Termination Gateways

Termination Gateways are used to call out from phone.systems™ using created SIP accounts or to forward calls to PSTN destinations. A third-party party termination provider must be added and configured under the Gateways tab section.

To access and manage your termination gateways, click Trunks in the sidebar menu and select the Termination Gateways tab.

Fig. 1. Termination Gateways

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Adding and Configuring Termination Gateways

Step 1. To add a new termination gateway, click on the button. Alternatively, click on the Edit button to modify a previously configured gateway.

Fig. 1. Editing And Creating Termination Gateways

Step 2: Input the General Settings:

Setting	Description
Name	A friendly name for this gateway.
Network Protocol	Specifies the network protocol configuration: IPv4 Only: Only IPv4 addresses are supported. IPv6 Only: Only IPv6 addresses are supported. Dualstack: Supports both IPv4 and IPv6. IPv4 Preferred: IPv4 is preferred, but IPv6 can be used if needed. IPv6 Preferred: IPv6 is preferred, but IPv4 can be used if needed.
Username	The username used to authenticate the gateway.
Password	The password used for authenticating the gateway.
Host	The IP address or domain name of the gateway.
Port	The port used on this gateway. Defaults to 5060 if not provided.
Allowed Codecs	Multiple codecs may be selected for call negotiation. Options include: OPUS: Audio codec used for high-quality voice. G722: Wideband codec for better voice quality. PCMU: G.711 µ-law codec for standard voice quality. PCMA: G.711 A-law codec, similar to PCMU but used in different regions. G729: Low-bitrate codec for voice, often used for bandwidth-constrained networks. GSM: Codec used for mobile network calls. telephone-event: Codec used for DTMF (Dual-Tone Multi-Frequency) signaling.

Note

the codec priority may be arranged by dragging the listed codecs into the required position, with the left-most codec being of the highest priority.

Important

To use DTMF codes for events such as Interactive Menu options or Feature Codes, ensure that the telephone-event codec is included in the Allowed Codecs list.

Fig. 2. General Settings

Step 3: Configure CLI Rules (Optional):

The CLI Rules are used to automatically modify the source and destination numbers sent to the termination gateway, allowing users to add and remove prefixes or completely rewrite the numbers on a per-gateway basis.

• SRC Rewrite Rule/Result (optional): Modifies the source number that phone.systems™ sends to the provider’s termination gateway.

• DST Rewrite Rule/Result (optional): Modifies the calling destination number that is sent to the provider’s gateway.

Fig. 3. CLI Rules

If no changes are required to be made to the numbers when making outbound calls, then the Rule and Result fields should be left blank.

Note

Knowledge in using POSIX Regular Expressions is required in order to use this feature.

Step 4: Click Save to finalize and create or edit your termination gateways.