ISDN Interworking Parameters
The ISDN interworking parameters are described in the table below.
ISDN Interworking Parameters
Parameter |
Description |
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ISDN Parameters |
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'Send Local Time To ISDN Connect' [SendLocalTimeToISDNConnect] |
Determines the device's handling of the date and time sent in the ISDN Connect message (Date / Time IE) upon receipt of SIP 200 OK messages.
Note:
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'Min Routing Overlap Digits' configure voip > gateway dtmf-supp-service dtmf-and-dialing > min-dg-b4-routing [MinOverlapDigitsForRouting] |
Defines the 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: The parameter is applicable when the ISDNRxOverlap parameter is set to [2] or [3]. |
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'ISDN Overlap IP to Tel Dialing' configure voip > gateway dtmf-supp-service dtmf-and-dialing > isdn-tx-overlap [ISDNTxOverlap] |
Enables ISDN overlap dialing for IP-to-Tel calls. This feature is part of ISDN-to-SIP overlap dialing according to RFC 3578.
Note: When IP-to-Tel overlap dialing is enabled, to send ISDN Setup messages without the Sending Complete IE, the ISDNOutCallsBehavior parameter must be set to USER SENDING COMPLETE (2). |
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'Mute DTMF In Overlap' configure voip > gateway dtmf-supp-service supp-service-settings > 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 doesn't accept DTMF digits received in the voice stream from the PSTN, but only accepts digits from ISDN Info messages.
Note: The parameter is applicable to ISDN Overlap mode only when dialed numbers are sent using Q.931 Information messages. |
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[ConnectedNumberType] |
Defines the Numbering Type of the ISDN Q.931 Connected Number IE that the device sends in the Connect message to the ISDN (for Tel-to-IP calls). This is interworked from the P-Asserted-Identity header in SIP 200 OK. The default is [0] (i.e., unknown). |
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configure voip > gateway dtmf-supp-service supp-service-settings > connected-number-type [ConnectedNumberPlan] |
Defines the Numbering Plan of the ISDN Q.931 Connected Number IE that the device sends in the Connect message to the ISDN (for Tel-to-IP calls). This is interworked from the P-Asserted-Identity header in SIP 200 OK. The default is [0] (i.e., unknown). |
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'Enable ISDN Tunneling Tel to IP' configure voip > gateway digital settings > isdn-tnl-tel2ip [EnableISDNTunnelingTel2IP] |
Enables ISDN Tunneling.
When ISDN Tunneling is enabled, the device sends all ISDN messages using the correlated SIP messages. The ISDN Setup message is tunneled using SIP INVITE, all mid-call messages are tunneled using SIP INFO, and ISDN Disconnect/Release message is tunneled using SIP BYE messages. The raw data from the ISDN is inserted into a proprietary SIP header (X-ISDNTunnelingInfo) or a dedicated message body (application/isdn) in the SIP messages. Note:
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'Enable ISDN Tunneling IP to Tel' configure voip > gateway digital settings > isdn-tnl-ip2tel [EnableISDNTunnelingIP2Tel] |
Enables ISDN Tunneling for IP-to-Tel calls.
When ISDN Tunneling is enabled, the device extracts raw data received in the proprietary SIP header, x-isdntunnelinginfo, or a dedicated message body (application/isdn) in the SIP message and then sends the data in an ISDN message to the PSTN. If the raw data in this SIP header is suffixed with the string "ADDE", then the raw data is extracted and added as Informational Elements (IE) in the outgoing Q.931 message. The tunneling of the x-isdntunnelinginfo SIP header with IEs is converted from INVITE, 180, and 200 OK SIP messages to Q.931 SETUP, ALERT, and CONNECT respectively. For example, if the following SIP header is received, x-isdntunnelinginfo: ADDE1C269FAA 06 800100820100A10F020136 0201F0A00702010102021F69 then it is added as an IE to the outgoing Q.931 message as 1C269FAA 06 800100820100A10F020136 0201F0A00702010102021F69, where, for example, "1C269F" is a 26 byte length Facility IE. Note: The feature is similar to that of the AddIEinSetup parameter. If both parameters are configured, the AddIEinSetup parameter is ignored. |
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'Enable QSIG Tunneling' configure voip > gateway digital settings > qsig-tunneling [EnableQSIGTunneling] |
Global parameter that enables QSIG tunneling-over-SIP for all calls. You can also configure this feature per specific calls, using IP Profiles ('QSIG Tunneling' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles. Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile. |
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configure voip > gateway digital settings > qsig-tunneling-mode [QSIGTunnelingMode] |
Defines the format of encapsulated QSIG message data in the SIP message MIME body.
Note: The parameter is applicable only if the QSIG Tunneling feature is enabled, using the [EnableQSIGTunneling] parameter. |
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'Enable Hold to ISDN' configure voip > gateway dtmf-supp-service supp-service-settings > hold-to-isdn [EnableHold2ISDN] |
Enables SIP-to-ISDN interworking of the Hold/Retrieve supplementary service.
Note:
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'ISDN SubAddress Format' configure voip > gateway digital settings > isdn-subaddr-frmt [ISDNSubAddressFormat] |
Determines the encoding format of the SIP Tel URI parameter 'isub', which carries the encoding type of ISDN subaddresses. This is used to identify different remote ISDN entities under the same phone number (ISDN Calling and Called numbers) for interworking between ISDN and SIP networks.
For IP-to-Tel calls, if the incoming SIP INVITE message includes subaddress values in the 'isub' parameter for the Called Number (in the Request-URI) and/or the Calling Number (in the From header), these values are mapped to the outgoing ISDN Setup message. If the incoming ISDN Setup message includes 'subaddress' values for the Called Number and/or the Calling Number, these values are mapped to the outgoing SIP INVITE message's ‘isub’ parameter in accordance with RFC 4715. |
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configure voip > gateway dtmf-supp-service supp-service-settings > ignore-isdn-subaddress [IgnoreISDNSubaddress] |
Determines whether the device ignores the Subaddress from the incoming ISDN Called and Calling numbers when sending to IP.
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[ISUBNumberOfDigits] |
Defines the number of digits (from the end) that the device takes from the called number (received from the IP) for the isub number (in the sent ISDN Setup message). This feature is applicable only for IP-to-ISDN calls. The valid value range is 0 to 36. The default is 0. This feature operates as follows:
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'Default Cause Mapping From ISDN to SIP' configure voip > gateway digital settings > dflt-cse-map-isdn2sip [DefaultCauseMapISDN2IP] |
Defines a single default ISDN release cause that is used (in ISDN-to-IP calls) instead of all received release causes, except when the following Q.931 cause values are received: Normal Call Clearing (16), User Busy (17), No User Responding (18), or No Answer from User (19). |
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'Enable Calling Party Category' configure voip > gateway digital settings > ni2-cpc [EnableCallingPartyCategory] |
Enables the mapping of the calling party category (CPC) between the incoming PSTN message and outgoing SIP message, and vice versa (i.e., for IP-to-Tel and Tel-to-IP calls). The CPC characterizes the station used to originate a call (e.g., a payphone or an operator).
The CPC is denoted in the PSTN message as follows:
The CPC is denoted in the SIP INVITE message using the 'cpc=' parameter in the From or P-Asserted-Identity headers. For example, the 'cpc=' parameter in the below INVITE message is set to "payphone": INVITE sip:bob@biloxi.example.com SIP/2.0 To: "Bob" <sip:bob@biloxi.example.com> From: <tel:+17005554141;cpc=payphone>;tag=1928301774 The table below shows the mapping of CPC between SIP and PSTN:
Note: This feature is applicable only to the NI-2 PRI and E1 MFC-R2 variants. |
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'Calling Party Category Mode' configure voip > gateway digital settings > cpc-mode [CallingPartyCategoryMode] |
Defines the regional Calling Party Category (CPC) mapping variant between SIP and PSTN for MFC-R2.
Note:
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'Remove CLI when Restricted' configure voip > gateway digital settings > rmv-cli-when-restr [RemoveCLIWhenRestricted] |
Determines (for IP-to-Tel calls) whether the Calling Number and Calling Name IEs are removed from the ISDN Setup message if the presentation is set to Restricted.
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'Remove Calling Name' configure voip > gateway digital settings > rmv-calling-name [RemoveCallingName] |
Enables the device to remove the Calling Name from SIP-to-ISDN calls for all trunks.
Note: Some PSTN switches / PBXs may not be configured to support the receipt of the “Calling Name” information. These switches might respond to an ISDN Setup message (including the Calling Name) with an ISDN "REQUESTED_FAC_NOT_SUBSCRIBED" failure. The parameter can be set to Enable (1) to remove the “Calling Name” from SIP-to-ISDN calls and allow the call to proceed. |
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'CID Notification' configure voip > gateway digital settings > cid-notification [CIDNotification] |
Enables presentation in the outgoing SIP message when the presentation indicator in the Calling Party Number information element of the incoming ISDN message has the value "number not available due to interworking".
Note: The parameter is applicable only to Tel-to-IP calls. |
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'CID Not Included Notification' configure voip > gateway digital settings > cid-not-included-notification [CIDNotIncludedNotification] |
Enables presentation in the outgoing SIP message when the Calling Party Number information element of the incoming ISDN message doesn't include the presentation indicator.
Note: The parameter is applicable only to Tel-to-IP calls. |
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[ConnectOnProgressInd] |
Enables the play of announcements from IP to Tel without the need to answer the Tel-to-IP call. It can be used with PSTN networks that don't support the opening of a TDM channel before an ISDN Connect message is received.
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configure voip > gateway dtmf-supp-service supp-service-settings > snd-isdn-ser-aftr-restart [SendISDNServiceAfterRestart] |
Enables the device to send an ISDN Service message per trunk upon device restart. The message (transmitted on the trunk's D-channel) indicates the availability of the trunk's B-channels (i.e., trunk in service).
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configure voip > sip-definition proxy-and-registration > redirect-in-facility [SupportRedirectInFacility] |
Determines whether the Redirect Number is retrieved from the Facility IE.
Note: To enable this feature, configure the [ISDNDuplicateQ931BuffMode] parameter to 1. |
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[EnableCIC] |
Enables the relay of the Carrier Identification Code (CIC) to the ISDN.
Note:
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'AoC Support' configure voip > gateway dtmf-supp-service supp-service-settings > aoc-support [EnableAOC] |
Enables the interworking of ISDN Advice of Charge (AOC) messages to SIP.
For more information on AOC, see Advice of Charge Services for Euro ISDN. |
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'Add IE in SETUP' configure voip > gateway digital settings > add-ie-in-setup [AddIEinSetup] |
Global parameter that defines an optional Information Element (IE) data (in hex format) to add to ISDN Setup messages. You can also configure this feature per specific calls, using IP Profiles ('Add IE In Setup' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles. Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile. |
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'Trunk Groups to Send IE' configure voip > gateway digital settings > trkgrps-to-snd-ie [SendIEonTG] |
Defines Trunk Group IDs (up to 50 characters) from where the optional ISDN IE (defined by the parameter AddIEinSetup) is sent. For example: '1,2,4,10,12,6'. Note:
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'Enable User-to-User IE for Tel to IP' configure voip > gateway digital settings > uui-ie-for-tel2ip [EnableUUITel2IP] |
Enables transfer of User-to-User (UU) IE from ISDN to SIP.
The device supports the following ISDN-to-SIP interworking: Setup to SIP INVITE, Connect to SIP 200 OK, User Information to SIP INFO, Alerting to SIP 18x response, and Disconnect to SIP BYE response messages. Note: The interworking of ISDN User-to-User IE to SIP INFO is applicable only to the Euro ISDN, QSIG, and 4ESS ISDN variants. |
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'Enable User-to-User IE for IP to Tel' configure voip > gateway digital settings > uui-ie-for-ip2tel [EnableUUIIP2Tel] |
Enables interworking of SIP user-to-user information (UUI) to User-to-User IE in ISDN Q.931 messages.
Note:
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[Enable911LocationIdIP2Tel] |
Enables interworking of Emergency Location Identification from SIP to PRI.
When enabled, the From header received in the SIP INVITE is translated into the following ISDN IE's:
Note: The parameter is applicable only to the NI-2 ISDN variant. |
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configure voip > gateway digital settings > early-answer-timeout [EarlyAnswerTimeout] |
Global parameter that defines the duration (in seconds) that the device waits for an ISDN Connect message from the called party (Tel side), started from when it sends a Setup message. You can also configure this feature per specific calls, using IP Profiles ('Early Answer Timeout' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles. Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile. |
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'Trunk Transfer Mode' configure voip > interface e1-t1 > trk-xfer-mode-type [TrunkTransferMode] |
Determines the trunk transfer method (for all trunks) when a SIP REFER message is received. The transfer method depends on the Trunk's PSTN protocol (configured by the parameter ProtocolType) and is applicable only when one of these protocols are used:
The valid values of the parameter are described below:
After completing the in-band transfer, the device waits for the ISDN Disconnect message. If the Disconnect message is received during the first 5 seconds, the device sends a SIP NOTIFY with 200 OK message; otherwise, the device sends a NOTIFY with 4xx message.
Note: To configure trunk transfer mode per trunk, use the parameter TrunkTransferMode_x. |
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[TrunkTransferMode_x] |
Determines the trunk transfer mode per trunk (where x denotes the Trunk number). To configure trunk transfer mode for all trunks and for a description of the parameter options, refer to the parameter TrunkTransferMode. |
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[EnableTransferAcrossTrunkGroups] |
Determines whether the device allows ISDN ECT, RLT or TBCT IP-to-Tel call transfers between B-channels of different Trunk Groups.
Note: The ISDN transfer also requires that you configure the parameter TrunkTransferMode_x to 2. |
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[TransferCapabilityForDataCalls] |
Defines the ISDN Transfer Capability for data calls.
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'ISDN Transfer On Connect' configure voip > gateway digital settings > isdn-trsfr-on-conn [SendISDNTransferOnConnect] |
The parameter is used for the ECT/TBCT/RLT/Path Replacement ISDN transfer methods. Usually, the device requests the PBX to connect an incoming and outgoing call. The parameter determines if the outgoing call (from the device to the PBX) must be connected before the transfer is initiated.
Note: For RLT ISDN transfer (TrunkTransferMode = 2 and ProtocolType = 14 DMS-100), the parameter must be set to 1. |
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configure voip > gateway dtmf-supp-service supp-service-settings > isdn-xfer-complete-timeout [ISDNTransferCompleteTimeout] |
Defines the timeout (in seconds) for determining ISDN call transfer (ECT, RLT, or TBCT) failure. If the device doesn't receive any response to an ISDN transfer attempt within this user-defined time, the device identifies this as an ISDN transfer failure and subsequently performs a hairpin TDM connection or sends a SIP NOTIFY message with a SIP 603 response (depending whether hairpin is enabled or disabled, using the parameter DisableFallbackTransferToTDM). The valid range is 1 to 10. The default is 4. |
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'Enable Network ISDN Transfer' configure voip > sip-definition settings > network-isdn-xfer [EnableNetworkISDNTransfer] |
Determines whether the device allows interworking of network-side received ECT/TBCT Facility messages (NI-2 TBCT - Two B-channel Transfer and ETSI ECT - Explicit Call Transfer) to SIP REFER.
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[DisableFallbackTransferToTDM] |
Enables "hairpin" TDM transfer upon ISDN (ECT, RLT, or TBCT) call transfer failure. When this feature is enabled and an ISDN call transfer failure occurs, the device sends a SIP NOTIFY message with a SIP 603 Decline response.
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configure voip > gateway digital settings > isdn-ignore-18x-without-sdp [ISDNIgnore18xWithoutSDP] |
Enables interworking SIP 18x without SDP and ISDN Q.931 Progress/Alerting messages.
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configure voip > gateway digital settings > isdn-send-progress-for-te [ISDNSendProgressForTE] |
Defines whether the device sends Q.931 Progress messages to the ISDN trunk if the trunk is configured as User side (TE) and/or Network (NT) side, for IP-to-Tel calls.
Note: To configure the trunk's ISDN termination side (TE or NT), use the 'ISDN Termination Side' parameter. |
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'Enable QSIG Transfer Update' configure voip > gateway digital settings > qsig-xfer-update [EnableQSIGTransferUpdate] |
Determines whether the device interworks QSIG Facility messages with CallTranferComplete or CallTransferUpdate invoke application protocol data units (APDU) to SIP UPDATE messages with P-Asserted-Identity and optional Privacy headers. This feature is supported for IP-to-Tel and Tel-to-IP calls.
For example, assume A and C are PBX call parties and B is the SIP IP phone:
In the above example, the PBX updates B that it is now talking with C. The PBX updates this by sending a QSIG Facility message with CallTranferComplete invoke APDU. The device interworks this message to a SIP UPDATE message containing a P-Asserted-Identity header with the number and name derived from the QSIG CallTranferComplete RedirectionNumber and RedirectionName. Note:
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configure voip > gateway digital settings > isdn-ntt-noid-interworking-mode [ISDNNttNoidInterworkingMode] |
Defines SIP-ISDN interworking between NTT Japan's No-ID cause in the Facility information element (IE) of the ISDN Setup message, and the calling party number (display name) in the From header of the SIP INVITE message. The No ID cause in the Facility IE indicates one of four reasons (see list of mapping below), for example, why the call was blocked.
The following lists the mapping between the SIP display name in the From header and the cause of the Facility IE in the ISDN Setup message (SIP:ISDN):
Below shows an example of an ISDN No-ID cause mapped to SIP for "Interaction with other service": From: "Interaction with other service" <sip:anonymous@anonymous.invalid;pstn-params=9082828088>;tag=gK09696ce6 Note: The parameter is applicable only to Trunks configured with the JAPAN NTT ISDN PRI (T1) protocol variant (i.e., [ProtocolType] parameter configured to 16). |
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configure voip > gateway digital settings > cug-data-mode [CugDataMode] |
Enables interworking between the ISDN Closed User Group (CUG) supplementary service and SIP, for Tel-to-IP calls. The CUG supplementary service enables users to form groups, where members of a specific closed user group can communicate among themselves but not, in general, with users outside the group. If the parameter is enabled and the device receives an ISDN Setup message whose Facility IE indicates CUG (cUGCall invoke), it adds an XML body containing CUG information (CUG index and outgoing access) to the outgoing SIP INVITE message.
The following shows an example of an added XML body containing CUG information: <?xml version="1.0" encoding="utf-8"?> |