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SIP Trunks Overview

A SIP trunk allows users to connect an external, self-hosted, self-maintained phone system. Using a SIP trunk is not recommended as it does not allow users to take advantage of existing features on a site and the associated hardware is typically prone to being compromised by external entities.

A SIP trunk requires a name used to identify it on the system, a login and the password used in SIP registration and call authentication.

An optional field is provided for Contact IP address, which takes the form of an IP address and a port (e.g '63.211.239.14:5060'). When set, the system will bypass any registration and send all inbound calls to the provided address and port. The default SIP port is 5060. The port is required in order for this to take place.

INVITEs sent to SIP trunks will be sent with DNIS (Dialed Number Information Service) in the format 1NPANXXXXXX. This cannot be changed.

Additional features available for SIP trunks include timeout and failover destination, unique Caller ID, and a specific Access Control List.

How to Add a New SIP Trunk

SIP Trunk Edit Form

  1. Enter a Name that will uniquely identify the SIP trunk from all the other SIP Trunk records in the SIP Trunks table (e.g. 'John Doe Inc. PBX')
  2. Enter a login name (e.g. 'JohnDoeInc')
  3. The automatically generated password information is available in this section. If you would like to change the password given, select the Regenerate Password button.
  4. Channel limiting SIP trunks can have up to 100 channels. A channel is a "line" that your phone uses to place or receive a call. Typically, one call requires one channel.
  5. Enter a timeout duration. A setting between 1 to 120 seconds that specifies how long the SIP trunk will ring before transferring the caller to a specified failover destination or, if no failover is assigned, changing up the call.
  6. Select the status of the disable comfort noise toggle. If your hardware does not support RFC3389 Comfort Noise (RTP AVP 13 CN 8000), this can be disabled using the "Disable comfort noise" toggle (switch toggle button to off).
  7. Failover specifies where callers will be directed if the SIP trunk does not answer within the timeout period. This can be any other routable destination like a menu or a voicemail box. If this a failover is not assigned, callers reaching the timeout will be disconnected.
  8. Enter a Caller ID Name (e.g. 'John Doe Inc.') Enter a specific name to be set and displayed on outbound calls.
  9. Enter a Caller ID Number (e.g. '17202345678') Enter a specific number to be set and displayed on outbound calls.
  10. The Access Control List (ACLs) can specify IPv4 address(es) or CIDR-notated subnets that override the default ACL. If this is not set, it is inherited from the instance's ACL specifications.

Allowed Codecs

PCMU (G.711u), PCMA (G.711a), G.722, and GSM which are all recommended at 20ms ptime.

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