SIP Trunk Details

•Proxy Server Address
In exceptional circumstances, the IP Address of the proxy server may be explicitly identified as either a different IP Address, or a different domain address resolvable by DNS.

•! WARNING - Reboot Required
Changing this setting requires the system to be rebooted for the change to take effect. Rebooting the system will end all calls currently in progress.

•DNS Server Address
If the proxy server address is set to a named server, the address of the DNS server used for name resolution should be entered here.

•Mobility Caller ID Format
This option corresponds to the standard "draft-ietf-sip-privacy-04". The options are None, Remote Party ID, P Asserted ID or Diversion Header.

•Use Tel URI: Default = SIP URI.
Select the format of numbering to be used in the From field on outgoing calls. The options are SIP URI or Tel URI. Tel URI uses the format TEL: +1-425-555-4567. SIP URI uses the format name@example.com).

•Check OOS: Default = On. Software level = 8.0+.
When enabled, the system will regularly check if the trunk is in service. Checking that SIP trunks are in service ensures that outgoing calls are not delayed waiting for response on a SIP trunk that is not currently usable. Depending on the trunk's Transport Protocol, the trunks current service status is checked using the following methods:

•For all trunks, regular OPTIONS messages are sent. If no reply is received, the trunk is taken out of service.

•For TCP trunks, if the TCP connection is disconnected the trunk will be taken out of service.

•For trunks using DNS, if the IP address is not resolved or the DNS resolution has expired, the trunk is taken out of service.

•Call Routing Method: Default = Request URI. Software level = 8.0+.
This field allows selection of which part of the incoming SIP information should be used for the incoming number. The options are to match either the Request URI or the To Header element provided with the incoming call.

•Association Method: Default = By Source IP address. Software level = 8.0+.
This field sets the method by which a SIP line is associated with an incoming SIP request. The search for a line match for an incoming request is done against each line until a match occurs. If no match occurs, the request is ignored. This method allow multiple SIP lines with the same address settings. This may be necessary for scenarios where it may be required to support multiple SIP lines to the same ITSP. For example when the same ITSP supports different call plans on separate lines or where all outgoing SIP lines are routed from the system via an additional on-site system.

•By Source IP Address
This option uses the source IP address and port of the incoming request for association. The match is against the configured remote end of the SIP line, using either an IP address/port or the resolution of a fully qualified domain name. This matches the method used by pre-8.0 systems.

•"From" header hostpart against ITSP domain
This option uses the host part of the From header in the incoming SIP request for association. The match is against the line's Domain Name.

•R-URI hostpart against ITSP domain
This option uses the host part of the Request-URI header in the incoming SIP request for association. The match is against the line's Domain Name.

•"To" header hostpart against ITSP domain
This option uses the host part of the To header in the incoming SIP request for association. The match is against the line's Domain Name.

•"From" header hostpart against DNS-resolved ITSP domain
This option uses the host part of the FROM header in the incoming SIP request for association. The match is found by comparing the FROM header against a list of IP addresses resulting from resolution of the line's Domain Name or, if set, the Proxy Server Address.

•"Via" header hostpart against DNS-resolved ITSP domain
This option uses the host part of the VIA header in the incoming SIP request for association. The match is found by comparing the VIA header against a list of IP addresses resulting from resolution of the line's Domain Name or, if set, the line's Proxy Server Address.

•"From" header hostpart against ITSP proxy
This option uses the host part of the “From” header in the incoming SIP request for association. The match is against the line's Proxy Server Address.

•"To" header hostpart against ITSP proxy
This option uses the host part of the From header in the incoming SIP request for association. The match is against the line's Proxy Server Address.

•R-URI hostpart against ITSP proxy
This option uses the host part of the Request-URI in the incoming SIP request for association. The match is against the line's Proxy Server Address.

•Name Priority: Default = Favour Trunk.
For SIP trunks, the caller name displayed on an extension can either be that supplied by the trunk or one obtained by checking for a number match in the system speed dials. This setting determines which method is used by the line. Select one of the following options:

•System Default
Use the system's Default Name Priority setting, the default being Favour Trunk.

•Favour Trunk
Display the name provided by the trunk. For example, the trunk may be configured to provide the calling number or the name of the caller. The system should display the caller information as it is provided by the trunk.

•Favour Directory
Search for a number match in the system speed dials. The first match is used and overrides the name provided by the SIP line. If no match is found, the name provided by the line is used.

•Calls Route Via Registrar: Default = On
Normally SIP REGISTER requests and INVITE requests use the same server destination. This option should only be deselected when the service provider does not expect REGISTER requests to go to the same destination as the INVITE requests. You should only set this under specific instruction from the service provider.

•Separate Registrar
This field is available when Calls Route Via Registrar is deselected. It is used to enter the address of the SIP server that should be used for registration. You should only set this under specific instruction from the service provider.

•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.

•UPDATE Supported: Default = Never. Software level = 8.0+.
The SIP UPDATE method (RFC 3311) allows a client to update parameters of a session (such as the set of media streams and their codecs) but has no impact on the state of a dialog. It is similar to re-INVITE, but can be sent before the initial INVITE has completed. This allows it to update session parameters within early dialogs.