SIP Trunk Administration

This menu is used to add SIP trunks to the phone system configuration. Before adding any SIP trunks, the system must be configured for general SIP operation through the STUN Settings for the Network section of the Advanced Parameters settings.

•SIP Trunk Channel Licenses
The system can support 3 simultaneous SIP calls without needing licenses. Additional simultaneous calls, up to 20 in total, require the addition of licenses to the configuration.

•VCM Channels
Note that for SIP calls the system also requires VCM channels. For a Basic Edition system those are provided by installing IP500 Combination base cards. Each of these cards provides 10 VCM channels.

SIP Trunk Setup

•Descriptive Name
The name of the trunk

•Domain Name: Default = Blank
Each SIP Trunk configuration has a unique ITSP Domain name needed by SIP end points in order to register with the IP Office. This is a string which may be directly resolved to an IP Address, or may require DNS lookup to resolve the domain name to the Service provider’s address. If this field is left blank, registration is against the LAN IP address.

•Authentication Name: Default = Blank.
This value is provided by the SIP ITSP.

•Password: Default = Blank.
This value is provided by the SIP ITSP.

•Number of Channels: Default = 10
Number of trunk channels between 1 and 24

•Transport Protocol: Default = Both TCP & UDP.
Both TCP and UDP SIP end points are supported. This field can be used to restrict the IP Office to just TCP or UDP if required.

•Send Port: Default = 5060.
The port to use for outgoing call support.

•Listen Port: Default = 5060.
The port to use for incoming call support.

•Advanced Setup
Clicking on Advanced Setup when an SIP Trunk line type is selected in the list and a domain name has been entered, accesses a menu of additional settings.

SIP Trunk Channel Setup

•Channel
Set by the system. Shown for information only.

•Appearance ID
Appearance ID numbers can be used to associate each channel a Line Appearance button on phones that support button programming. That button can then be used to make and answer calls using the channel. The line appearance ID for each channel is automatically assigned to those channels that have their Direction set as Bothways.

•Direction: Default = Bothways
Sets the allowed operation mode of the line. For systems running in Key mode, a line can be set to either Bothway (incoming and outgoing) or Incoming Call by Call (incoming only). For a system running in PBX mode, a line can be set to either Bothway (incoming and outgoing) or Call by Call (incoming and outgoing).

•Bothway
When set to Bothway, incoming calls are presented to line appearance buttons matching the channels Appearance ID and to the channels Coverage Destination if set. For Key mode systems, outgoing calls are routed to the channel by pressing the matching line appearance button selection or by automatic line selection. In addition, on PBX mode systems, outgoing calls can be routed to the channel by including the line appearance in the ARS Selector that matches the dialed digits.

•Incoming Call by Call
For systems running in Key mode, when set to Incoming Call by Call, incoming calls are routed using the Call by Call table. The Appearance ID, Coverage Destination and Unique Line Ringing fields are greyed-out as those settings are not applied. The trunk channel is not used for outgoing calls.

•Call by Call
For systems running in PBX mode, when set to Call by Call, incoming calls are routed using the Call by Call table. The Appearance ID, Coverage Destination and Unique Line Ringing fields are greyed-out as those settings are not applied. In PBX mode, call by call entries can be used given ARS selector numbers (see below) which allow the trunk channel to also be used for outgoing calls.

•Display Name: Default = Use Authentication Name
This field sets the 'Name' value for SIP calls using this URI.

•Local URI:
The user part of the SIP URI. This specifies the contents of the FROM field when making a call (sending an INVITE).

•Anonymous:
Withhold the calling parties information.

•Coverage Destination: Default = None. System Mode = Key System
This option sets where incoming calls should alert in addition to alerting on those extension that have a line appearance button programmed for the line. When the phone system is in night service mode, calls alert at the members of the Night Service group.

•The Coverage Destination is not used for SIP trunks with their direction set to Incoming Call by Call.

•Unique Line Ringing: Default = 1. Software level = 6.1+.
Selects the ring pattern that should be used for calls when alerting on an extension. Calls forwarded, sent to call coverage or to a hunt group will always use the line ring pattern. Calls direct to an extension will use the line ringing pattern unless the user has Override Line Ringing set. Not used for calls presented to the user as a member of the Operator group. This feature is also not used for BST phones.

•Registration Required
When on, each local URI with unique Authentication credentials will register independently.

•Authentication Name
When making a call, some service providers will often send an authentication challenge to validate the call before it is connected. This challenge requires the INVITE is re-submitted with Authentication data, including a network account name (provided by the service provider during installation). The network account name is the “Auth name”. It can be blank, in which case the Local URI is used.

•Password: Default = Blank.
This value is provided by the SIP ITSP.

•P-Assert-ID
If this field is configured, the channel can be used in SIPConnect Option 1 model for separating Public and Private PSTN identity (Sipconnect technical recommendation v 10, section 12.1.1). You can only use Explicit CLI configurations over SIP if using Option1 model for identity. In this case, calls over this channel will always have a fixed P-Assert-ID, but the From field may vary.

Call by Call Table

These settings are used to match calls received on SIP trunks channels set to Incoming Call-by-Call above. For systems operating in Key System mode, the default entry is used for all calls for which there is no other match and is fixed to route those calls to the Operator Group.

•ARS
This setting is only shown for PBX System mode systems. For those systems, each call-by-call entry can be assigned to an ARS Selector number. That selector number can then be used as the destination for outgoing calls.

•Local URI:
The user part of the SIP URI. This specifies the contents of the FROM field when making a call (sending an INVITE).

•Display Name: Default = Use Authentication Name
This field sets the 'Name' value for SIP calls using this URI. The value can either be entered manually or the options Use Authentication Name or Use Internal Data selected.

•Destination
Where incoming calls with matching digits should be routed. The drop-down list contains the extensions and groups on the IP Office system.

•Registration Required
When on, each local URI with unique Authentication credentials will register independently.

•Authentication Name
When making a call, some service providers will often send an authentication challenge to validate the call before it is connected. This challenge requires the INVITE is re-submitted with Authentication data, including a network account name (provided by the service provider during installation). The network account name is the “Auth name”. It can be blank, in which case the Local URI is used.

•Password: Default = Blank.
This value is provided by the SIP ITSP.

•P-Assert-ID
If this field is configured, the channel can be used in SIPConnect Option 1 model for separating Public and Private PSTN identity (Sipconnect technical recommendation v 10, section 12.1.1). You can only use Explicit CLI configurations over SIP if using Option1 model for identity. In this case, calls over this channel will always have a fixed P-Assert-ID, but the From field may vary.