endpoint has a fixed, reachable address. DIDWW always delivers calls
to a single configured SIP URI, built from User part of R-URI, Host, and Port.
User part of R-URI – Define the user part of the R-URI in the INVITE request.
Placeholder Variables
{DID} – Inserts the called DID in E.164.
{CALL_CPC} – Calling party category. See CPC usage.
Host – Host part of the R-URI in the SIP INVITE request. This can be an IP address or a domain.
Port – Port part of the R-URI in the SIP INVITE request.
Port Auto-Resolution Behavior
If Port is left empty, DIDWW first attempts an SRV record lookup.
If no SRV record is found, it falls back to resolving the A record.
Resolve rURI – Replace the host part of R-URI with its resolved DNS address.
Preferred Server – Choose the DIDWW Point of Presence (POP) for routing:
Auto (recommended): Let DIDWW select the optimal SBC dynamically
Ensure all POPs are enabled in settings. Disabling any POP may cause unnecessary routing hops.
Routing Examples by Preferred Server
The following figures illustrate how incoming calls are routed based on the selected preferred server:
Preferred Server: FRA - Calls from the PSTN network reach the DIDWW HK SBC and are routed to the FRA POP.
Fig. 3.1. Routing example with FRA as the preferred server.
Preferred Server: Auto - Calls are dynamically routed from the same DIDWW SBC that received them.
Fig. 3.2. Routing example with Auto as the preferred server.
Use SIP Registration when your PBX or SBC has a dynamic IP address, operates behind NAT, or when you prefer registration-based routing. DIDWW sends incoming calls to the Contact address learned during the SIP REGISTER process.
Enable SIP Registration – Activates dynamic call destination based on Contact information obtained during the SIP REGISTER sequence.
Use DID in R-URI – Replaces the user part of the R-URI in the SIP INVITE with the DID number, instead of the user part received in the Contact header during registration.
Important
Enabling SIP Registration disables the Host, Port, and User part of R-URI fields. Leave these fields empty.
A maximum of 10 simultaneous registrations are supported per trunk.
View your registration username and password by selecting the key icon in the Credentials column of the Inbound Trunks table. For detailed steps, see how to view credentials.
SIP Trunk group configuration is optional and should only be used when multiple trunks need to be assigned to a single trunk group for failover or load balancing.
Authentication is an optional SIP Trunk setting that allows the trunk to be authenticated using digest authentication (credentials). Use this feature only if your SIP server requires authentication.
The Media & DTMF section allows you to configure codec preferences, Dual-Tone Multi-Frequency (DTMF) signaling, and real-time transport protocol (RTP) settings.
The Advanced Signaling Settings section allows you to configure SIP session management, transaction timeouts, failover behavior, and diversion headers settings. These optional settings help maintain call stability, prevent unnecessary call drops, and optimize failover handling in case of network failures.
Session Timers ensure that SIP sessions remain active by periodically exchanging messages. If a session remains idle for too long and the other endpoint is unresponsive, the dialog is automatically terminated. For more details, refer to RFC 4028 .
Setting
Description
SST Enabled
Activates SIP Session Timers. Ensures that SIP dialogs remain active by sending periodic keep-alive messages.
This feature requires support from both endpoints and is standardized by the IETF.
SST Accept 501
Prevents call drops when receiving a SIP 501 response for non-critical messages.
SST MIN Timer
Sets the minimum session timer value (default: 600 seconds).
SST MAX Timer
Sets the maximum session timer value (default: 900 seconds).
SST Session Expires
Defines the Session-Expires header value (must be within the MIN and MAX Timer range).
SST Refresh Method
Specifies the SIP method for session updates: INVITE, UPDATE, UPDATE (Fallback to INVITE).
The Diversion Headers Settings section allows you to configure how diversion headers are handled for calls received on the inbound SIP trunk. These headers are used to indicate call forwarding or redirection events prior to reaching the trunk and may be required by downstream systems for routing, billing, or caller identity purposes.
Setting
Description
Diversion Relay Policy
Specifies how DIDWW should handle Diversion headers if they are received from the remote party. Options include:
Do not relay – Remove the Diversion header from outgoing SIP messages.
Relay as SIP URI – Format and relay the Diversion header as a SIP URI (e.g., sip:user@sip.didww.com).
Relay as TEL URI – Format and relay the Diversion header as a TEL URI (e.g., tel:+123456789).
Diversion Inject Mode
Controls whether DIDWW inserts a Diversion header in SIP messages. Options include:
Do not add Diversion – A Diversion header will not be generated or included by DIDWW.
Add Diversion as +DID number – A DIDWW-generated Diversion header will be added using the associated +DID number as the value, formatted in E.164 (e.g., tel:+123456789).
Note
If both Diversion Relay Policy and Diversion Inject Mode are enabled at the same time, two Diversion headers may be included in the SIP message, which may cause unexpected behavior on your system.
The STIR/SHAKEN framework helps prevent caller ID spoofing by verifying the authenticity of the calling number. This section allows you to configure caller ID attestation headers.
Adds verification status and attestation headers to the SIP INVITE request.
This option is supported by default and does not require enabling Identity Header Transit.
Note
The Transit Identity Header is not enabled by default. To enable it for your DIDWW account or from the originating side, contact our sales team at sales@didww.com .
If your inbound SIP trunk uses SIP Registration for dynamic routing, the system automatically generates a unique set of credentials. You can view these credentials by clicking the key icon in the Credentials column on the Inbound Trunks page.
The unique, system-generated username required for your SIP endpoint to register with DIDWW.
Password
The password required for registration. Select the eye icon to view the password in clear text.
Host
The DIDWW SIP server hostname your SIP endpoint must register to (e.g., sip.didww.com). For more details, see General SIP Information.
Use DID in R-URI
Indicates the status of the Use DID in R-URI setting for this trunk. If Enabled, the called DID number will be used as the R-URI user part during call delivery.
The Registration History is available only for trunks that use SIP Registration. It displays recent successful registration intervals for the selected trunk.
Follow these steps to view the Registration History:
Go to the Voice section in the menu.
Select Inbound Trunks.
Find the trunk you want to inspect and open the Actions menu.
Click View Registration History.
Fig. 14. Opening the Registration History window.
The Registration History chart shows successful registration intervals, indicating when a valid Contact header was active for call routing.
Note
The chart shows the times when the trunk was successfully registered and online. Individual SIP REGISTER requests or failed attempts are not displayed.
Registration data is available for the last 7 days. If no successful registrations occurred in this period, no data will appear.
Registration activity is shown in one-minute intervals.
Hover over any point in the chart to view details about that registration interval, including:
Field
Description
Contact
The Contact address used by your system during registration.
IP/Port
The source IP address and port used for the registration.
Transport Protocol
The transport protocol used for the registration (UDP, TCP, or TLS).
User Agent
The User-Agent string reported by the registering system.
Expires
The registration lifetime (Expires value) provided by your endpoint.
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