Skip to content

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.

Important

For SIP trunk, all numbers must be sent to us in full e164 format, for partners in South Africa this means all numbers should include the leading +27. We would recommend that your remote system should have a dialplan defined that converts numbers starting 0[1-9]XXXXXXX into +27[1-9]XXXXXXXX before sending them to us.

Registering a SIP trunk

Use the following settings to establish SIP trunk peer connection.

Parameter Value
Host sip.iptel.co
or
sip.iptelecom.co.za (South Africa)
Username {yoursipusername} the username of the SIP trunk
Secret {yoursipsecret}
Type peer
Insecure very
From domain sip.iptel.co
or
sip.iptelecom.co.za (South Africa)
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

  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 https://ietf.org/rfc/rfc3325.txt.
  3. Disable reinvites.

SIP minimum network requirements

  • 195.191.28.50 and 195.191.29.50 on UDP ports 5060 and 7000
  • 195.191.28.104/29 and 195.191.29.104/29 on UDP ports between 20480 and 24576

Monitoring:

  • ICMP echo-request and echo-reply from 195.191.29.21 on the public facing interface of your router/firewall

IP Telecom requires that the traffic in your network should be open to and from our network.

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.

Sample SIP invite

A SIP invite is sent to set up a VoIP call. The main fields that are included in a SIP INVITE request are shown. A SIP INVITE message usually has between 4 and 6 header entries with contact information inside them. Different devices or providers use these headers in different ways and an understanding of the main SIP fields can help with troubleshooting.

    INVITE sip:0001234567@191.164.100.XXX:5060 SIP/2.0
    Record-Route: <sip:195.XXX.29.50;lr;ftag=4cD6vm83N0XXX;natedua=Y;cudg=153.5684fXXX>
    Via: SIP/2.0/UDP 195.XXX.29.50:5060;branch=z8hG4bKb7a4.d0440a3.0
    Via: SIP/2.0/UDP195.XXX.29.30:5080;received=195.XXX.29.30;rport=5080;branch=z9hA4bK56XN0UFXXXXrg
    Max-Forwards: 69
    From: "000012345432" <sip:00441257473445@195.191.29.30>;tag=4cB6vm83XXXXX
    To: <sip:0214772107@195.191.28.50:5060>
    Call-ID: 1234b33b-bd8c-123a-da94-00900b123456
    CSeq: 12345678 INVITE
  • Request-Line-URI (RURI): The INVITE header field includes the destination of the call. It contains the same information as the "To" field, omitting the display name.
  • Record-Route: Optional header field inserted into requests by proxies that wanted to be in the path of subsequent requests for the same call-id. It is then used by the user agent to route subsequent requests.
  • Via: Every proxy in the request path adds the address and port on which it received the message to the top of "Via", then forwards it onwards. When processing responses, each proxy in the return path processes the contents of the “Via” field in reverse order, removing its address from the top.

  • Max-Forwards: The "Max-Forwards" header limits the number of times a request can be forwarded on its way to the recipient. The number is reduced by one each time the request is forwarded. The Max-Forwards header prevents a request from endlessly circling the SIP network if the recipient cannot be found. The default value is 70.

  • From: The “From” header field indicates the identity of the initiator of the request from the point of view of the PBX Server – similar in construction to email addresses (user@domain – where “user” is, for example, the extension number, and “domain” is the server domain or IP address). Like the “To” header field, it contains a URI and optionally a display name. It is used to determine which processing rules to apply to a request.

  • To: The “To” header field first and foremost specifies the desired “logical” recipient of the request, or the address-of-record of the user or resource that is the target of this request. This may or may not be the ultimate recipient of the request.

  • Call-ID: The Call-ID header field acts as a unique identifier to group together a series of messages. It MUST be the same for all requests and responses sent by either UA (user-agent) in a dialog. It SHOULD be the same in each registration from a UA.

  • CSeq: The CSeq header field serves as a way to identify and order transactions. It consists of a sequence number and a method. The method MUST match that of the request. For non-REGISTER requests outside of a dialog, the sequence number value is arbitrary. The sequence number value MUST be expressible as a 32-bit unsigned integer and MUST be less than 2**31. As long as it follows the above guidelines, a client may use any mechanism it would like to select CSeq header field values.

Source RFC3261 SIP: Session Initiation Protocol

Set up divert on SIP trunk phones

To divert numbers via handset

  1. Dial 1821 and follow the prompts.

    • 2 to "Always Divert"
    • 3 to "Divert when Unavailable"
    • 4 to "Cancel"

  2. For option 2 or 3, enter the number to divert to then hit # to follow the prompts; the system will respond back the number it's about to divert.

    • Press Option 1 to confirm, the system will confirm all is ok, or give an error
    • Press Option 2 to cancel, system will respond ok, and hangup

  3. For option 4 - system will confirm the number that's about to be cancelled.

    • Press 1 to cancel the divert or
    • Press 2 to cancel the operation

You can also dial 1821 to divert calls to a number instantly.

More information