How SIP Trunking works with Thinktel?

This article explains how SIP Trunking service is provided by ThinkTel.

The following are the basic settings required to start a SIP trunk interop with ThinkTel.

Our main switch is located in Edmonton: Metaswitch -

Signaling and RTP Information by Proxy


Signaling - If using any of these proxies you will need to allow UDP 5060 signaling as appropriate:

  • (     Primarily assigned to Western Canada based PBX trunks - accepts NAT and public IP
  •    ( Primarily assigned to Eastern Canada based PBX trunks - accepts NAT and public IP
  •  (   Legacy proxy for public IP PBX trunks

RTP - If using any of these proxies you will need to allow RTP UDP for ports 10000 through 65500 as appropriate:

  •,,,,,, (Added Feb 2018), (Added Jun 2018), (Added May 2020)


Signaling - For proxies you will need to allow TCP 5060 signaling as appropriate:

  • (   Western Canada
  •   ( Eastern Canada 

RTP - For proxies you will need to allow RTP UDP for ports 49152 through 57500 as appropriate. You must allow all outbound UDP traffic to us on ports 10000 through 65535.


For more information on the required ports your mediation server must have published please see the following Microsoft documentation.

  • We identify each SIP trunk with a pilot number, a 10-digit number local to your VoIP gateway, similar to how Telco's assign pilot numbers to conventional PRI's. Please refer to your pilot number when placing orders for DIDs, channels, etc. The requested number of channels for a SIP trunk is activated once the initial interop stage is complete and your account is active with ThinkTel. This can also be done via uControl.

  • Ensure that your PBX or gateway supports DNS SRV records.

  • By default, we offer the following codecs: G711, G729 (renegotiation to T.38 is accepted from g711 when supported for Fax over IP). Other codecs might be available upon request.

  • Configure your gateway to send and receive 10 digit North American numbers. 11 digit is available upon request.

  • Configure DTMF for RFC2833 (out of band).

  • If you are running Asterisk, TrixBox or OpenPBX, please ensure that canreinvite=no is set in the SIP settings.

  • Disable any SIP ALG service on routers or firewalls as it is not compatible with our services. If enabled, you may experience registration or audio related issues that will negatively impact your ability to use our service.

  • Our switch sends out a SIP keep-alive OPTIONS request to the contact IP for your SIP trunk every 30 seconds. This is a standard SIP OPTIONS request that must be responded to with any valid SIP response. If no response is received within 30 seconds of the request, the switch will send a request every 4 seconds, disable the SIP trunk binding associated with that particular IP and will not send further SIP invites until a response is received. This negates the need for REGISTER and registration should be disabled whenever possible.

  • We authenticate calls by both methods:

    •  IP address: the VIA and Contact header must contain your contact IP address and port.

    •  SIP credentials: the supplied username and password credentials must be sent in the authentication digest for all SIP requests that require authorization (INVITE, UPDATE, BYE, etc).

  • We allow up to 8 IP addresses per SIP trunk.

    • Each IP address/port combination creates a "binding" that is assigned to your SIP trunk.

    • Assigning the same binding to multiple SIP trunks can be accomplished with the following restriction:

    •  Only one trunk is allowed to manage changing the caller ID with the use of a Custom Calling Party ID number.
      o Example: If 3 different SIP trunks all use the same binding, only one of the SIP trunks will be configured to allow an arbitrary calling party number. That SIP trunk and associated charge number (Pilot) will be billed for all calls that have a Custom Calling Party caller ID.

  • Each binding/IP assigned to a SIP trunk is used for origination and termination.
    • The following options are available to route calls to your binding(s) / IP address(es):
      • Round Robin: Incoming calls are sent to each binding assigned in your SIP trunk in turn. If the SIP response from your gateway is > 303 (404/Not Found or 503/Unavailable), our switch will send an INVITE to the next IP address until the INVITE is accepted with a SIP response < 303.

        o Lowest Available: Use the preferred binding for as long as the trunk/binding will permit it and move on to the next in the list if the ones before are full/unavailable.
  • You can configure an Unavailable Call Forward for the SIP trunk as well as for individual DIDs. If all your bindings are down (not responding to our OPTION requests) calls will be forwarded to the defined Call Forward. The full procedure is documented here.