Trunk Group and Routing Parameters

Defines the method for allocating incoming IP-to-Tel calls to a channel. The parameter applies to the following:

For all channels that are configured without a Trunk Group ID:

Note: This option is applicable only to FXS interfaces.

For a detailed description of the parameter's options, see Configuring Trunk Group Settings.

For IP-to-Tel calls, the parameter defines the default destination (called) phone number if the received SIP message doesn't contain a called party number and a phone number ('Channels' parameter) is not configured in the Trunk Groups table (see Configuring Trunk Groups). The final destination number is the value of this parameter plus the channel ID.

For Tel-to-IP calls, the parameter defines the default source (calling) phone number if the received ISDN message doesn't contain a calling party number and a phone number ('Channels' parameter) is not configured in the Trunk Groups table.

The parameter is used as a starting number for the channels of all the trunks.

The default is 1000.

For example, for a Tel-to-IP call, if you configure the parameter to "2000" and the 'Channels' parameter in the Trunks Groups table to "34", the source number is 2034.

Defines which IP address the device uses to determine the source of incoming INVITE messages for IP-to-Tel routing.

Defines the use of the Tel Source Number and Display Name for Tel-to-IP calls.

Defines how the display name (caller ID) received from the IP side (in the SIP From header) effects the source number sent to the Tel side, for IP-to-Tel calls.

Defines the ENUM service for translating telephone numbers to IP addresses or domain names (FQDN), for example, e164.arpa, e164.customer.net, or NRENum.net.

The valid value is a string of up to 50 characters. The default is "e164.arpa".

Note: ENUM-based routing is configured in the Tel-to-IP Routing table using the "ENUM" string value as the destination address to denote the parameter's value.

Determines whether to use the device's routing table to obtain the URI host name and optionally, an IP profile (per call) even if a Proxy server is used.

Note:

Determines whether to route Tel calls to an IP destination before or after manipulation of the destination number. This applies to Tel-to-IP routing rules configured in the Tel-to-IP Routing table.

Note:

Determines whether to route IP calls to the Trunk Group before or after manipulation of the destination number (configured in Configuring Source/Destination Number Manipulation Rules).

Defines the device's policy for accepting or blocking SIP calls (IP-to-Tel calls). This is useful in preventing unwanted SIP calls, SIP messages, and/or VoIP spam.

Note: If the parameter is set to [0] or [1], when using Proxies or Proxy Sets, it is unnecessary to configure the Proxy IP addresses in the routing table. The device allows SIP calls received from the Proxy IP addresses even if these addresses are not configured in the routing table.

Enables filtering of Tel-to-IP calls when a Proxy Set is used.

When the parameter is enabled and a proxy is used, the device first checks the Tel-to-IP Routing table before making a call through the proxy. If the number is not allowed (i.e., number isn't listed in the table or a call restriction routing rule of IP address 0.0.0.0 is applied), the call is released.

Note: When no proxy is used, the parameter must be disabled and filtering is according to the Tel-to-IP Routing table.

Assigns the Dial Plan (by name) to be used for tag-based IP-to-Tel routing rules. The Dial Plan's tags can be used as matching criteria (source and destination) for routing rules in the IP-to-Tel Routing table.

For more information, see Using Dial Plans for IP-to-Tel or Tel-to-IP Call Routing.

Assigns the Dial Plan (by name) to be used for tag-based Tel-to-IP routing rules. The Dial Plan's tags can be used as matching criteria (source and destination) for routing rules in the Tel-to-IP Routing table.

For more information, see Using Dial Plans for IP-to-Tel or Tel-to-IP Call Routing.

Determines whether to add the Carrier Identification Code (CIC) as a prefix to the destination phone number for IP-to-Tel calls. When the parameter is enabled, the 'cic' parameter in the incoming SIP INVITE can be used for IP-to-Tel routing decisions. It routes the call to the appropriate Trunk Group based on the parameter's value.

For example, as a result of receiving the below INVITE, the destination number after number manipulation is cic+167895550001:
INVITE sip:5550001;cic=+16789@172.18.202.60:5060;user=phone SIP/2.0

Note: After the cic prefix is added, the IP-to-Tel Routing table can be used to route this call to a specific Trunk Group. The Destination Number IP to Tel Manipulation table must be used to remove this prefix before placing the call to the Tel side.

Enables the re-routing of incoming Tel-to-IP calls that are identified as fax calls. If a CNG tone is detected on the Tel side of a Tel-to-IP call, the device adds the string, "FAX" as a prefix to the destination number before routing and manipulation. A routing rule in the Tel-to-IP Routing table having the value "FAX" (case-sensitive) as the destination number is then used to re-route the call to a fax destination and the destination number manipulation mechanism is used to remove the "FAX" prefix before sending the fax, if required. If the initial INVITE used to establish the voice call (not fax) was already sent, a CANCEL (if not connected yet) or a BYE (if already connected) is sent to release the voice call.

Note: The parameter has replaced the EnableFaxRerouting parameter. For backward compatibility, the EnableFaxRerouting parameter set to 1 is equivalent to the FaxReroutingMode parameter set to 1.

Defines how the device handles the receipt of multiple SIP 18x forking responses for Tel-to-IP calls. The forking 18x response is the response with a different SIP to-tag than the previous 18x response. These responses are typically generated (initiated) by Proxy / Application servers that perform call forking, sending the device's originating INVITE (received from SIP clients) to several destinations, using the same Call ID.

Note: Regardless of the parameter setting, once a SIP 200 OK response is received, the device uses the RTP information and re-opens the voice stream, if necessary.

Defines the timeout (in seconds) that is started after the first SIP 2xx response has been received for a User Agent when a Proxy server performs call forking (Proxy server forwards the INVITE to multiple SIP User Agents). The device sends a SIP ACK and BYE in response to any additional SIP 2xx received from the Proxy within this timeout. Once this timeout elapses, the device ignores any subsequent SIP 2xx.

The number of supported forking calls per channel is 20. In other words, for an INVITE message, the device can receive up to 20 forking responses from the Proxy server.

The valid range is 0 to 30. The default is 30.

Enables Tel-to-IP call forking, whereby a Tel call can be routed to multiple IP destinations.

Note: Once enabled, routing rules must be assigned Forking Groups in the Tel-to-IP Routing table.