SIP Trunking Setup

  • Last updated on March 20, 2025 at 7:07 PM

IMPORTANT:  For first time SIP Trunk Configurations, we highly recommend requesting SIP Engineering support via a support ticket.

  • SkySwitch can assist with creating the trunk and troubleshooting incoming registration to our switch. 
  • While it is the partner's responsibility to register the trunk on their PBX/Endpoint, we are happy to provide Best Effort Support guidance for certain PBXs such as, but not limited to, 3CX, Grandstream UCM, and Asterisk. However, some brands have been deemed incompatible due to a lack of cooperation from the vendor (such as Unifi and Adtran).

REQUEST ASSISTANCE - Please complete this information in the form below. 

SIP Trunking Request Form

Looking for BYOT?  Please see this page for more information.

When you have a customer with an IP PBX or other SIP-enabled equipment, it is sometimes desirable to provision calling paths with no 'features' instead of a full PBX seat. Referred to as a SIP Trunk, this configuration is easier to create and maintain than a WebCentrex User and will allow your customer to make and receive calls using your service. You can limit the total number of Inbound, Outbound, and Total call sessions associated with the SIP Trunk.

Step-by-step guide

  1. In the PBX portal click on SIP Trunks in the top menu.
  2. Click the Add SIP Trunk button found on the upper right area and enter the SIP Trunk details in the Add SIP Trunk dialog.
    SettingDescription
    NameThis is the text string that will be used to match calls coming from the customer's IP PBX or SIP Device. It must be configured to match the hostname that will be sent in the SIP From: header by the customer's equipment.  See note below.
    DomainThis is the customer Domain that will be associated with this account.
    DescriptionAny description of the customer's equipment. It is helpful to put the type of equipment or software here (eg. Asterisk PBX).
    Relay MediaThis setting allows you to control whether the RTP traffic (voice payload) will be routed through the SkySwitch servers. If the IP PBX or SIP device has a static public IP address, then this should be set to 'No' as this often results in a better quality of service. If you are unsure, then select 'Don't Care' and the switch will determine the most appropriate setting.
    Trunk TypeSelect whether this customer should be allowed to make Inbound Only, Outbound Only, or Bidirectional calls.
    Record Trunk CallsSelect whether to turn On (Yes) or Off (No).
    Note the following FROM line copied from the sample SIP INVITE below:
    From: "2032625093" <2032625093@trunking-customer.15611.service>;tag=1721003968

    When accepting calls from the IP PBX or SIP Device, the switch will attempt to match the hostname in the FROM Header (trunking-customer.15611.service). This must match what you have configured as the SIP trunk "Name". If you have problems configuring a SIP Trunk, the first step in troubleshooting should be to obtain a SIP trace from the equipment in question and ensure that these match.

    @2013-09-18_19:24:46
Received Packet from 75.201.192.178:5060
INVITE sip:18452130635@dtoys.18235.service;user=phone SIP/2.0
Via: SIP/2.0/UDP 75.201.192.178:5060;branch=z9hG4bK-55257c08b011e59d61b2d3b1b28533aa;rport
From: "2032625093" <2032625093@trunking-customer.15611.service>;tag=1721003968
To: 12032130635@trunking-customer.15611.service>
Call-ID: 153915ad@pbx
CSeq: 26847 INVITE
Max-Forwards: 70
Contact: 2032625093@75.201.192.178:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Asterisk-PBX
Content-Type: application/sdp
Content-Length: 322
  3. SIP Trunk Caller ID Passthrough
    By default, the Caller ID for all SIP Trunks is overwritten by the Domain's Caller ID. If you need SIP Trunk caller ID and Name passthrough, click to place a check in the 'Passthrough' checkbox found to the right of the field.Note: If you are not familiar with Emergency Caller ID, 911, E911, and 933 Testing, please visit our 911 & E911 Services on SkySwitch - Set-up, Management & Testing Article.
  4. Scroll down further and select whether the IP PBX or SIP Device has a static IP Address. 
    If the IP PBX or SIP device does have a static IP Address, then enter it in the given field.

    Note: If desired, you may enter an additional IP address in the 'IP Address' field. Please note each IP address must contain a space in between IP addresses. For example, you would enter two IP addresses as '1.2.3.4 9.8.7.6' with spaces and NOT with commas. 

    If the IP PBX or SIP Device does not have a Static IP Address, then select 'Require Registration.'

    Whenever possible, do not use the 'Require Registration' unless you know what you are doing. This option often requires expert knowledge of the IP PBX or SIP Device on the customer's premise.
  5. Click on the Dial Planning & Limits tab and select the number of concurrent calls that you wish to allow this customer.
    You may set different limits for Inbound calls, Outbound calls, and Total calls for IP Based Trunks.

    Note:  The default path total is 15 paths.  Please adjust accordingly.  

    For SIP Trunk channel billing-related information please review 'SIP Trunk Calculation' in the linked article.

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