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.
Note
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 |
|---|---|
| Host | sip.iptel.co |
| Username | {yoursipusername} the username of the SIP trunk |
| Secret | {yoursipsecret} |
| Type | peer |
| Insecure | very |
| From domain | sip.iptel.co |
| From user | {yoursipusername} |
| Codec | g711a |
| Dtmf mode | RFC 2833 |
| Fraud protection | enabled |
Note
Use these parameters for "peer definitions" in asterisk config files.
Additional instructions
- To send caller ID, send us the PAI (p-asserted-identity) header. We don’t support any other mechanism at this time.
- For details on PAI for asterisk and other relevant PBX, see https://ietf.org/rfc/rfc3325.txt.
- 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.
Important
Check with your account manager about the costs associated with using this feature.