SIP Forwarding

This SIP Forwarding section of the Contact Method menu in the Cloud PBX phone.systems™ interface allows you to perform the following tasks:

  • Create SIP Forwarding Contact Methods.

  • Edit SIP Forwarding Contact Methods.

  • Relations of SIP Forwarding Contact Methods.

  • Delete SIP Forwarding Contact Methods.

  • Download SIP Forwarding Contact Methods To A CSV File.

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Fig. 1. SIP Forwarding Contact Methods

Creating SIP Forwarding Contact Methods

To create a SIP Forwarding Contact Method, click the +-symbol symbol, which will open the SIP Forwarding Contact Method creation screen.

The SIP Forwarding configuration window contains two sections:

1. General

  • User option allows you to select the user for the Contact Method.

  • Username field requires the SIP username for authentication.

  • Domain field requires the domain name of the SIP server.

  • Port field allows you to specify the port number used for the SIP connection.

  • Network Protocol option allows you to choose the communication protocol.

  • Process 30X Redirects toggle enables processing 30X SIP redirects when turned on.

    Note

    Enabling “Process 30X Redirect” option may impact call routing based on SIP response codes.

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Fig. 1. General Configuration

2. Inbound Calls

The Inbound Calls section allows you to configure the DID and internal numbers associated with the SIP forwarding contact method, with the option to configure a fallback routing option to voicemail.

  • DID Number specifies which DID numbers will perform the SIP forwarding.

  • Internal Number specifies which internal numbers will perform the SIP forwarding.

  • When Unavailable option allows you to select how calls are forwarded when the SIP destination is unavailable. You may select:

    • Do Nothing: The call is ended if the SIP destination is not responding.

    • Route to Voicemail: The call is routed to the selected voicemail audio message.

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Fig. 2. Inbound Calls

3. Advanced

You can select the Codecs supported by this SIP account. Multiple codecs can be added from the dropdown list, with the following options available:

  • OPUS

  • G722

  • G729

  • PCMU

  • PCMA

  • telephone-event

Important

The telephone-event codec must be included in the list of Allowed Codecs if interactive menus or feature codes requiring digit input are to be used.

The Allowed Media Types field specifies the network protocols permitted for transporting media streams (typically audio) during SIP sessions. The following options are available:

  • RTP

  • SRTP-SDES

  • SRTP-DTLS

  • SRTP-ZRTP

The Default Media Type defines which media transport protocol will be preferred when multiple options are available. It should match one of the values set in the Allowed Media Types list. For example:

  • RTP

  • SRTP-SDES

  • SRTP-DTLS

  • SRTP-ZRTP

The Transport Protocol field allows you to select the protocol used for communication between phone.systems™ and the end user’s SIP device. Supported transport methods include:

  • UDP

  • TCP

  • TLS

Note

Encrypted media types and transport protocols are disabled by default on phone.systems™ accounts. Contact sales@didww.com to enable these features.

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Fig. 3. Advanced