ISDN and CAS Interworking Parameters

The ISDN and CAS interworking parameters are described in the table below.

ISDN and CAS Interworking Parameters

Parameter

Description

ISDN Parameters

'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.

■ [0] Disable = (Default) If the SIP 200 OK includes the Date header, the device sends its value in the ISDN Connect Date / Time IE. If the 200 OK doesn't include this header, it doesn't add the Date / Time IE to the sent ISDN Connect message.
■ [1] Enable = If the SIP 200 OK includes the Date header, the device sends its value (i.e. date and time) in the ISDN Connect Date / Time IE. If the 200 OK doesn't include this header, the device uses its internal, local date and time for the Date / Time IE, which it adds to the sent ISDN Connect message.
■ [2] Always Send Local Date and Time = The device always sends its local date and time (obtained from its internal clock) to PBXs in ISDN Q.931 Connect messages (Date / Time IE). It does this regardless of whether or not the incoming SIP 200 OK includes the Date header. If the SIP 200 OK includes the Date header, the device ignores its value.

Note:

■ This feature is applicable only to Tel-to-IP calls.
■ For IP-to-Tel calls, the parameter is not applicable. Only if the incoming ISDN Connect message contains the Date / Time IE does the device add the Date header to the sent SIP 200 OK message.

'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].

'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.

■ [0] Disable (default)
■ [1] Through SIP = The device sends the first received digits from the initial INVITE to the Tel side in an ISDN Setup message. For each subsequently received re-INVITE message of the same dialog session, the device sends the collected digits to the Tel side in ISDN Info Q.931 messages. For each received re-INVITE, the device sends a SIP 484 Address Incomplete response to maintain the current dialog session and to receive additional digits from subsequent re-INVITEs.
■ [2] Through SIP INFO = The device sends the first received digits from the initial INVITE to the Tel side in an ISDN Setup message and then responds to the IP side with a SIP 183. For each subsequently received SIP INFO message with additional digits of the same dialog session, the device sends the collected digits to the Tel side in ISDN Info Q.931 messages. For each received SIP INFO, the device sends a SIP 200 OK response to maintain the current dialog session and to receive additional digits from subsequent INFOs.

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).

'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.

■ [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).

Note: The parameter is applicable to ISDN Overlap mode only when dialed numbers are sent using Q.931 Information messages.

[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).

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).

'Enable ISDN Tunneling Tel to IP'

configure voip > gateway digital settings > isdn-tnl-tel2ip

[EnableISDNTunnelingTel2IP]

Enables ISDN Tunneling.

■ [0] Disable (default).
■ [1] Using Header = Enable ISDN Tunneling from ISDN to SIP using a proprietary SIP header.
■ [2] Using Body = Enable ISDN Tunneling from ISDN to SIP using a dedicated message body.

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:

■ For this feature to function, configure the [ISDNDuplicateQ931BuffMode] parameter to 128 (i.e., duplicate all messages).
■ ISDN tunneling is applicable for all ISDN variants as well as QSIG.

'Enable ISDN Tunneling IP to Tel'

configure voip > gateway digital settings > isdn-tnl-ip2tel

[EnableISDNTunnelingIP2Tel]

Enables ISDN Tunneling for IP-to-Tel calls.

■ [0] Disable (default)
■ [1] Enable ISDN Tunneling from IP to ISDN

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.

'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.

configure voip > gateway digital settings > qsig-tunneling-mode

[QSIGTunnelingMode]

Defines the format of encapsulated QSIG message data in the SIP message MIME body.

■ [0] = (Default) ASCII presentation of Q.931 QSIG message.
■ [1] = Binary encoding of Q.931 QSIG message (according to ECMA-355, RFC 3204, and RFC 2025).

Note: The parameter is applicable only if the QSIG Tunneling feature is enabled, using the [EnableQSIGTunneling] parameter.

'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.

■ [0] Disable (default)
■ [1] Enable

Note:

■ The parameter is applicable to Euro ISDN variants - from TE (user) to NT (network).
■ If the parameter is disabled, the device plays a held tone to the Tel side when a SIP request with 0.0.0.0 or "inactive" in SDP is received. An appropriate CPT file with the held tone should be used.

'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.

■ [0] ASCII = (Default) IA5 format that allows up to 20 digits. Indicates that the 'isub' parameter value needs to be encoded using ASCII characters.
■ [1] BCD = (Binary Coded Decimal) - allows up to 40 characters (digits and letters). Indicates that the 'isub' parameter value needs to be encoded using BCD when translated to an ISDN message.
■ [2] User Specified

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.

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.

■ [0] = (Default) If an incoming ISDN Q.931 Setup message contains a Called/Calling Number Subaddress, the Subaddress is interworked to the SIP 'isub' parameter according to RFC.
■ [1] = The device removes the ISDN Subaddress and doesn't include the 'isub' parameter in the Request-URI and doesn't process INVITEs with the parameter.

[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:

1. If an isub parameter is received in the Request-URI, for example,
INVITE sip:9565645;isub=1234@host.domain:user=phone SIP/2.0
then the isub value is sent in the ISDN Setup message as the destination subaddress.
2. If the isub parameter is not received in the user part of the Request-URI, the device searches for it in the URI parameters of the To header, for example,
To: "Alex" <sip: 9565645@host.domain;isub=1234>
If present, the isub value is sent in the ISDN Setup message as the destination subaddress.
3. If the isub parameter is not present in the Request-URI header nor To header, the device does the following:
✔ If the called number (that appears in the user part of the Request-URI) starts with zero (0), for example,
INVITE sip:05694564@host.domain:user=phone SIP/2.0
then the device maps this called number to the destination number of the ISDN Setup message, and the destination subaddress in this ISDN Setup message remains empty.
✔ If the called number (that appears in the user part of the Request-URI) doesn't start with zero, for example,
INVITE sip:5694564@host.domain:user=phone SIP/2.0
then the device maps this called number to the destination number of the ISDN Setup message, and the destination subaddress in this ISDN Setup message then contains y digits from the end of the called number. The y number of digits can be configured using the ISUBNumberOfDigits parameter. The default value of ISUBNumberOfDigits is 0, thus, if the parameter is not configured, and 1) and 2) scenarios (described above) have not provided an isub value, the subaddress remains empty.

'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).
The range is any valid Q.931 release cause (0 to 127). The default is 0 (i.e., not configured - static mapping is used).

'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).

■ [0] Disable = (Default) CPC is not relayed between SIP and PSTN.
■ [1] Enable

The CPC is denoted in the PSTN message as follows:

■ ISDN PRI NI-2: In the Originating Line Information (OLI) Information Element (IE) of the ISDN Setup message.
■ MFC-R2: ANI II digits. The device supports the Brazilian and Argentinian variants. This regional support is configured using the CallingPartyCategoryMode.

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:

SIP CPC

NI-2 PRI

MFC-R2

Argentina

Brazil

ordinary

23

II-1

II-1

priority

n/a

II-2

II-2

data

n/a

II-6

II-6

test

n/a

II-3

II-3

operator

35

II-5

II-5

payphone

70

II-4

II-7

unknown

n/a

II-1

II-1

subscriber

23

n/a

II-1

cellular

61

II-13

n/a

locutorio

n/a

II-11

n/a

servicio-publico

n/a

II-12

n/a

red-privada-virtual / private-virtual-network

n/a

II-14

n/a

linea-especial / special-operator-handling-required

n/a

II-15

n/a

operadora-con-intervencion / telco-operator-handled-call

n/a

II-5

n/a

prison

29

n/a

n/a

hotel

66

n/a

n/a

cellular-roaming

63

n/a

n/a

Note: This feature is applicable only to the NI-2 PRI and E1 MFC-R2 variants.

'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.

■ [0] None (default)
■ [1] Brazil R2
■ [2] Argentina R2

Note:

■ To enable CPC mapping, set the EnableCallingPartyCategory parameter to 1.
■ The parameter is applicable only to the E1 MFC-R2 variant.

'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.

■ [0] No = (Default) IE's are not removed.
■ [1] Yes = IE's are removed.

'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.

■ [0] Disable = (Default) Does not remove Calling Name.
■ [1] Enable = Removes Calling Name.

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.

'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".

■ [0] Disable = (Default) The device restricts presentation in the outgoing SIP message. The device sends the SIP message with "anonymous" in the From header (e.g., From: "anonymous" <sip:anonymous@anonymous.invalid>).
■ [1] Enable = The device allows presentation in the outgoing SIP message (e.g., From: "Bob" <sip:12345@10.33.1.6>;tag=1c172113195).

Note: The parameter is applicable only to Tel-to-IP calls.

'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.

■ [0] Disable = The device restricts presentation in the outgoing SIP message. The device sends the SIP message with "anonymous" in the From header (e.g., From: "anonymous" <sip:anonymous@anonymous.invalid>).
■ [1] Enable = (Default) The device allows presentation in the outgoing SIP message (e.g., From: "Bob" <sip:12345@10.33.1.6>;tag=1c172113195).

Note: The parameter is applicable only to Tel-to-IP calls.

[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.

■ [0] = (Default) Connect message isn't sent after SIP 183 Session Progress message is received.
■ [1] = Connect message is sent after SIP 183 Session Progress message is received.

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).

■ [0] = Disable (default)
■ [1] = Enable

configure voip > sip-definition proxy-and-registration > redirect-in-facility

[SupportRedirectInFacility]

Determines whether the Redirect Number is retrieved from the Facility IE.

■ [0] = (Default) Not supported.
■ [1] = Supports partial retrieval of Redirect Number (number only) from the Facility IE in ISDN Setup messages. This is applicable to Redirect Number according to ECMA-173 Call Diversion Supplementary Services.

Note: To enable this feature, configure the [ISDNDuplicateQ931BuffMode] parameter to 1.

[EnableCIC]

Enables the relay of the Carrier Identification Code (CIC) to the ISDN.

■ [0] = (Default) Disabled - CIC is not relayed to the ISDN.
■ [1] = Enabled - CIC (received in the INVITE Request-URI) is relayed to the ISDN in the Transit Network Selection (TNS) IE of the Setup message. For example: INVITE sip:555666;cic=2345@100.2.3.4 sip/2.0.

Note:

■ This feature is supported only for SIP-to-ISDN calls.
■ The parameter AddCicAsPrefix can be used to add the CIC as a prefix to the destination phone number for routing IP-to-Tel calls.

'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.

■ [0] Disable (default)
■ [1] Enable

For more information on AOC, see Advice of Charge Services for Euro ISDN.

'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.

'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:

■ You can configure different IE data for Trunk Groups by defining the parameter for different IP Profile IDs (using the parameter IPProfile), and then assigning the required IP Profile ID in the IP-to-Tel Routing table (PSTNPrefix).
■ When IP Profiles are used for configuring different IE data for Trunk Groups, the parameter is ignored.

'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.

■ [0] Disable (default)
■ [1] Enable

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.

'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.

■ [0] Disable = (Default) Received UUI is not sent in ISDN message.
■ [1] Enable = The device interworks UUI from SIP to ISDN messages. The device supports the following SIP-to-ISDN interworking of UUI:
✔ SIP INVITE to Q.931 Setup
✔ SIP REFER to Q.931 Setup
✔ SIP 200 OK to Q.931 Connect
✔ SIP INFO to Q.931 User Information
✔ SIP 18x to Q.931 Alerting
✔ SIP BYE to Q.931 Disconnect

Note:

■ The interworking of ISDN User-to-User IE to SIP INFO is applicable only to the Euro ISDN, QSIG, and 4ESS ISDN variants.
■ To interwork the UUIE header from SIP-to-ISDN messages with the 4ESS ISDN variant, the ISDNGeneralCCBehavior parameter must be set to 16384.

[Enable911LocationIdIP2Tel]

Enables interworking of Emergency Location Identification from SIP to PRI.

■ [0] = Disabled (default)
■ [1] = Enabled

When enabled, the From header received in the SIP INVITE is translated into the following ISDN IE's:

■ Emergency Call Control.
■ Generic Information - to carry the Location Identification Number information.
■ Generic Information - to carry the Calling Geodetic Location information.

Note: The parameter is applicable only to the NI-2 ISDN variant.

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.

'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:

PSTN Protocol

Transfer Method (Described Below)

E1 Euro ISDN [1]

ECT [2] or InBand [5]

E1 QSIG [21],
T1 QSIG [23]

Single Step Transfer [4], Path Replacement Transfer [2], or InBand [5]

T1 NI2 ISDN [10],
T1 4ESS ISDN [11],
T1 5ESS 9 ISDN [12]

TBCT [2] or InBand [5]

T1 DMS-100 ISDN [14]

RTL [2] or InBand [5]

T1 RAW CAS [3], T1 CAS [2], E1 CAS [8], E1 RAW CAS [9]

[1] CAS NFA DMS-100 or [3] CAS Normal transfer

T1 DMS-100 Meridian ISDN [35]

RTL [2] or InBand [5]

The valid values of the parameter are described below:

■ [0] = Not supported (default).
■ [1] = Supports CAS NFA DMS-100 transfer. When a SIP REFER message is received, the device performs a Blind Transfer by executing a CAS Wink, waits for an acknowledged Wink from the remote side, dials the Refer-to number to the switch, and then releases the call.
Note: A specific NFA CAS table is required.
■ [2] = Supports ISDN PRI transfer - Release Link Trunk (RLT) (DMS-100), Two B Channel Transfer (TBCT) (NI2), Explicit Call Transfer (ECT) (EURO ISDN), and Path Replacement (QSIG). When a SIP REFER message is received, the device performs a transfer by sending Facility messages to the PBX with the necessary information on the call's legs to be connected. The different ISDN variants use slightly different methods (using Facility messages) to perform the transfer.
Note:
✔ For RLT ISDN transfer, the parameter SendISDNTransferOnConnect must be set to 1.
✔ The parameter SendISDNTransferOnConnect can be used to define if the TBCT/ECT transfer is performed after receipt of Alerting or Connect messages. For RLT, the transfer is always done after receipt of Connect (SendISDNTransferOnConnect is set to 1).
✔ This transfer can be performed between B-channels from different trunks or Trunk Groups, by using the parameter EnableTransferAcrossTrunkGroups.
✔ The device initiates the ECT process after receiving a SIP REFER message only for trunks that are configured to User side.
■ [3] = Supports CAS Normal transfer. When a SIP REFER message is received, the device performs a Blind Transfer by executing a CAS Wink, dialing the Refer-to number to the switch, and then releasing the call.
■ [4] = Supports QSIG Single Step transfer PRI :
✔ IP-to-Tel: When a SIP REFER message is received, the device performs a transfer by sending a Facility message to the PBX, initiating Single Step transfer. Once a success return result is received, the transfer is completed.
✔ Tel-to-IP: When a Facility message initiating Single Step transfer is received from the PBX, a SIP REFER message is sent to the IP side.
■ [5] = IP-to-Tel Blind Transfer mode supported for ISDN PRI protocols and implemented according to AT&T Toll Free Transfer Connect Service (TR 50075) “Courtesy Transfer-Human-No Data”. When the device receives a SIP REFER message, it performs a blind transfer by first dialing the DTMF digits (transfer prefix) defined by the parameter XferPrefixIP2Tel (configured to "*8" for AT&T service), and then (after 500 msec) the device dials the DTMF of the number (referred) from the Refer-To header sip:URI userpart. If the hostpart of the Refer-To sip:URI contains the device's IP address, and if the Trunk Group selected according to the IP-to-Tel Routing table is the same Trunk Group as the original call, then the device performs the in-band DTMF transfer; otherwise, the device sends the INVITE according to regular transfer rules.

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.

■ [6] = Supports AT&T toll free out-of-band blind transfer for trunks configured with the 4ESS ISDN protocol. AT&T courtesy transfer is a supplementary service which enables a user (e.g., user "A") to transform an established call between it and user "B" into a new call between users "B" and "C", whereby user "A" doesn't have a call established with user "C" prior to call transfer. The device handles this feature as follows:
✔ IP-to-Tel (user side): When a SIP REFER message is received, the device initiates a transfer by sending a Facility message to the PBX.
✔ Tel-to-IP (network side): When a Facility message initiating an out-of-band blind transfer is received from the PBX, the device sends a SIP REFER message to the IP side (if the EnableNetworkISDNTransfer parameter is set to 1).

Note: To configure trunk transfer mode per trunk, use the parameter TrunkTransferMode_x.

[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.

[EnableTransferAcrossTrunkGroups]

Determines whether the device allows ISDN ECT, RLT or TBCT IP-to-Tel call transfers between B-channels of different Trunk Groups.

■ [0] = (Default) Disable - ISDN call transfer is only between B-channels of the same Trunk Group.
■ [1] = Enable - the device performs ISDN transfer between any two PSTN calls (between any Trunk Group) handled by the device.

Note: The ISDN transfer also requires that you configure the parameter TrunkTransferMode_x to 2.

[TransferCapabilityForDataCalls]

Defines the ISDN Transfer Capability for data calls.

■ [0] = (Default) ISDN Transfer Capability for data calls is 64k unrestricted (data).
■ [1] = ISDN Transfer Capability for data calls is determined according to the ISDNTransferCapability parameter.

'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.

■ [0] Alert = (Default) Enables ISDN Transfer if the outgoing call is in Alerting or Connect state.
■ [1] Connect = Enables ISDN Transfer only if the outgoing call is in Connect state.

Note: For RLT ISDN transfer (TrunkTransferMode = 2 and ProtocolType = 14 DMS-100), the parameter must be set to 1.

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.

'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.

■ [0] Disable = Rejects ISDN transfer requests.
■ [1] Enable = (Default) The device sends a SIP REFER message to the remote call party if ECT/TBCT Facility messages are received from the ISDN side (e.g., from a PBX).

[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.

■ [0] = (Default) The device performs a hairpin TDM transfer upon ISDN call transfer.
■ [1] = Hairpin TDM transfer is disabled.

configure voip > gateway digital settings > isdn-ignore-18x-without-sdp

[ISDNIgnore18xWithoutSDP]

Enables interworking SIP 18x without SDP and ISDN Q.931 Progress/Alerting messages.

■ [0] = Disable. Incoming SIP 18x messages without SDP are replied by the device by PRACK (if required), but the device doesn't interwork these SIP messages with Q.931 Progress or Alerting messages (i.e., doesn't send to PSTN).
■ [1] = (Default) Enable. The device interworks 18x SIP messages with Q.931 Progress and Alerting messages (if required) and sends them to the PSTN.

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.

■ [0] = Disable. The device sends Progress messages to the trunk only if the trunk is configured as NT.
■ [1] = (Default) Enable. The device sends Q.931 Progress messages to the trunk if the trunk is configured as TE or NT.

Note: To configure the trunk's ISDN termination side (TE or NT), use the 'ISDN Termination Side' parameter.

'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.

■ [0] Disable = (Default) Ignores QSIG Facility messages with CallTranferComplete or CallTransferUpdate invokes.
■ [1] Enable

For example, assume A and C are PBX call parties and B is the SIP IP phone:

1. A calls B; B answers the call.
2. A places B on hold and calls C; C answers the call.
3. A performs a call transfer (the transfer is done internally by the PBX); B and C are connected to one another.

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:

■ For IP-to-Tel calls, the RedirectionNumber and RedirectionName in the CallTRansferComplete invoke is derived from the P-Asserted-Identity and Privacy headers in the received SIP INFO message.
■ To include the P-Asserted-Identity header in outgoing SIP UPDATE messages, set the AssertedIDMode parameter to Add P-Asserted-Identity.

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.

■ [0] =(Default) No interworking of No-ID cause.
■ [1] = Interwork No-ID cause only from IP to Tel.
■ [2] = Interwork No-ID cause only from Tel to IP.
■ [3] = Interwork No-ID cause from IP-to-Tel side and Tel-to-IP side.

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):

■ Unavailable: IE[03]=1c 11 91 a1 0e 02 01 00 06 06 02 83 38 66 01 01 0a 01 00
■ Anonymous: IE[03]=1c 11 91 a1 0e 02 01 00 06 06 02 83 38 66 01 01 0a 01 01
■ Interaction with other service: IE[03]=1c 11 91 a1 0e 02 01 00 06 06 02 83 38 66 01 01 0a 01 02
■ Coin line/payphone: IE[03]=1c 11 91 a1 0e 02 01 00 06 06 02 83 38 66 01 01 0a 01 03

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).

is-cas-sndhook-flsh

[CASSendHookFlash]

Enables sending Wink signal toward CAS trunks.

■ [0] = Disable (default)
■ [1] = Enable

If the device receives a mid-call SIP INFO message with flashhook event body (as shown below) and the parameter is set to 1, the device generates a wink signal toward the CAS trunk. The CAS wink signal is done by changing the A bit from 1 to 0, and then back to 1 for 450 msec.

INFO sip:4505656002@192.168.13.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.13.2:5060
From: <sip:06@192.168.13.2:5060>
To: <sip:4505656002@192.168.13.40:5060>;tag=132878796-1040067870294
Call-ID: 0010-0016-D69A7DA8-1@192.168.13.2
CSeq:2 INFO
Content-Type: application/broadsoft
Content-Length: 17
event flashhook

Note: The parameter is applicable only to T1 CAS protocols.

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.

■ [0] = (Default) Disable. The device doesn't add the XML body containing CUG information to the outgoing SIP INVITE message.
■ [1] = Enable. The device adds the XML body containing CUG information 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"?>
<xs:schema xmlns:xs="http://www.w3.org/2001/XMLSchema"
xmlns="http://uri.etsi.org/ngn/params/xml/simservs/xcap"
targetNamespace="http://uri.etsi.org/ngn/params/xml/simservs/xcap" elementFormDefault=" qualified"
attributeFormDefault="unqualified">
<xs:annotation>
<xs:documentation>XML Schema Definition for the closed user group
parameter</xs:documentation>
</xs:annotation>
<xs:include schemaLocation="xcap.xsd"/>
<!--Definition of simple types-->
<xs:simpleType name="twobitType">
<xs:restriction base="xs:string">
<xs:pattern value="[0-1][0-1]"/>
</xs:restriction>
</xs:simpleType>
<xs:simpleType name="networkIdentityType">
<xs:restriction base="xs.hexBinary">
<xs.length value="2"/>
</xs:restriction>
</xs:simpleType>
<xs:simpleType name="sixteenbitType">
<xs:restriction base="xs:hexBinary">
<xs:length value="2"/>
</xs:restriction>
</xs:simpleType>
<xs:simpleType name="cugIndexType">
<xs:restriction base="xs:integer">
<xs:minInclusive value="0"/>
<xs:maxInclusive value="32767"/>
</xs:restriction>
</xs:simpleType>
<!--Definition of complex types-->
<xs:complexType name="cugRequestType">
<xs:sequence>
<xs:element name="outgoingAccessRequest" type="xs:boolean"/>
<xs:element name="cugIndex" type="cugIndexType" minOccurs="0"/>
</xs:sequence>
</xs:complexType>
<!--Definition of document structure-->
<xs:element name="cug" substitutionGroup="ss:absService">
<xs:complexType>
<xs:complexContent>
<xs:extension base="ss:simservType">
<xs:sequence>
<xs:element name="cugCallOperation" type="cugRequestType" minOccurs="0">
<xs:complexType>
<xs:sequence>
<xs:element name="outgoingAccessRequest" type="xs:boolean" value="True"/>
<xs:element name="cugIndex" type="xs:integer" value="32767"/>
</xs:sequence>
</xs:complexType>
</xs:element>
<xs:element name="networkIndicator" type="networkIdentityType" minOccurs="0"/>
<xs:element name="cugInterlockBinaryCode" type="sixteenbitType" minOccurs="0"/>
<xs:element name="cugCommunicationIndicator" type="twobitType" minOccurs="0"/>
</xs:sequence>
</xs:extension>
</xs:complexContent>
</xs:complexType>
</xs:element>
</xs:schema>