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

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Concept


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 will be able to limit the total number of Inbound, Outbound and Total call sessions associated with the SIP Trunk.

Step-by-step guide

1. Browse to the SIP Trunking item in the top menu.

2. Click "Add SIP Trunk". The resulting form will allow you to enter the SIP Trunk details.

Name This 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.
Domain This is the customer Domain that will be associated with this account.
Description Any description of the customer's equipment. It is helpful to put the type of equipment or software here (eg. Asterisk PBX).
Relay Media This 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 better quality of service. If you are unsure, then select 'Don't Care' and the switch will determine the most appropriate setting.
Trunk Type Select whether this customer should be allowed to make Inbound Only, Outbound Only or Bidirectional calls.
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. Click on the Connectivity section 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.

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.

4. Click on the Limitations section 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. Registration Trunks only have an Inbound and Outbound Limit.