Integrate third-party SIP trunks

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This topic describes how to integrate third-party SIP trunks with Smart Outbound Calling.

Prerequisites

  • You have enabled Smart Outbound Calling.

  • You have agreed on the integration fees with your Alibaba Cloud business representative.

SIP infrastructure requirements

  • Your equipment must fully support the SIP protocol and loose routing.

  • You must ensure the SIP protocol transparently transmits the original caller and callee numbers.

  • Your equipment must have a static public IP address.

  • You need qualified SIP operations engineers for the integration and subsequent maintenance.

  • You are responsible for the high availability of your equipment. You can provide multiple public IP addresses for this purpose.

  • Enable the OPTIONS probing mechanism. If a Smart Outbound Calling server fails to respond to an OPTIONS request, your system must remove that server from the inbound load to prevent service disruption for new calls.

SIP integration procedure

1. Provide SIP service information

Contact Alibaba Cloud Smart Outbound Calling operations support and provide your SIP service information.

2. Configure the SIP gateway

Configure your SIP gateway with the Smart Outbound Calling SIP service information.

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Use the following information to configure your connection to our SIP service:

  • Your SIP gateway must connect to the following Smart Outbound Calling SIP service address over UDP:

    • IP address: Contact Alibaba Cloud Smart Outbound Calling operations support for the SIP server IP addresses. This includes the signaling address and the media address.

    • Port: 8000 (public port)

Important

You may need to add our signaling address and media address to your allowlist.

  • You must ensure that the specified protocol, addresses, and port are accessible over the public internet.

Note

Gateway signaling must be routed back to the source IP address of the SBC from which the request was received.

  • The Request-URI of a BYE message must match the Contact header from the initial INVITE message, which typically contains a FreeSWITCH server address on port 5080.

    • Figure 1. INVITE message

    • Figure 2. BYE message returned from your gateway. For an ACK that responds to a 200 OK, the Request-URI must match the Contact header from that 200 OK message (which typically contains a FreeSWITCH server address on port 5080).

    • Figure 3. 200 OK message

    • Figure 4. ACK response to a 200 OK message

  • Your SIP gateway must support loose routing.

3. Configure instance and initiate test call

  1. Create an outbound instance.

  2. On the Number Management page, bind numbers to the outbound instance.

  3. Initiate an outbound task to test the connection.