1.10 New Parameters
1.10.1 SIP Parameters
The table below describes the new SIP parameters for Release 6.0. Table 1-1: New SIP Parameters for Release 6.0
Parameter Description
[SIPForceRport]
Determines whether the device sends SIP responses to the UDP port from where SIP requests are received even if the "rport" parameter is not included in the Via header.
[0] (default) = Disabled - the device sends the SIP response to the UDP port defined in the Via header. If the Via header contains the "rport" parameter, the response is sent to the UDP port from where the SIP request is received.
[1] = Enabled - SIP responses are sent to the UDP port from where SIP requests are received even if the "rport" parameter is not included in the Via header.
[MLPPNetworkIdentifier] Defines the MLPP network identifier (i.e., International prefix or Telephone Country Code/TCC) for IP-to-ISDN calls, according to the UCR 2008 and ITU Q.955 specifications.
The valid range is 1 to 999. The default is 1 (i.e., USA).
The MLPP network identifier is sent in the Facility IE of the Setup message. For example:
MLPPNetworkIdentifier set to default (i.e., USA, 1): PlaceCall- MLPPNetworkID:0100 MlppServiceDomain:123abc, MlppPrecLevel:5
Fac(1c): 91 a1 15 02 01 05 02 01 19 30 0d 0a 01 05 0a 01 01 04 05 01 00 12 3a bc
MLPPNetworkIdentifier set to 490:
PlaceCall- MLPPNetworkID:9004 MlppServiceDomain:123abc, MlppPrecLevel:5
Parameter Description Web: Redirect Number Tel -> IP
[RedirectNumberMapTel2I P]
This ini file table parameter manipulates the redirect number for Tel-to- IP calls. The manipulated Redirect Number is sent in the SIP
Diversion, History-Info, or Resource-Priority headers. The format of this parameter is as follows:
[RedirectNumberMapTel2Ip] FORMAT RedirectNumberMapTel2Ip_Index = RedirectNumberMapTel2Ip_DestinationPrefix, RedirectNumberMapTel2Ip_RedirectPrefix, RedirectNumberMapTel2Ip_NumberType, RedirectNumberMapTel2Ip_NumberPlan, RedirectNumberMapTel2Ip_RemoveFromLeft, RedirectNumberMapTel2Ip_RemoveFromRight, RedirectNumberMapTel2Ip_LeaveFromRight, RedirectNumberMapTel2Ip_Prefix2Add, RedirectNumberMapTel2Ip_Suffix2Add, RedirectNumberMapTel2Ip_IsPresentationRestricted, RedirectNumberMapTel2Ip_SrcTrunkGroupID, RedirectNumberMapTel2Ip_SrcIPGroupID; [\RedirectNumberMapTel2Ip] For example: RedirectNumberMapTel2Ip 1 = *, 4, 255, 255, 0, 0, 255, , 972, 255, 1, 2; Notes:
This parameter table can include up to 20 indices (1-20).
The parameters NumberType and NumberPlan are applicable only for the SIP Resource-Priority header.
If the table's matching characteristics rule (i.e., DestinationPrefix, RedirectPrefix, SrcTrunkGroupID, and SrcIPGroupID) is located for the Tel-to-IP call, then the redirect number manipulation rule (defined by the other parameters) is applied to the call.
The manipulation rules are performed in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Mediant 2000 & Mediant 3000
Parameter Description
Web: Redirect Number IP -> Tel [RedirectNumberMapIp2T el]
This ini file table parameter manipulates the redirect number for IP-to- Tel calls. This manipulates the value of the SIP Diversion, History-Info, or Resource-Priority headers (including the reason the call was redirected).
The format of this parameter is as follows: [RedirectNumberMapIp2Tel] FORMAT RedirectNumberMapIp2Tel_Index = RedirectNumberMapIp2Tel_DestinationPrefix, RedirectNumberMapIp2Tel_RedirectPrefix, RedirectNumberMapIp2Tel_SourceAddress, RedirectNumberMapIp2Tel_NumberType, RedirectNumberMapIp2Tel_NumberPlan, RedirectNumberMapIp2Tel_RemoveFromLeft, RedirectNumberMapIp2Tel_RemoveFromRight, RedirectNumberMapIp2Tel_LeaveFromRight, RedirectNumberMapIp2Tel_Prefix2Add, RedirectNumberMapIp2Tel_Suffix2Add, RedirectNumberMapIp2Tel_IsPresentationRestricted; [\RedirectNumberMapIp2Tel] For example: RedirectNumberMapIp2Tel 1 = *, 88, *, 1, 1, 2, 0, 255, 9, , 255; Notes:
This parameter table can include up to 20 indices (1-20).
If the table's matching characteristics rule (i.e., DestinationPrefix, RedirectPrefix, and SourceAddress) is located for the IP-to-Tel call, then the redirect number manipulation rule (defined by the other parameters) is applied to the call.
The manipulation rules are executed in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and Suffix2Add.
The RedirectPrefix parameter is used before any manipulation has been performed on it.
The redirect manipulation is performed after the parameter CopyDest2RedirectNumber.
Web: Forward On Busy Trunk Destination [ForwardOnBusyTrunkDe
st]
This ini file table parameter configures the Forward On Busy Trunk Destination table. This table allows you to define an alternative call destination (IP address) per Trunk Group for IP-to Tel calls. The IP-to-
Parameter Description FORMAT ForwardOnBusyTrunkDest_Index = ForwardOnBusyTrunkDest_TrunkGroupId, ForwardOnBusyTrunkDest_ForwardDestination; [\ForwardOnBusyTrunkDest] Where:
TrunkGroupId = Trunk Group for which you want to define a call forwarding destination. The default is 0.
ForwardDestination = Alternative destination, using the syntax "host:port;transport=xxx"(i.e., IP address, port and transport type). For example, the below configuration forwards IP-to-Tel calls to destination IP address 10.13.4.12, port 5060 using transport protocol TCP, if Trunk Group ID 2 is busy:
ForwardOnBusyTrunkDest 1 = 2, 10.13.4.12:5060;transport=tcp;
Notes:
The maximum number of indices (starting from Index 1) depends on the maximum number of Hunt/Trunk Groups.
For the destination, instead of a dotted-decimal IP address, FQDN can be used.
[SASEnableContactRepla ce]
Enables the device to change the Contact header so that it points to the SAS host, and therefore, the top-most Via header and the Contact header point to the same host.
[0] (default) = Disable - when relaying requests, the SAS agent adds a new SIP Via header (with the SAS IP address) as the top- most Via header and retains the original SIP Contact header. Thus, the top-most Via header and the Contact header point to different hosts.
[1] = Enable - the device changes the Contact header so that it points to the SAS host and therefore, the top-most Via header and the Contact header point to the same host.
Note: Operating in this mode causes all incoming dialog requests to traverse the SAS and thus, may cause load problems.
Web: Enable Record-Route [SASEnableRecordRoute]
Determines whether the device's SAS application adds the SIP Record-Route header to SIP requests. This ensures that SIP messages traverse the device's SAS agent, by including the SAS IP address in the Record-Route header.
[0] Disable (default) [1] Enable
The Record-Route header is inserted in a request by a SAS proxy to force future requests in the dialog session to be routed through the SAS agent. Each traversed proxy in the path can insert this header, causing all future dialogs in the session to pass through it as well.
When this feature is enabled, the SIP Record-Route header includes the URI "lr" parameter. The presence of this parameter indicates loose routing; the lack of 'lt' indicates strict routing. For example:
Loose routing: Record-Route: <sip:server10.biloxi.com;lr> Strict routing: Record-Route: <sip:bigbox3.site3.atlanta.com>
Mediant 2000 & Mediant 3000
Parameter Description
Web: LDAP Service
[LDAPServiceEnable]
Determines whether to enable the LDAP service.
[0] Disable (default) [1] Enable
Note: For this parameter to take effect, a device reset is required. Web: LDAP Server IP
[LDAPServerIP]
Defines the LDAP server's IP address in dotted-decimal notation (e.g., 192.10.1.255). The default is 0.0.0.0.
Web: LDAP Server Port [LDAPServerPort]
Defines the LDAP server's port number.
The valid value range is 0 to 65535. The default port number is 389. Web: LDAP Server Domain
Name
[LDAPServerDomainName ]
Defines the host name of the LDAP server.
Web: LDAP Password [LDAPPassword]
Defines the LDAP server's user password. Web: LDAP Bind DN
[LDAPBindDN]
Defines the LDAP server's bind DN. This is used as the username during connection and binding to the server.
For example:
LDAPBindDN = "CN=Search user,OU=Labs,DC=OCSR2,DC=local" Web: LDAP Search Dn
[LDAPSearchDN]
Defines the search DN for LDAP search requests. This is the top DN of the subtree where the search is performed. This parameter is mandatory for the search.
For example:
LDAPSearchHDN = "CN=Search user,OU=Labs,DC=OCSR2,DC=local" Web: LDAP Server Max
Respond Time
[LDAPServerMaxRespond Time]
Defines the time (in seconds) that the device waits for LDAP server responses.
The valid value range is 0 to 86400. The default is 3000.
[LDAPDebugMode] Determines whether to enable the LDAP task debug messages. This is used for providing debug information regarding LDAP tasks. The valid value range is 0 to 3. The default is 0.
Web: MS LDAP OCS Number attribute name [MSLDAPOCSNumAttribut eName]
The name of the attribute that represents the user OCS number in the Microsoft AD database.
The valid value is a string of up to 49 characters. The default is "msRTCSIP-PrimaryUserAddress".
Parameter Description [EnableRekeyAfter181]
Enables the device to send a Re-INVITE with a new (different) SRTP key (in the SDP) upon receipt of a SIP 181 response ("call is being forwarded").
[0] Disable (default) [1] Enable
Note: This parameter is applicable only if SRTP is used. Web: Min Routing Overlap
Digits
[MinOverlapDigitsForRout ing]
Minimum number of overlap digits to collect (for ISDN overlap dialing) before sending the first SIP message for routing Tel-to-IP calls. The valid value range is 0 to 49. The default is 1.
Note: This parameter is applicable when the ISDNRxOverlap parameter is set to [2].
Web: ISDN Overlap IP to Tel Dialing
[ISDNTxOverlap]
Enables ISDN overlap IP-to-Tel dialing. This feature is part of ISDN-to- SIP overlap dialing according to RFC 3578.
[0] Disable (default)
When enabled, for each received INVITE of the same dialog session, the device sends an ISDN Setup (and subsequent ISDN INFO Q.931 messages) with the collected digits to the Tel side. For all subsequent INVITEs received, the device sends a SIP 484 Address Incomplete response to maintain the current dialog session and receive additional digits from subsequent INVITEs.
[1] Enable
Note: When IP-to-Tel overlap dialing is enabled, to send ISDN Setup message without Sending Complete IE, the
CC_USER_SENDING_COMPLETE bit (=2) must be enabled for the ISDNOutgoingCallsBehavior parameter.
Web: 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.
[0] No (default) [1] Yes
When this 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 this parameter's value. The SIP cic parameter enables the transmission of the cic parameter from the SIP network to the ISDN (CIC=+<country code of
carrier><CIC of preferred interLATA carrier of caller>). 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 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:
INVITE sip:5550001;[email protected]:5060;user=phone SIP/2.0
Mediant 2000 & Mediant 3000
Parameter Description
cic+167895550001.
Note: After the cic prefix is added, the IP-to-Trunk Group 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 ISDN.
Web: Mute DTMF In Overlap
[MuteDTMFInOverlap]
Enables the muting of in-band DTMF detection until the device
receives the complete destination number from the ISDN (for Tel-to-IP calls). In other words, the device does not accept DTMF digits
received in the voice stream from the PSTN, but only accepts digits from PSTN INFO messages.
[0] Don't Mute (default)
[1] Mute DTMF in Overlap Dialing = The device ignores in-band DTMF digits received during ISDN overlap dialing (disables the DTMF in-band detector).
Notes:
When enabled and at least one digit is received from the ISDN (Setup), the device stops playing a dial tone.
Web: Fake Retry After [sec]
This parameter is applicable only to ISDN Overlap mode, when dialed numbers are sent using Q.931 INFO messages.
Determines whether the device, upon receipt of a SIP 503 response without a Retry-After header, behaves as if the 503 response includes a Retry-After header and with the period (in seconds) specified by this parameter. [FakeRetryAfter] [0] Disable Any
When enabled, this feature allows the device to operate with proxy servers that do not include the Retry-After SIP header in SIP 503 (Service Unavailable) responses to indicate an unavailable service.
positive value (in seconds) for enabling this feature
The Retry-After header is used with the 503 (Service Unavailable) response to indicate how long the service is expected to be
unavailable to the requesting SIP client. The device maintains a list of available proxies, by using the Keep-Alive (KA) mechanism. The device checks the availability of proxies by sending SIP OPTIONS every KA timeout to all proxies.
If the device receives a SIP 503 response to an INVITE, it also marks that the proxy is out of service for the defined "Retry-After" period. [TDMoIPInitiateInviteTime] Determines the time (in msec) between the first INVITE issued within
Parameter Description [UseBroadsoftDTG]
Determines whether the device uses the “dtg” parameter for routing IP-to-Tel calls to a specific Trunk Group.
[0] Disable (default) [1] Enable
When this parameter is enabled, if the Request URI in the received SIP INVITE includes the “dtg” parameter, the device routes the call to the Trunk Group according to its value. This parameter is used instead of the "tgrp/trunk-context" parameters. The "dtg" parameter appears in the INVITE Request URI (and in the To header).
For example, the received SIP message below routes the call to Trunk Group ID 56:
INVITE sip:[email protected];dtg=56;user=phone SIP/2.0 Note: If the Trunk Group is not found based on the "dtg" parameter, the IP to Trunk Group Routing table is used instead for routing the call to the appropriate Trunk Group.
Web: TGRP Routing Precedence
[TGRProutingPrecedence]
Determines the precedence method for routing IP-to-Tel calls - according to the IP to Trunk Group Routing table or tgrp/dtg parameters.
[0] (default) = IP-to-Tel routing is determined by the IP to Trunk Group Routing table (PSTNPrefix ini file parameter). If a matching rule is not found in this table, the device uses the Trunk Group parameters for routing the call.
[1] = The device first places precedence on the tgrp/dtg
parameters for IP-to-Tel routing. If the received INVITE request URI does not contain the tgrp/dtg parameters, or if the Trunk Group number is not defined, then the IP to Trunk Group Routing table is used for routing the call.
Below is an example of an INVITE request URI with the tgrp
parameter, indicating that the IP call should be routed to Trunk Group 7:
INVITE sip:200;tgrp=7;trunk-
[email protected];user=phone SIP/2.0 •
Notes:
• For IP-to-Tel routing based on the dtg parameter (instead of the tgrp parameter), use the parameter UseBroadsoftDTG
For enabling routing based on the SIP tgrp parameter, the UseSIPTgrp parameter must be set to 2.
Web: Enable Early 183 [EnableEarly183]
Determines whether the device sends a SIP 183 response with SDP to the IP immediately upon receipt of an INVITE message (for IP-to- Tel calls). The device sends the RTP packets only once it receives an ISDN Progress, Alerting with Progress indicator, or Connect message from the PSTN.
[0] Disable (default) [1] Enable
For example, if enabled and the device receives an ISDN Progress message, it starts sending RTP packets according to the initial negotiation without sending the 183 response again. Therefore, this feature reduces clipping of early media.
Mediant 2000 & Mediant 3000
Parameter Description
Notes:
To enable this feature, the EnableEarlyMedia parameter must also be set to 1.
This feature is applicable only to ISDN interfaces. Web: SIT Q850 Cause For
NC
[SITQ850CauseForNC]
Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-NC (No Circuit Found Special Information Tone) is detected from the PSTN for IP-to- Tel calls.
The valid range is 0 to 127. The default value is 34.
Note: When not configured (i.e., default), the SITQ850Cause parameter is used.
Web: SIT Q850 Cause For IC
[SITQ850CauseForIC]
Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-IC (Operator Intercept Special Information Tone) is detected from the PSTN for IP- to-Tel calls.
The valid range is 0 to 127. The default value is -1 (not configured). Note: When not configured (i.e., default), the SITQ850Cause parameter is used.
Web: SIT Q850 Cause For VC
[SITQ850CauseForVC]
Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-VC (Vacant Circuit - non-registered number Special Information Tone) is detected from the PSTN for IP-to-Tel calls.
The valid range is 0 to 127. The default value is -1 (not configured). Note: When not configured (i.e., default), the SITQ850Cause parameter is used.
Web: SIT Q850 Cause For RO
[SITQ850CauseForRO]
Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-RO (Reorder - System Busy Special Information Tone) is detected from the PSTN for IP-to-Tel calls.
The valid range is 0 to 127. The default value is -1 (not configured). Note: When not configured (i.e., default), the SITQ850Cause parameter is used.
Determines the SIP header used for obtaining the called (destination) number (for IP-to-Tel calls).
[SelectSourceHeaderForC alledNumber]
[0]
Request-URI header (default) = Obtains the destination number from the user part of the Request-URI.
Parameter Description
Determines whether the device performs an explicit unregister. [UnregistrationMode]
[0]
Disable (default) [1]
This parameter removes SIP UA registration bindings in a Registrar, according to RFC 3261. Registrations are soft state and expire unless refreshed, but can also be explicitly removed. A client can attempt to influence the expiration interval selected by the registrar. A UA requests the immediate removal of a binding by specifying an expiration interval of "0" for that contact address in a REGISTER request. UAs should support this mechanism so that bindings can be removed before their expiration interval has passed. Use of the "*" Contact header field value allows a registering UA to remove all bindings associated with an address-of-record (AOR) without knowing their precise values.
Enable = The device sends an asterisk (“*”) value in the Contact header, instructing the registrar server to remove all previous registration bindings.
Note: The REGISTER-specific Contact header field value of "*" applies to all registrations, but it can only be used if the Expires header field is present with a value of "0".
The maximum size (in bytes) threshold of logged, bundled (into a single UDP packet) Syslog messages, after which they are sent to a Syslog server.
[MaxBundleSyslogLength]
The valid value range is 0 to 1220 (where 0 indicates that no bundling occurs). The default is 1220.
Note: This parameter is applicable only if the GWDebugLevel parameter is set to 7.
[NotificationIPGroupID] Determines the IP Group ID to where the device sends SIP NOTIFY MWI messages.
Web: Coders Table/Coder Group Settings
This ini file table parameter defines the device's coders. Up to five groups of coders can be defined, where each group can consist of up