SIP Trunking Setup- Domain or Carrier connectivity (BYOT)
IMPORTANT: For first time SIP Trunk Configurations we highly recommend requesting SIP Engineering support via a support ticket.
REQUEST ASSISTANCE - Please complete this information in the form below.
Looking for BYOT? SkySwitch supports interoperability with the most common US based carriers. Carriers must be sending calls in e164 format. Resellers are responsible for support and troubleshooting carrier issues outside of the SkySwitch network. To request Interop with your own carrier for Origination or Termination, please visit the following page for more information on pricing. Then, fill out this form to get started.
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.
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 a 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" <firstname.lastname@example.org>;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 126.96.36.199:5060 INVITE sip:email@example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 188.8.131.52:5060;branch=z9hG4bK-55257c08b011e59d61b2d3b1b28533aa;rport From: "2032625093" <firstname.lastname@example.org>;tag=1721003968 To: email@example.com> Call-ID: 153915ad@pbx CSeq: 26847 INVITE Max-Forwards: 70 Contact: firstname.lastname@example.org: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. For SIP Trunk channel billing related information please review 'SIP Trunk Calculation' in the linked article.
5. 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 passthrough, change the Domain's Caller ID to [*], this can be found at the bottom of the Domain Settings under Defaults.