Trunk Group and Routing Parameters

The routing parameters are described in the table below.

Routing Parameters

Parameter

Description

'Channel Select Mode'

ch-select-mode

[ChannelSelectMode]

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

Trunks configured without a channel select mode in the Trunk Group Settings table (see Configuring Trunk Group Settings).
Channels and trunks configured without a Trunk Group ID.

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

[0] By Dest Phone Number
[1] Cyclic Ascending (default)
[2] Ascending
[3] Cyclic Descending
[4] Descending
[5] Dest Number + Cyclic Ascending
[6] By Source Phone Number
[7] Trunk Cyclic Ascending
[8] Trunk & Channel Cyclic Ascending
[11] Dest Number + Ascending

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

'Default Destination Number'

configure voip > gateway dtmf-supp-service dtmf-and-dialing > dflt-dest-nb

[DefaultNumber]

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.

'Source IP Address Input'

configure voip > gateway routing settings > src-ip-addr-input

[SourceIPAddressInput]

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

[-1] Not Configure = (Default) The parameter is automatically set to SIP Contact header (1).
[0] SIP Contact Header = The IP address in the Contact header of the incoming INVITE message is used.
[1] Layer 3 Source IP = The actual IP address (Layer 3) from where the SIP packet is received is used.

'Use Source Number As Display Name'

configure voip > sip-definition settings > src-nb-as-disp-name

[UseSourceNumberAsDisplayName]

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

[0] No = (Default) If a Tel Display Name is received, the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name. If no Display Name is received from the Tel side, the IP Display Name remains empty.
[1] Yes = If a Tel Display Name is received, the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name. If no Display Name is received from the Tel side, the Tel Source Number is used as the IP Source Number and also as the IP Display Name.
[2] Overwrite = The Tel Source Number is used as the IP Source Number and also as the IP Display Name (even if the received Tel Display Name is not empty).
[3] Original = Similar to option [2], except that the operation is done before regular calling number manipulation.

'Use Display Name as Source Number'

configure voip > sip-definition settings > disp-name-as-src-nb

[UseDisplayNameAsSourceNumber]

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.

[0] No = (Default) If a display name is received from the IP side, the source number of the IP side is used as the Tel source number.
[1] Yes = If a display name is received from the IP side, the display name of the IP side is used as the Tel source number and Presentation is set to Allowed (0). If no display name is received from the IP side, the source number of the IP side is used as the Tel source number and Presentation is set to Restricted (1). For example:
If 'From: 100 <sip:200@201.202.203.204>' is received from the IP side, the outgoing source number (and display name) are set to "100" and Presentation is set to Allowed (0).
If 'From: <sip:400@101.102.103.104>' is received from the IP side, the outgoing source number is set to "400" and Presentation is set to Restricted (1).
[2] Preferred = If a display name is received from the IP side, the display name of the IP side is used as the Tel source number. If no display name is received from the IP side, this setting doesn't affect the Tel source number.

'ENUM Resolution'

configure voip > sip-definition settings > enum-service-domain

[EnumService]

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.

'Use Routing Table for Host Names and Profiles

configure voip > sip-definition settings > rte-tbl-4-host-names

[AlwaysUseRouteTable]

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.

[0] Disable = (Default) Don't use the Tel-to-IP Routing table.
[1] Enable = Use the Tel-to-IP Routing table.

Note:

The parameter appears only if the 'Use Default Proxy' parameter is enabled.
The domain name is used instead of a Proxy name or IP address in the INVITE SIP URI.

'Tel to IP Routing Mode'

configure voip > gateway routing settings > tel2ip-rte-mode

[RouteModeTel2IP]

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.

[0] Route calls before manipulation = Calls are routed before the number manipulation rules are applied (default).
[1] Route calls after manipulation = Calls are routed after the number manipulation rules are applied.

Note:

The parameter is not applicable if outbound proxy routing is used.
For number manipulation, see Configuring Source/Destination Number Manipulation.
To configure Tel-to-IP routing rules, see Configuring Tel-to-IP Routing Rules.

'IP-to-Tel Routing Mode'

configure voip > gateway routing settings > ip2tel-rte-mode

[RouteModeIP2Tel]

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).

[0] Route calls before manipulation = (Default) Calls are routed before the number manipulation rules are applied.
[1] Route calls after manipulation = Calls are routed after the number manipulation rules are applied.

'IP Security'

configure voip > sip-definition settings > ip-security

[SecureCallsFromIP]

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.

[0] Disable = (Default) The device accepts all SIP calls.
[1] Secure Incoming calls = The device accepts SIP calls only from IP addresses that are configured in the Tel-to-IP Routing table or Proxy Sets table, or IP addresses resolved by DNS servers from FQDN values configured in the Proxy Sets table. All other incoming calls are rejected.
[2] Secure All calls = The device accepts SIP calls only from IP addresses (in dotted-decimal notation format) that are configured in the Tel-to-IP Routing table and rejects all other incoming calls. In addition, if an FQDN is configured in the Tel-to-IP Routing table or Proxy Sets table, the call is allowed to be sent only if the resolved DNS IP address appears in one of these tables; otherwise, the call is rejected. Therefore, the difference between this option and option [1] is that this option is concerned only about numerical IP addresses that are defined in the tables.

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.

'Filter Calls to IP'

configure voip > sip-definition settings > filter-calls-to-ip

[FilterCalls2IP]

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

[0] Don't Filter = (Default) The device doesn't filter calls when using a proxy.
[1] Filter = Filtering is enabled.

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.

'Tel-to-IP Dial Plan Name'

configure voip > gateway routing settings > tel-dial-plan-name

[Tel2IPDialPlanName]

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.

'IP-to-Tel Dial Plan Name'

configure voip > gateway routing settings > ip-dial-plan-name

[IP2TelDialPlanName]

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.

'IP-to-Tel Tagging Destination Dial Plan Index'

configure voip > gateway routing settings > ip2tel-tagging-dst

[IP2TelTaggingDestDialPlanIndex]

Defines the Dial Plan index in the Dial Plan file for called prefix tags for representing called number prefixes in Inbound Routing rules.

The valid values are 0 to 7, where 0 denotes PLAN1, 1 denotes PLAN2, and so on. The default is -1 (i.e., no dial plan file used).

For more information on this feature, see Dial Plan Prefix Tags for IP-to-Tel Routing.

'IP to Tel Tagging Source Dial Plan Index'

cconfigure voip > gateway routing settings > ip-to-tel-tagging-src

[IP2TelTaggingSourceDialPlanIndex]

Defines the Dial Plan index in the Dial Plan file for calling prefix tags for representing calling number prefixes in Inbound Routing rules.

The valid values are 0 to 7, where 0 denotes PLAN1, 1 denotes PLAN2, and so on. The default is -1 (i.e., no dial plan file used).

For more information on this feature, see Dial Plan Prefix Tags for IP-to-Tel Routing.

configure voip > gateway digital settings > etsi-diversion

[EnableETSIDiversion]

Defines the method in which the Redirect Number is sent to the Tel side.

[0] = (Default) Q.931 Redirecting Number Information Element (IE).
[1] = ETSI DivertingLegInformation2 in a Facility IE.

'Add CIC'

configure voip > gateway manipulation settings > add-cic

[AddCicAsPrefix]

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.

[0] No (default)
[1] Yes

The SIP 'cic' parameter enables the transmission of the 'cic' parameter from the SIP network to the ISDN. The 'cic' parameter is a three- or four-digit code used in routing tables to identify the network that serves the remote user when a call is routed over many different networks. The 'cic' parameter is carried in the SIP INVITE and maps to the ISDN Transit Network Selection Information Element (TNS IE) in the outgoing ISDN Setup message (if the EnableCIC parameter is set to 1). The TNS IE identifies the requested transportation networks and allows different providers equal access support, based on customer choice.

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.

[FaxReroutingMode]

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.

[0] Disable (default)
[1] Rerouting without Delay = Upon detection of a CNG tone, the device immediately releases the call of the initial INVITE and then sends a new INVITE to a specific IP Group or fax server according to the Tel-to-IP Routing table. To enable this feature, set the CNGDetectorMode parameter to 2 and the IsFaxUsed parameter to 1, 2, or 3.
[2] Progress and Delay = Incoming ISDN calls are delayed until a CNG tone detection or timeout, set by the FaxReroutingDelay parameter. If the EnableComfortTone parameter is set to 1, a Q.931 Progress message with Protocol Discriminator set to 1 is sent to the PSTN and a comfort tone is played accordingly to the PSTN. When the timeout expires, the device sends an INVITE to a specific IP Group or to a fax server, according to the Tel-to-IP Routing table rules.
[3] Connect and Delay = Incoming ISDN calls are delayed until a CNG tone detection or timeout, set by the FaxReroutingDelay parameter. A Q.931 Connect message is sent to the PSTN. If the EnableComfortTone parameter is set to 1, a comfort tone is played to the PSTN. When the timeout expires, the device sends an INVITE to a specific IP Group or to a fax server according to the Tel-to-IP Routing table rules.

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.

[FaxReroutingDelay]

Defines the maximum time interval (in seconds) that the device waits for CNG detection before re-routing calls identified as fax calls to fax destinations (terminating fax machine).

The valid value range is 1-10. The default is 5.

Call Forking Parameters

'Forking Handling Mode'

forking-handling

[ForkingHandlingMode]

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.

[0] Parallel handling = (Default) If SIP 18x with SDP is received, the device opens a voice stream according to the received SDP and disregards any subsequently received 18x forking responses (with or without SDP). If the first response is 180 without SDP, the device responds according to the PlayRBTone2TEL parameter and disregards the subsequent forking 18x responses.
[1] Sequential handling = If 18x with SDP is received, the device opens a voice stream according to the received SDP. The device re-opens the stream according to subsequently received 18x responses with SDP, or plays a ringback tone if 180 response without SDP is received. If the first received response is 180 without SDP, the device responds according to the PlayRBTone2TEL parameter and processes the subsequent 18x forking responses.

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.

'Forking Timeout'

configure voip > gateway advanced > forking-timeout

[ForkingTimeOut]

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.

'Tel2IP Call Forking Mode'

configure voip > sip-definition settings > tel2ip-call-forking-mode

[Tel2IPCallForkingMode]

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

[0] Disable (default)
[1] Enable

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

'Forking Delay Time For Invite'

configure voip > sip-definition settings > forking-delay-time-invite

[ForkingDelayTimeForInvite]

Defines the interval (in seconds) to wait before sending INVITE messages to the other members of the forking group. The INVITE is immediately sent to the first member.

The valid value range is 0 to 40. The default is 0 (i.e., sends immediately).