Creating a New SIP Trunk

A video guide on how to create a SIP trunk video guide can be found here.

Step 1. From the “Trunks” option in the Management Portal, select the “Voice Trunks” subcategory and an “Inbound” voice trunk profile. Click the “+Add new” button and select “SIP Trunk” (Fig. 1).


Fig. 1. Adding a new SIP Trunk.

Step 2. You will be prompted to enter your SIP Trunk details (Fig. 2). Basic configuration options are as follows:

Friendly name - a desired trunk name.

Username - the user part of R-URI in INVITE request. You also may use “{DID}” pattern which will be replaced by the called DID number (DNIS) in E164 format.

Host - your server’s public IP address or domain

Port - SIP port on your server. In order to use DNS SRV failover mechanism leave port empty. In case of empty value DIDWW system will try to resolve DNS SRV records, then A record will be tried(with default SIP signaling port 5060)

Preferred server - the preferred DIDWW POP server, with the options being US: LA, MIA, NY, Germany: FRA, SG (Singapore) or Local.

Protocol- the underlying transport layer protocol (UDP or TCP) that will be responsible for SIP requests and responses.

Capacity Limit - the maximum number of simultaneous calls for the trunk.

Resolve rURI - if checked, host part or R-URI will be replaced by the resolved IP address.

Resolve DNS SRV record - if checked, the system will attempt to perform a DNS SRV lookup for the host part of the R-URI to override the port settings.

CLI Format - Following format options are available:

  • “raw” - DIDWW will pass the incoming caller ID unchanged (default).

  • “e164” - attempt to convert the CLI to E.164 format (Country Code + Area Code + Number)

  • “local” - attempt to convert the CLI to local format (Area Code + Number)


Note that CLI format conversion may not work correctly for phone calls originating from outside the country of the specific DID

CLI Prefix - a field that may prefix the CLI with an optional ‘+’ sign, followed by up to 6 characters (digits and ‘#’)

Map all DID(s) - if checked, then all DIDs in your account will be assigned to this inbound trunk.

In addition to the basic configuration options, there are advanced options which may be left unchanged, or recommended values can be set by pressing “Use recommended values” button. These options include:

  • Auth enabled - if checked, then authorization for the SIP server is enabled

  • Auth user - optional authorization user for the SIP server

  • Auth password - optional authorization password for the SIP server

  • Auth “From” User - Specify user in a “from:” field instead of the Caller ID

  • Auth “From” Domain - Set the default “From” domain in SIP messages

  • Media & DTMF - Optional signalling configuration with parameters:

  • Available codecs/Selected codecs - codecs that will be sent with an SDP offer may be selected from the list of supported codecs

  • Customer To DIDWW DTMF format - the method of receiving DTMF signals from the CPE (options are RFC 2833, SIP INFO application/dtmf-relay OR application/dtmf, or RFC 2833 OR SIP INFO)

  • DIDWW to Customer DTMF format - the method of sending DTMF signals to the CPE CPE (options are disable sending, RFC 2833, SIP INFO application/dtmf-relay, or SIP INFO application/dtmf)

  • RTP Timeout - timeout (seconds) for disconnecting the call if RTP packets are not received

  • RTP Ping - if checked, an RTP ping should be used for connecting the call

  • Force Symmetric RTP - if checked, the trunk will operate in the Symmetric RTP/COMEDIA mode

  • Symmetric RTP Ignore RTCP - if checked, then only RTP packets will be considered while operating in the Symmetric RTP/COMEDIA mode, and switching of the RTP session based on RTCP packets will be avoided

  • Advanced Signalling Settings - Optional SIP configuration parameters as follows:

  • SST enabled - (RFC 4028) if checked, then the customization of SIP Session Timers is enabled in order to ensure that a session remains alive

  • SST Accept 501 - if checked, then the call will not be dropped after receiving SIP 501 response for non-critical messages

  • SST MIN Timer - (RFC 4028) minimum SIP session timer value (default 600 seconds)

  • SST MAX Timer - (RFC 4028) maximum SIP session timer value (default 900 seconds)

  • SST Session expires - (RFC 4028) Optional ‘Session-Expires’ header values (should be in range SST MIN Timer to SST MAX Timer)

  • SST Refresh method - (RFC 4028) SIP method that is used for session update (options are Invite, Update and Update Fallback Invite)

  • SIP Timer-B - (RFC 3261, Section INVITE transaction timeout (default 8000 ms)

  • DNS SRV failover timer - Invite transaction timeout for each of the gateways with DNS SRV rerouting (default 2000 ms)

  • Max Transfers - maximum REFER Request count

  • Max 30x Redirects - maximum 301/302 SIP Redirect count


Fig. 2. SIP Trunk configuration details window.

Step 3. Click “Submit” to complete the trunk configuration.