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About SIP Trunk

If you want to get the benefits of VoIP but your PBX is not IP capable, you can use a SIP trunk to connect to IP Telecom. To connect to IP Telecom, a gateway device sits between your phone system and the internet and uses a SIP trunk over your broadband connection.


When you use IP Telecom VoIP service with a SIP trunk connection, you do not have access to our hosted PBX system and applications.

Registering a SIP trunk

Use the following settings to establish SIP trunk peer connection.

Parameter Value
Username {yoursipusername} the username of the SIP trunk
Secret {yoursipsecret}
Type peer
Insecure very
From domain
From user {yoursipusername}
Codec g711a
Dtmf mode RFC 2833
Fraud protection enabled


Use these parameters for "peer definitions" in asterisk config files.

Additional instructions

  1. To send caller ID, send us the PAI (p-asserted-identity) header. We don’t support any other mechanism at this time.
  2. For details on PAI for asterisk and other relevant PBX, see
  3. Disable reinvites.

Encrypted SIP trunking/encrypted SIP messaging with secure RTP

To avail of encryption support on the SIP trunk:

  • The SIP Trunk endpoint must use port 5061 for signalling and TLS as the transport.
  • After that is set up, you can choose to encrypt the media using SRTP on your PBX.
  • You will get offers of encrypted media on inbound calls.
  • You can opt to do Transport over TLS (SIP messaging) only, and enable or disable SRTP.


Check with your account manager about the costs associated with using this feature.

More information