Call Initiation Timeout (s)

Default = 4 seconds. Range = 1 to 99 seconds.

Sets how long the IP Office system should wait for a response to an attempt to initiate a call before following the alternate routes set in an ARS form.

Call Queuing Timeout (m)

Default = 5 minutes.

• For incoming calls, this sets how many minutes the IP Office waits before dropping a call that is waiting for VCM resources or has remained in the unanswered state.

• For outgoing calls, this sets how many minutes the IP Office waits for a call to be answered after receiving a provisional response.

Service Busy Response

Default = 486 - Busy Here (503 - Service Unavailable for the France2 locale).

on No User Responding Send

Default = 408-Request Timeout.

Specifies the cause to be used when releasing incoming calls from SIP trunks, when the cause of releasing is that user did not respond. The options are 408-Request Timeout or 480 Temporarily Unavailable.

Action on CAC Location Limit

Default = Allow Voicemail

When set to Allow Voicemail, the call is allowed to go to a user's voicemail when the user's location call limit has been reached. When set to Reject Call, the call is rejected with the failure response code configured in the Service Busy Response field.

Suppress Q.850 Reason Header

Default = Off.

When SIP calls are released by sending BYE and CANCEL, a release reason header is added to the message. When set to On, the Q.850 reason header is not included.

Emulate NOTIFY for REFER

Default = Off.

Use for SIP providers that do not send NOTIFY messages. When set to On, after IP Office issues a REFER, and the provider responds with 202 ACCEPTED, IP Office will assume the transfer is complete and issue a BYE.

No REFER if using Diversion

Default = Off.

When enabled, REFER is not sent on the trunk if the forwarding was done with 'Send Caller ID = Diversion Header'. Applies to Forwards and Twinning.

Allow Empty INVITE

Default = Off.

When set to On, allows 3pcc devices to initiate calls to IP Office by sending an INVITE without SDP.

Send Empty re-INVITE

Default = Off.

This option is only available if SIP Line > SIP VoIP > Reinvite Supported is selected.

If set to On, when connecting a call between two endpoints, IP Office sends an INVITE without SDP in order to solicit the full media capabilities of both parties.

Allow To Tag Change

Default = Off.

When set to On, allows the IP Office to change media parameters when connecting a call to a different party than that which was advertised in the media parameters of provisional responses, such as 183 Session Progress.

P-Early-Media Support

Default = None.

The options are:

• None: IP Office will not advertise support of this SIP header and will always take incoming early media into account regardless of presence of this header

• Receive: IP Office will advertise support of this SIP header and will discard incoming early media unless this header is present in the SIP message.

• All: IP Office will advertise support of this SIP header, will discard incoming early media unless this header is present in the SIP message and will include this SIP header when providing early media.

Send SilenceSupp=off

Default = Off.

Used for the G711 codec. When checked, the silence suppression off attribute is sent in SDP on this trunk.

Force Early Direct Media

Default = Off.

When set to On, allows the direct connection of early media streams to IP endpoints rather than anchoring it at the IP Office.

Media Connection Preservation

Default = Disabled.

When enabled, allows established calls to continue despite brief network failures. Call handling features are no longer available when a call is in a preserved state. Preservation on public SIP trunks is not supported until tested with a specific service provider.

Indicate HOLD

Default = Off.

When enabled, the system sends a HOLD INVITE to the SIP trunk endpoint.

Media Security

Default = Off

When enabled, the IP Office advertises support of this SIP header, to indicate that audio is configured to be secure and is enforced to use SRTP only. This supports the SIP security header defined by RFC3329.

This option is available only when:

• TLS is being used.

• Line | SIP Line | VoIP > Media Security is selected and set to Enforced.

• Line | SIP Line | VoIP > Fax Transport Support is not set to T38 or T38 Fallback.

When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line are dropped.

Use Phone Context

Default = Off.

When enabled, signals SIP enabled PBXs that the call routing identifier is a telephone number.

Add user=phone

Default = Off.

This setting is available when Use Phone Context is enabled.

When enabled, this setting adds the SIP parameter User with value Phone to the From and To SIP headers of outgoing calls.

Use + for International

Default = Off.

When enabled, outgoing international calls use E.164/International format with + followed by the country code and then the telephone number.

Use PAI for Privacy

Default = Off.

When enabled, if the caller ID is withheld:

• The SIP message From header is made anonymous

• The caller identity is inserted into the P-Asserted-Identity header.

This should only be used in a trusted network and must be stripped out of the SIP message before it is forwarded outside the trusted domain.

Use Domain for PAI

Default = Off.

• When disabled, the DNS resolved IP address of the ITSP Proxy is used for the host part in the P-Asserted-Identity header.

• When enabled, the Domain is used.

Caller ID FROM Header

Default = Off.

Incoming calls can include caller ID information in both the From field and in the PAI fields. When this option is enabled, the caller ID information in the From field is used rather than that in the PAI fields.

Send From In Clear

Default = Off.

When enabled, the user ID of the caller is included in the From field. This applies even if the caller has selected to be or is configured to be anonymous. However, their anonymous state is still honored in other fields used to display the caller identity.

Cache Auth Credentials

Default = On.

When enabled, credentials challenge and response information from a registration transaction is cached by the IP Office and automatically inserted into later SIP messages without waiting for a subsequent challenge. This speeds up connections but must be supported by the other end of the connection.

Add UUI header

Default = Off.

When enabled, the User-to-User Information (UUI) is passed in SIP headers to applications.

Add UUI header to redirected calls

Default = Off.

When enabled, the UUI is passed in SIP headers for calls that are redirected. For example, on forwarded and twinned calls.

This field can be enabled if Add UUI header is enabled.

User-Agent and Server Headers

Default = Blank (Use system type and software level).

The value set in this field is used as the User-Agent and Server value included in SIP request headers made the line.

• If blank, the type of IP Office system and its software level are used.

• Setting a unique value can be useful in call diagnostics when the IP Office has multiple SIP trunks.

Send Location Info

Default = Never.

This option is useable with SIP ISPs that support RFC 4119/RFC 5139. When enabled, emergency calls send the address information associated with the dialing extension's location. See Configuration for Emergency Calls.

The options are:

• Never: Do not send location information.

• Emergency Calls: For Dial Emergency calls, send the address information configured for the dialing extension's location.

By Source IP Address

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 resolution of an FQDN. For UDP calls, the local Listen Port is also used for the match.

• For TCP/TLS connections, the IP Office establishes a connection to the remote address and port specified on the SIP line.

• For UDP, non-call dialogs and call starting dialogs must use the remote address and port specified on the SIP line.

It is recommended that the remote end does not change these value as that may prevent NAT traversal.

"From" header hostpart against ITSP domain

Uses the host part of the From header in the incoming SIP request for association.

• The match is against the Line > SIP Line > ITSP Domain Name.

R-URI hostpart against ITSP domain

Uses the host part of the Request-URI header in the incoming SIP request for association.

• The match is against the Line > SIP Line > ITSP Domain Name.

"To" header hostpart against ITSP domain

Uses the host part of the To header in the incoming SIP request for association.

• The match is against the Line > SIP Line > ITSP Domain Name.

"From" header hostpart against DNS-resolved ITSP domain

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 the IP address resolution of the Line > SIP Line > ITSP Domain Name or, if set, the Line > SIP Transport > ITSP Proxy Address setting.

"Via" header hostpart against DNS-resolved ITSP domain

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 the IP address resolution of the Line > SIP Line > ITSP Domain Name or, if set, the Line > SIP Transport > ITSP Proxy Address setting.

"From" header hostpart against ITSP proxy

Uses the host part of the From header in the incoming SIP request for association.

• The match is against the Line > SIP Transport > ITSP Proxy Address setting.

"To" header hostpart against ITSP proxy

Uses the host part of the From header in the incoming SIP request for association.

• The match is against the Line > SIP Transport > ITSP Proxy Address setting.

R-URI hostpart against ITSP proxy

Uses the host part of the Request-URI in the incoming SIP request for association.

• The match is against the Line > SIP Transport > ITSP Proxy Address setting.

Call Routing Method

Default = Request URI.

This field selects which incoming SIP information is used for incoming number matching by the IP Office to route incoming calls. The options are to match the Request URI or the To Header element provided with the incoming call.

Use P-Called-Party

Default = Off.

When enabled, IP Office reads the P-Called-Party ID header if present in the SIP message and routes the incoming SIP calls based on it. The feature can be enabled on public SIP trunk interfaces.

If enabled and the header is not present in the SIP message, the IP Office uses the header configured in the Call Routing Method for incoming call routing.

Suppress DNS SRV Lookups

Default = Off.

Controls whether to send SRV queries for this endpoint, or just NAPTR and A record queries.