Contact Center provisioning configures Cisco Unified Communications Manager (CUCM) to communicate with Contact Center.

How to Set Up a Contact Center Server

Field	Description
Contact Center Server Name	Unique server name. This field is mandatory.
Description	Server description.
Cisco Unified Communications Manager	The Cluster you want to use for Contact Center Server. This field is mandatory.
Transfer Conference Pattern	Transfer conference pattern used when transferring calls between agents. This field is mandatory.
Network VRU	Pattern used to route calls to a CVP. This field is mandatory.
SIP Trunks	This field is mandatory. See fields below:
Trunk Destination Type	CVP or CUBE (ENT) or CUSP SIP Trunk. This field is mandatory. Note: Both CVP and CUBE (ENT) trunks must be added for this Contact Center Server to be added successfully.
Trunk Destination Address	The destination address of the CVP or CUBE (ENT) or CUSP SIP Trunk. This field is mandatory. Multiple destination addresses & ports can be added for each trunk type.
Trunk Destination Port	The destination port of the CVP or CUBE (ENT) or CUSP SIP Trunk, if no value provided system takes 5060 as default.
Trunk Security Profile	The SIP trunk Security Profile that needs to be used by each trunk. This field is mandatory.
SIP Profile	The SIP trunk profile that needs to be used by each trunk. This field is mandatory.

For 500/1000/4000/12K/SCC - You must provide information for a CVP and a CUBE (ENT) SIP Trunk. For Small Contact Center, both the CVP and CUBE (ENT) trunks should have the same IP address with a different Trunk Security Profile selected in the Trunk Security Profile drop-down for each trunk.

For CUSP - You must provide information for a CUSP SIP Trunk. Only one trunk type can be added.

Enable or Disable the Built-in-Bridge

Prerequisites: Overview

Built-in-Bridge

Built-in-Bridge (BIB) is not enabled by default for the phones. It is disabled at the system level as it is not used by all customers by default. It is used only by customers with Contact Center.

The provider has to perform the following procedures to enable BIB for the customers having contact center.

Built-in-Bridge (provider deployment)

Built-in-Bridge (BIB) is not enabled by default for the phones. It is disabled at the system level as it is not used by all the customer by default. It is used only by the customers having Contact Center.

The provider has to perform the following procedures to enable BIB for the customers having contact center.

Configure the Built-in-Bridge

Configure the Built-in-Bridge (provider deployment)

Model Details: relation/HcsCCServerREL

An asterisk * in the title indicates the field is mandatory.
If a field has a default value, it is shown in the Description column.
If the field type is an array, the Field Name has a .[n] suffix, where n is the array index placeholder.
Object and array field names are listed to provide context. If a field belongs to an object or an array, the full field name is shown in dot separated notation.

Title						Description	Details
Contact Center Server Name *						Contact Center Server name to distinguish each instance	Field Name: name Type: String MaxLength: 512 Pattern: ^(\S)+$
Internal ID						Internal ID of Contact Center Server Default: Auto Generated	Field Name: InternalCCServerId Type: String Default: Auto Generated MaxLength: 1024
Description							Field Name: description Type: String MaxLength: 512
CUCM *						The CUCM device associated with this Contact Center Server.	Field Name: networkDevice Type: String Target: data/CallManager MaxLength: 1024 Format: uri
Transfer Conference Pattern *						Transfer conference pattern used when transfering calls between agents Default: 8XXXX	Field Name: transConfPattern Type: String Default: 8XXXX MaxLength: 512
Network VRU *						Pattern used to send request from CUCM to CVP Default: 8881111000	Field Name: vru Type: String Default: 8881111000 MaxLength: 512
	SIP Trunks						Field Name: trunkArray.[n] Type: Array
		Trunk Name				Hidden value to store Sip Trunk name for ease of use later on	Field Name: trunkArray.[n].trunkName Type: String MaxLength: 1024
		Trunk Destination Type *				Trunk destination type which can be either to a CVP, CUBE or CUSP	Field Name: trunkArray.[n].trunkType Type: String MaxLength: 1024 Choices: ["CVP", "CUBEE", "CUSP"]
			Destination Addresses				Field Name: destinationAddresses.[n] Type: Array
				Trunk Destination Address *		Trunk destination address to either a CVP, CUBE or CUSP	Field Name: trunkArray.[n].destinationAddresses.[n].trunkDestinationAddr Type: String MaxLength: 1024
				Trunk Destination Port		Trunk destination port to either a CVP, CUBE or CUSP	Field Name: trunkArray.[n].destinationAddresses.[n].trunkDestinationPort Type: String MaxLength: 1024
		Trunk Security Profile *				SIP Trunk Security Profile for CC Server related SIP trunk	Field Name: trunkArray.[n].trunkSecurityProfileName Type: String Target: device/cucm/SipTrunkSecurityProfile Target attr: name MaxLength: 1024 Format: uri
		SIP Profile *				SIP Trunk Profile for CC Server related SIP trunk	Field Name: trunkArray.[n].trunkProfileName Type: String Target: device/cucm/SipProfile Target attr: name MaxLength: 1024 Format: uri
	Sip Trunk						Field Name: sipTrunk.[n] Type: Array Cardinality: [1..n]
		Device Protocol *				Protocol option is read-only,except when creating a device.	Field Name: sipTrunk.[n].protocol Type: String Cardinality: [1..1] Choices: ["SCCP", "Digital Access PRI", "H.225", "Analog Access", "Digital Access T1", "Route Point", "Unicast Bridge", "Multicast Point", "Inter-Cluster Trunk", "RAS", "Digital Access BRI", "SIP", "MGCP", "Static SIP Mobile Subscriber", "SIP Connector", "Remote Destination", "Mobile Smart Client", "Digital Access E1 R2", "CTI Remote Device", "Protocol Not Specified"]
		Prefix DN				Enter the prefix digits that are appended to the called party number on incoming calls. Cisco Unified Communications Manager adds prefix digits after first truncating the number in accordance with the Significant Digits setting. You can enter the international escape character +.	Field Name: sipTrunk.[n].prefixDn Type: ["String", "Null"] Cardinality: [0..1] MaxLength: 50 Pattern: ^[0-9*#+]{0,50}$
		Calling Line ID Presentation				Cisco Unified Communications Manager uses calling line ID presentation (CLIP) as a supplementary service to provide the calling party number. The SIP trunk level configuration takes precedence over the call-by-call configuration. The default value for Calling Line ID Presentation specifies Default, which translates to Allowed. Choose Default if you want Cisco Unified Communications Manager to send calling number information. Choose Restricted if you do not want Cisco Unified Communications Manager to send the calling number information. Default: Default	Field Name: sipTrunk.[n].callingLineIdPresentation Type: String Cardinality: [0..1] Default: Default Choices: ["Default", "Allowed", "Restricted"]
		Maintain Original Caller ID DN and Caller Name in Identity Headers				This tag when enabled will prevent external presentation name and external presentation number to be populated in SIP headers like P-Asserted ID and RPID.	Field Name: sipTrunk.[n].useCallerIdCallerNameinUriOutgoingRequest Type: Boolean Cardinality: [0..1]
		Redirecting Diversion Header Delivery - Inbound				Check this check box to accept the Redirecting Number in the incoming INVITE message to the Cisco Unified Communications Manager. Uncheck the check box to exclude the Redirecting Number in the incoming INVITE message to the Cisco Unified Communications Manager. You use Redirecting Number for voice-messaging integration only. If your configured voice-messaging system supports Redirecting Number, you should check the check box. The default value for Redirecting Number IE Deliver - Inbound specifies not checked.	Field Name: sipTrunk.[n].acceptInboundRdnis Type: Boolean Cardinality: [0..1]
		Calling Search Space				From the drop-down list box, choose the appropriate calling search space for the trunk. The calling search space specifies the collection of route partitions that are searched to determine how to route a collected (originating) number. You can configure the number of items that display in this drop-down list box by using the Max List Box Items enterprise parameter. If more calling search spaces exist than the Max List Box Items enterprise parameter specifies, the Find button displays next to the drop-down list box. Click the Find button to display the Find and List Calling Search Spaces window. Find and choose a calling search space name. Note To set the maximum list box items, choose System > Enterprise Parameters and choose CCMAdmin Parameters. The default value for Calling Search Space specifies None.	Field Name: sipTrunk.[n].callingSearchSpaceName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Use Trusted Relay Point				From the drop-down list box, enable or disable whether Cisco Unified Communications Manager inserts a trusted relay point (TRP) device with this media endpoint. Choose one of the following values: Default—If you choose this value, the device uses the Use Trusted Relay Point setting from the common device configuration with which this device associates. Off—Choose this value to disable the use of a TRP with this device. This setting overrides the Use Trusted Relay Point setting in the common device configuration with which this device associates. On—Choose this value to enable the use of a TRP with this device. This setting overrides the Use Trusted Relay Point setting in the common device configuration with which this device associates. A Trusted Relay Point (TRP) device designates an MTP or transcoder device that is labeled as Trusted Relay Point. Cisco Unified Communications Manager places the TRP closest to the associated endpoint device if more than one resource is needed for the endpoint (for example, a transcoder or RSVPAgent). If both TRP and MTP are required for the endpoint, TRP gets used as the required MTP. See the Cisco Unified Communications Manager System Guide for details of call behavior. If both TRP and RSVPAgent are needed for the endpoint, Cisco Unified Communications Manager first tries to find an RSVPAgent that can also be used as a TRP. If both TRP and transcoder are needed for the endpoint, Cisco Unified Communications Manager first tries to find a transcoder that is also designated as a TRP. See the Cisco Unified Communications Manager System Guide for a complete discussion of network virtualization and trusted relay points. Default: Default	Field Name: sipTrunk.[n].useTrustedRelayPoint Type: String Cardinality: [0..1] Default: Default Choices: ["Off", "On", "Default"]
		Transmit UTF-8 Names in QSIG APDU				This device uses the user locale setting of the device pool to determine whether to send unicode and whether to translate received Unicode information. For the sending device, if you check this check box and the user locale setting in the device pool matches the terminating phone user locale, the device sends unicode and encodes in UTF-8 format. If the user locale settings do not match, the device sends ASCII and encodes in UTF-8 format. If the configuration parameter is not set and the user locale setting in the device pool matches the terminating phone user locale, the device sends unicode (if the name uses 8-bit format) and encodes in ISO8859-1 format. The default value for Transmit UTF-8 Names in QSIG APDU leaves the check box unchecked.	Field Name: sipTrunk.[n].enableQsigUtf8 Type: Boolean Cardinality: [0..1]
		Trunk Service Type				Choose one of the following options from the Trunk Service Type drop-down list box: None—Choose this option if the trunk will not be used for call control discovery, Extension Mobility Cross Cluster, or Cisco Intercompany Media Engine. Call Control Discovery—Choosing this option enables the trunk to support call control discovery. If you assign this trunk to the CCD advertising service in the Advertising Service window, the trunk handles inbound calls from remote call-control entities that use the SAF network. If you assign this trunk to the CCD requesting service in the Requesting Service window, the trunk handles outgoing calls to learned patterns. For more information on the call control discovery feature, see the Cisco Unified Communications Manager Features and Services Guide. Extension Mobility Cross Cluster—Choose this option to enable the trunk to support the Extension Mobility Cross Cluster (EMCC) feature. Choosing this option causes the following settings to remain blank or unchecked and become unavailable for configuration, thus retaining their default values: Media Termination Point Required, Unattended Port, Destination Address, Destination Address IPv6, and Destination Address is an SRV. For more information about the EMCC feature, see the Cisco Unified Communications Manager Features and Services Guide. Cisco Intercompany Media Engine—Ensure that the Cisco IME server is installed and available before you configure this field. Tip After you choose Call Control Discovery, Extension Mobility Cross Cluster, or Cisco Intercompany Media Engine for the trunk service type and click Next, you cannot change the trunk to a different type. Default: None(Default)	Field Name: sipTrunk.[n].sipTrunkType Type: String Cardinality: [0..1] Default: None(Default) Choices: ["None(Default)", "Call Control Discovery", "Extension Mobility Cross Cluster", "Cisco Intercompany Media Engine", "IP Multimedia Subsystem Service Control (ISC)"]
		Enable Trace				Check this check box to enable tracing within the script or uncheck this check box to disable tracing. When checked, the trace.output API provided to the Lua scripter produces SDI trace. Note Cisco recommends that you only enable tracing while debugging a script. Tracing impacts performance and should not be enabled under normal operating conditions.	Field Name: sipTrunk.[n].scriptTraceEnabled Type: Boolean Cardinality: [0..1]
		Tunneled Protocol				Select the QSIG option if you want to use SIP trunks or SIP gateways to transport (tunnel) QSIG messages from Cisco Unified Communications Manager to other PINXs. QSIG tunneling supports the following features: Call Back, Call Completion, Call Diversion, Call Transfer, Identification Services, Path Replacement, and Message Waiting Indication (MWI). Note Remote-Party-ID (RPID) headers coming in from the SIP gateway can interfere with QSIG content and cause unexpected behavior with Call Back capabilities. To prevent interference with the QSIG content, turn off the RPID headers on the SIP gateway. To turn off RPID headers on the SIP gateway, apply a SIP profile to the voIP dial peer on the gateway, as shown in the following example: voice class sip-profiles 1000request ANY sip-header Remote-Party_ID remove response ANY sip-header Remote-Party-ID remove dial-peer voice 124 voip destination-pattern 3... signaling forward unconditional session protocol sipv2 session target ipv4:<ip address> voice-class sip profiles 1000 Default: None	Field Name: sipTrunk.[n].tunneledProtocol Type: String Cardinality: [0..1] Default: None Choices: ["None", "QSIG"]
		Script Parameters					Field Name: sipTrunk.[n].scriptParameters Type: String Cardinality: [0..1]
		Use Device Pool Redirecting Party Transformation CSS				Default: True	Field Name: sipTrunk.[n].useDevicePoolRdnTransformCss Type: Boolean Cardinality: [0..1] Default: True
		Consider Traffic on This Trunk Secure				This field provides an extension to the existing security configuration on the SIP trunk, which enables a SIP trunk call leg to be considered secure if SRTP is negotiated, independent of the signaling transport. Choose one of the following values: When using both sRTP and TLS—Default When using sRTP Only—Displays when you check the SRTP Allowed check box For more information on security and trunks, see the Cisco Unified Communications Manager Security Guide. Default: When using both sRTP and TLS	Field Name: sipTrunk.[n].trunkTrafficSecure Type: String Cardinality: [0..1] Default: When using both sRTP and TLS Choices: ["When using both sRTP and TLS", "When using sRTP Only"]
		Called Party Transformation CSS				This settings allows you to send the transformed called party number in INVITE message for outgoing calls made over SIP Trunk. Make sure that the Called Party Transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device. Note If you configure the Called Party Transformation CSS as None, the transformation does not match and does not get applied. Ensure that you configure the Called Party Transformation CSS in a non-null partition that is not used for routing.	Field Name: sipTrunk.[n].cdpnTransformationCssName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Device Name *				Enter a unique identifier for the trunk. Enter a unique identifier for the trunk. The device name can include up to 50 alphanumeric characters: A-Z, a-z, numbers, hyphens (-) and underscores (_) only.	Field Name: sipTrunk.[n].name Type: String Cardinality: [1..1] MaxLength: 128
		Retry Video Call as Audio				This check box pertains to outgoing SIP trunk calls and does not impact incoming calls. By default, the system checks this check box to specify that this device should immediately retry a video call as an audio call (if it cannot connect as a video call) prior to sending the call to call control for rerouting. If you uncheck this check box, a video call that fails to connect as video does not try to establish as an audio call. The call then fails to call control, and call control routes the call via Automatic Alternate Routing (AAR) and/or route/hunt list. Default: True	Field Name: sipTrunk.[n].retryVideoCallAsAudio Type: Boolean Cardinality: [0..1] Default: True
		SIP Privacy				From the drop-down list, choose one of the following values to specify the type of SIP privacy header for SIP trunk messages to include: Default—This option represents the default value; Name/Number Presentation values that the SIP trunk receives from the Cisco Unified Communications Manager Call Control compose the SIP Privacy header. For example, if Name/Number presentation specifies Restricted, the SIP trunk sends the SIP Privacy header; however, if Name/Number presentation specifies Allowed, the SIP trunk does not send the Privacy header. None—The SIP trunk includes the Privacy:none header and implies Presentation allowed; this value overrides the Presentation information that comes from Cisco Unified Communications Manager. ID—The SIP trunk includes the Privacy:id header and implies Presentation restricted for both name and number; this value overrides the Presentation information that comes from Cisco Unified Communications Manager. ID Critical—The SIP trunk includes the Privacy:id;critical header and implies Presentation restricted for both name and number. The label critical implies that privacy services that are requested for this message are critical, and, if the network cannot provide these privacy services, this request should get rejected. This value overrides the Presentation information that comes from Cisco Unified Communications Manager. Note These headers get sent only if the Asserted Identity check box is checked. Default: Default	Field Name: sipTrunk.[n].sipPrivacy Type: String Cardinality: [0..1] Default: Default Choices: ["Default", "None", "ID", "ID Critical"]
		Geolocation Filter				From the drop-down list box, choose a geolocation filter. If you leave the <None> setting, no geolocation filter gets applied for this device. You can also choose a geolocation filter that has been configured with the System > Geolocation Filtermenu option. For an explanation of geolocation filters, including configuration details, see the Cisco Unified Communications Manager Features and Services Guide. For an overview and details of how logical partitioning uses geolocation filters, see the Cisco Unified Communications Manager Features and Services Guide.	Field Name: sipTrunk.[n].geoLocationFilterName Type: ["String", "Null"] Target: device/cucm/GeoLocationFilter Target attr: name Cardinality: [0..1] Format: uri
		Terminating Parameter Value				Valid only for IP Multimedia Subsystem Service Control (ISC) Sip trunk	Field Name: sipTrunk.[n].terminatingParameterValue Type: String Cardinality: [0..1]
		Parameter Label				Valid only for IP Multimedia Subsystem Service Control (ISC) Sip trunk	Field Name: sipTrunk.[n].parameterLabel Type: String Cardinality: [0..1]
		Request URI Domain Name				Valid only for IP Multimedia Subsystem Service Control (ISC) Sip trunk	Field Name: sipTrunk.[n].requestUriDomainName Type: String Cardinality: [0..1]
		Outbound Uri Routing Instructions				Valid only for IP Multimedia Subsystem Service Control (ISC) Sip trunk	Field Name: sipTrunk.[n].outboundUriRoutingInstructions Type: String Cardinality: [0..1]
		Media Termination Point Required				You can configure Cisco Unified Communications Manager SIP trunks to always use an MTP. Check this check box to provide media channel information in the outgoing INVITE request. When this check box is checked, all media channels must terminate and reoriginate on the MTP device. If you uncheck the check box, the Cisco Unified Communications Manager can decide whether calls are to go through the MTP device or be connected directly between the endpoints. Note If check box remains unchecked (default case), Cisco Unified Communications Manager will attempt to dynamically allocate an MTP if the DTMF methods for the call legs are not compatible. For example, existing phones that run SCCP support only out-of-band DTMF, and existing phones that run SIP support RFC2833. Because the DTMF methods are not identical, the Cisco Unified Communications Manager dynamically allocates an MTP. If, however, a new phone that runs SCCP, which supports RFC2833 and out-of-band, calls an existing phone that runs SIP, Cisco Unified Communications Manager does not allocate an MTP because both phones support RFC2833. So, by having the same type of DTMF method supported on each phone, no need exists for MTP.	Field Name: sipTrunk.[n].mtpRequired Type: Boolean Cardinality: [0..1]
		Use Orig Calling Party Pres On Divert					Field Name: sipTrunk.[n].useOrigCallingPartyPresOnDivert Type: Boolean Cardinality: [0..1]
		Cgpn Transformation Unknown Css Name					Field Name: sipTrunk.[n].cgpnTransformationUnknownCssName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		MTP Preferred Originating Codec				Indicate the preferred outgoing codec: 711ulaw 711alaw G729/G729a G729b/G729ab Note To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec. For more information, see the Cisco Unified Communications Manager System Guide. This field gets used only when the MTP Termination Point Required check box is checked. Default: 711ulaw	Field Name: sipTrunk.[n].tkSipCodec Type: String Cardinality: [0..1] Default: 711ulaw Choices: ["711ulaw", "711alaw", "G729/G729a", "G729b/G729ab"]
		Called Party Unknown Prefix					Field Name: sipTrunk.[n].calledPartyUnknownPrefix Type: String Cardinality: [0..1] MaxLength: 16
		Significant Digits				Significant digits represent the number of final digits that are retained on inbound calls. Use for the processing of incoming calls and to indicate the number of digits that are used to route calls that are coming in to the SIP device. Choose the number of significant digits to collect, from 0 to 32, or choose All. Note Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called. The default value for Significant Digits specifies All. Default: 99	Field Name: sipTrunk.[n].sigDigits Type: ["Integer", "Null"] Cardinality: [0..1] Default: 99
		Run On All Active Unified CM Nodes				To enable the trunk to run on every node, check this check box.	Field Name: sipTrunk.[n].runOnEveryNode Type: Boolean Cardinality: [0..1]
		Enable Cisco Recording Qsig Tunneling					Field Name: sipTrunk.[n].enableCiscoRecordingQsigTunneling Type: Boolean Cardinality: [0..1]
		Use Device Pool Connected Party Transformation CSS				To use the Connected Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Connected Party Transformation CSS that you configured for this device in the Trunk Configuration window. Default: True	Field Name: sipTrunk.[n].useDevicePoolCntdPnTransformationCss Type: Boolean Cardinality: [0..1] Default: True
		SUBSCRIBE Calling Search Space				Supported with the Presence feature, the SUBSCRIBE calling search space determines how Cisco Unified Communications Manager routes presence requests from the device/server/application that connects to the SIP trunk. This setting allows you to apply a calling search space separate from the call-processing search space for presence (SUBSCRIBE) requests for the SIP trunk. From the drop-down list box, choose the SUBSCRIBE calling search space to use for presence requests for the SIP trunk. All calling search spaces that you configure in Cisco Unified Communications Manager Administration display in the SUBSCRIBE Calling Search Space drop-down list box. If you do not select a different calling search space for the SIP trunk from the drop-down list, the SUBSCRIBE calling search space defaults to None. To configure a SUBSCRIBE calling search space specifically for this purpose, you configure a calling search space as you do all calling search spaces.	Field Name: sipTrunk.[n].subscribeCallingSearchSpaceName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Use Device Pool Called Party Transformation CSS				To use the Called Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Called Party Transformation CSS that you configured for this device in the Trunk Configuration window. Default: True	Field Name: sipTrunk.[n].useDevicePoolCdpnTransformCss Type: Boolean Cardinality: [0..1] Default: True
		Connected Party Id Presentation				Default: Default	Field Name: sipTrunk.[n].connectedPartyIdPresentation Type: String Cardinality: [0..1] Default: Default Choices: ["Default", "Allowed", "Restricted"]
		QSIG Variant				To display the options in the QSIG Variant drop-down list box, select QSIG from the Tunneled Protocol drop-down list box. This parameter specifies the protocol profile that is sent in outbound QSIG facility information elements. From the drop-down list box, select one of the following options: No Changes—Default. Keep this parameter set to the default value unless a Cisco support engineer instructs otherwise. Not Selected ECMA—Select for ECMA PBX systems that use Protocol Profile 0x91. ISO—Select for PBX systems that use Protocol Profile 0x9F. For more information, see the following information: Be aware that the QSIG Variant can also be defined as a clusterwide parameter. For information on QSIG support with Cisco Unified Communications Manager, see the Cisco Unified Communications Manager System Guide. Default: No Changes	Field Name: sipTrunk.[n].qsigVariant Type: String Cardinality: [0..1] Default: No Changes Choices: ["No Changes", "ECMA", "ISO"]
		Trust Received Identity				Default: Trust All (Default)	Field Name: sipTrunk.[n].trustReceivedIdentity Type: ["String", "Null"] Cardinality: [0..1] Default: Trust All (Default) Choices: ["Trust All (Default)", "Trust PAI Only", "Trust None"]
		SIP Trunk Security Profile *				Choose the security profile to apply to the SIP trunk. You must apply a security profile to all SIP trunks that are configured in Cisco Unified Communications Manager Administration. Installing Cisco Unified Communications Manager provides a predefined, nonsecure SIP trunk security profile for autoregistration. To enable security features for a SIP trunk, configure a new security profile and apply it to the SIP trunk. If the trunk does not support security, choose a nonsecure profile. To identify the settings that the profile contains, choose System > Security Profile > SIP Trunk Security Profile. For information on how to configure security profiles, see the Cisco Unified Communications Manager Security Guide. The default value for SIP Trunk Security Profile specifies Not Selected.	Field Name: sipTrunk.[n].securityProfileName Type: String Target: device/cucm/SipTrunkSecurityProfile Target attr: name Cardinality: [1..1] Format: uri
		Srtp Fallback Allowed				This tag is valid only if srtpAllowed is set to false.	Field Name: sipTrunk.[n].srtpFallbackAllowed Type: Boolean Cardinality: [0..1]
		Protocol Side *				Side information is read-only except when creating a device Default: User	Field Name: sipTrunk.[n].protocolSide Type: String Cardinality: [1..1] Default: User Choices: ["Network", "User"]
		MLPP Domain				From the drop-down list, choose an MLPP domain to associate with this device. If you leave this field blank, this device inherits its MLPP domain from the value that is set for the device pool. If the device pool does not have an MLPP Domain setting, this device inherits its MLPP Domain from the value that is set for the MLPP Domain Identifier enterprise parameter. The default value for MLPP Domain specifies None.	Field Name: sipTrunk.[n].mlppDomainId Type: ["String", "Null"] Cardinality: [0..1] MaxLength: 128 Pattern: ^[0-9a-fA-F]{6}$
		AAR Group				Choose the automated alternate routing (AAR) group for this device. The AAR group provides the prefix digits that are used to route calls that are otherwise blocked due to insufficient bandwidth. An AAR group setting of None specifies that no rerouting of blocked calls will be attempted. The default value for AAR Group specifies None.	Field Name: sipTrunk.[n].aarNeighborhoodName Type: ["String", "Null"] Target: device/cucm/AarGroup Target attr: name Cardinality: [0..1] Format: uri
		Location *				Use locations to implement call admission control (CAC) in a centralized call-processing system. CAC enables you to regulate audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls over links between locations. The location specifies the total bandwidth that is available for calls to and from this location. From the drop-down list box, choose the appropriate location for this trunk. A location setting of Hub_None means that the locations feature does not keep track of the bandwidth that this trunk consumes. A location setting of Phantom specifies a location that enables successful CAC across intercluster trunks that use H.323 protocol or SIP. To configure a new location, use the System > Location menu option. For an explanation of location-based CAC across intercluster trunks, see the Cisco Unified Communications Manager System Guide. The location also associates with the RSVP policy with regard to other locations. The configuration allows RSVP to be enabled and disabled based upon location pairs.	Field Name: sipTrunk.[n].locationName Type: String Target: device/cucm/Location Target attr: name Cardinality: [1..1] Format: uri
		Media Resource Group List				This list provides a prioritized grouping of media resource groups. An application chooses the required media resource, such as a Music On Hold server, from among the available media resources according to the priority order that a Media Resource Group List defines. The default value for Media Resource Group List specifies None.	Field Name: sipTrunk.[n].mediaResourceListName Type: ["String", "Null"] Target: device/cucm/MediaResourceList Target attr: name Cardinality: [0..1] Format: uri
		Originating Parameter Value				Valid only for IP Multimedia Subsystem Service Control (ISC) Sip trunk	Field Name: sipTrunk.[n].originatingParameterValue Type: String Cardinality: [0..1]
		Device Pool *				Choose the appropriate device pool for the trunk. For trunks, device pools specify a list of Cisco Unified Communications Managers that the trunk uses to distribute the call load dynamically. Note Calls that are initiated from a phone that is registered to a Cisco Unified Communications Manager that does not belong to the device pool of the trunk use different Cisco Unified Communications Managers of this device pool for different outgoing calls. Selection of Cisco Unified Communications Manager nodes occurs in a random order. A call that is initiated from a phone that is registered to a Cisco Unified Communications Manager that does belong to the device pool of the trunk uses the same Cisco Unified Communications Manager node for outgoing calls if the Cisco Unified Communications Manager is up and running. The default value for Device Pool specifies Not Selected.	Field Name: sipTrunk.[n].devicePoolName Type: ["String", "Null"] Target: device/cucm/DevicePool Target attr: name Cardinality: [1..1] Format: uri
		Route Class Signaling Enabled				From the drop-down list, enable or disable route class signaling for the port. Choose one of the following values: Default—If you choose this value, the device uses the setting from the Route Class Signaling service parameter. Off—Choose this value to enable route class signaling. This setting overrides the Route Class Signaling service parameter. On—Choose this value to disable route class signaling. This setting overrides the Route Class Signaling service parameter. Route class signaling communicates special routing or termination requirements to receiving devices. It must be enabled for the port to support the Hotline feature. Default: Default	Field Name: sipTrunk.[n].routeClassSignalling Type: String Cardinality: [0..1] Default: Default Choices: ["Off", "On", "Default"]
		Use Device Pool Called Css Unkn				Default: True	Field Name: sipTrunk.[n].useDevicePoolCalledCssUnkn Type: Boolean Cardinality: [0..1] Default: True
		User Hold Moh Audio Source Id				This tag is not valid for H323Phone,H323trunk and SIPTrunk	Field Name: sipTrunk.[n].userHoldMohAudioSourceId Type: ["String", "Null", "Integer"] Target: device/cucm/MohAudioSource Target attr: sourceId Cardinality: [0..1] Format: uri Choices: ["0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "10", "11", "12", "13", "14", "15", "16", "17", "18", "19", "20", "21", "22", "23", "24", "25", "26", "27", "28", "29", "30", "31", "32", "33", "34", "35", "36", "37", "38", "39", "40", "41", "42", "43", "44", "45", "46", "47", "48", "49", "50", "null", ""]
		DTMF Signaling Method				Choose from the following options: No Preference (default)— Cisco Unified Communications Manager will pick the DTMF method to negotiate DTMF, so the call does not require an MTP. If Cisco Unified Communications Manager has no choice but to allocate an MTP (if the Media Termination Point Required check box is checked), SIP trunk will negotiate DTMF to RFC2833. RFC 2833—Choose this configuration if the preferred DTMF method to be used across the trunk is RFC2833. Cisco Unified Communications Manager makes every effort to negotiate RFC2833, regardless of MTP usage. Out of band provides the fallback method if the peer endpoint supports it. OOB and RFC 2833—Choose this configuration if both out of band and RFC2833 should be used for DTMF. Note If the peer endpoint supports both out of band and RFC2833, Cisco Unified Communications Manager will negotiate both out-of-band and RFC2833 DTMF methods. As a result, two DTMF events would get sent for the same DTMF keypress (one out of band and the other, RFC2833). Default: No Preference	Field Name: sipTrunk.[n].dtmfSignalingMethod Type: String Cardinality: [0..1] Default: No Preference Choices: ["No Preference", "Out of Band", "RFC 2833", "OOB and RFC 2833"]
		SIP Profile *				From the drop-down list box, choose the SIP profile that is to be used for this SIP trunk. The default value for SIP Profile specifies None Selected.	Field Name: sipTrunk.[n].sipProfileName Type: String Target: device/cucm/SipProfile Target attr: name Cardinality: [1..1] Format: uri
		Destinations					Field Name: destinations Type: ["Object", "Null"] Cardinality: [0..1]
				Destination			Field Name: destination.[n] Type: Array Cardinality: [0..16]
					Address Ipv6		Field Name: sipTrunk.[n].destinations.destination.[n].addressIpv6 Type: String Cardinality: [0..1] MaxLength: 255
					Address Ipv4		Field Name: sipTrunk.[n].destinations.destination.[n].addressIpv4 Type: String Cardinality: [0..1] MaxLength: 255
					Port	Default: 5060	Field Name: sipTrunk.[n].destinations.destination.[n].port Type: Integer Cardinality: [0..1] Default: 5060
					Sort Order *		Field Name: sipTrunk.[n].destinations.destination.[n].sortOrder Type: Integer Cardinality: [1..1]
		MLPP Preemption				This setting only affects devices that support MLPP. Default: Default	Field Name: sipTrunk.[n].preemption Type: String Cardinality: [0..1] Default: Default Choices: ["Disabled", "Forceful", "Default"]
		Service				Valid only for IP Multimedia Subsystem Service Control (ISC) Sip trunk	Field Name: sipTrunk.[n].service Type: String Cardinality: [0..1]
		PSTN Access				If you use the Cisco Intercompany Media Engine feature, check this check box to indicate that calls made through this trunk might reach the PSTN. Check this check box even if all calls through this trunk device do not reach the PSTN. For example, check this check box for tandem trunks or an H.323 gatekeeper routed trunk if calls might go to the PSTN. When checked, this check box causes the system to create upload voice call records (VCRs) to validate calls made through this trunk device. By default, this check box remains checked. For more information on Cisco Intercompany Media Engine, see the Cisco Intercompany Media Engine Installation and Configuration Guide. Default: True	Field Name: sipTrunk.[n].pstnAccess Type: Boolean Cardinality: [0..1] Default: True
		Calling Party Selection				Choose the directory number that is sent on an outbound call. The following options specify which directory number is sent: Originator—Send the directory number of the calling device. First Redirect Number—Send the directory number of the redirecting device. Last Redirect Number—Send the directory number of the last device to redirect the call. First Redirect Number (External)—Send the external directory number of the redirecting device. Last Redirect Number (External)—Send the external directory number of the last device to redirect the call. The default value for Calling Party Selection specifies Originator. Default: Originator	Field Name: sipTrunk.[n].callingPartySelection Type: String Cardinality: [0..1] Default: Originator Choices: ["Originator", "First Redirect Number", "Last Redirect Number", "First Redirect Number (External)", "Last Redirect Number (External)"]
		Common Device Configuration				Choose the common device configuration to which you want this trunk assigned. The common device configuration includes the attributes (services or features) that are associated with a particular user. Common device configurations are configured in the Common Device Configuration window.	Field Name: sipTrunk.[n].commonDeviceConfigName Type: ["String", "Null"] Target: device/cucm/CommonDeviceConfig Target attr: name Cardinality: [0..1] Format: uri
		Unknown Prefix					Field Name: sipTrunk.[n].unknownPrefix Type: ["String", "Null"] Cardinality: [0..1] MaxLength: 16 Pattern: ^([0-9*#+]{0,16})\|([Dd]efault)$
		Redirecting Diversion Header Delivery - Outbound				Check this check box to include the Redirecting Number in the outgoing INVITE message from the Cisco Unified Communications Manager to indicate the original called party number and the redirecting reason of the call when the call is forwarded. Uncheck the check box to exclude the first Redirecting Number and the redirecting reason from the outgoing INVITE message. You use Redirecting Number for voice-messaging integration only. If your configured voice-messaging system supports Redirecting Number, you should check the check box. The default value for Redirecting Number IE Delivery - Outbound specifies check box does not get checked.	Field Name: sipTrunk.[n].acceptOutboundRdnis Type: Boolean Cardinality: [0..1]
		Use Device Pool Cgpn Transform Css				Default: True	Field Name: sipTrunk.[n].useDevicePoolCgpnTransformCss Type: Boolean Cardinality: [0..1] Default: True
		Path Replacement Support					Field Name: sipTrunk.[n].pathReplacementSupport Type: Boolean Cardinality: [0..1]
		Destination Address is an SRV				This field specifies that the configured Destination Address is an SRV record. The default value specifies unchecked.	Field Name: sipTrunk.[n].destAddrIsSrv Type: Boolean Cardinality: [0..1]
		Trace Flag					Field Name: sipTrunk.[n].traceFlag Type: Boolean Cardinality: [0..1]
		Geolocation				From the drop-down list box, choose a geolocation. You can choose the Unspecified geolocation, which designates that this device does not associate with a geolocation. You can also choose a geolocation that has been configured with the System > Geolocation Configuration menu option. For an explanation of geolocations, including configuration details, see the Cisco Unified Communications Manager Features and Services Guide. For an overview and details of how logical partitioning uses geolocations, see the Cisco Unified Communications Manager Features and Services Guide.	Field Name: sipTrunk.[n].geoLocationName Type: ["String", "Null"] Target: device/cucm/GeoLocation Target attr: name Cardinality: [0..1] Format: uri
		Connected Party Transformation CSS				This setting is applicable only for inbound calls. This setting allows you to transform the connected party number on the device to display the connected number in another format, such as a DID or E164 number. Cisco Unified Communications Manager includes the transformed number in the headers of various SIP messages, including 200 OK and mid-call update/reinvite messages. Make sure that the Connected Party Transformation CSS that you choose contains the connected party transformation pattern that you want to assign to this device. Note If you configure the Connected Party Transformation CSS as None, the transformation does not match and does not get applied. Ensure that you configure the Calling Party Transformation pattern used for Connected Party Transformation in a non-null partition that is not used for routing.	Field Name: sipTrunk.[n].cntdPnTransformationCssName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Calling Name Presentation				Cisco Unified Communications Manager uses calling name ID presentation (CNIP) as a supplementary service to provide the calling party name. The SIP trunk level configuration takes precedence over the call-by-call configuration. Choose Allowed, which is the default, if you want Cisco Unified Communications Manager to send calling name information. Choose Restricted if you do not want Cisco Unified Communications Manager to send the calling name information. The default value for Calling Name Presentation specifies Default. Note Be aware that this service is not available when QSIG tunneling is enabled. Default: Default	Field Name: sipTrunk.[n].callingname Type: String Cardinality: [0..1] Default: Default Choices: ["Default", "Allowed", "Restricted"]
		Network Hold Moh Audio Source Id				This tag is not valid for H323Phone,H323trunk and SIPTrunk	Field Name: sipTrunk.[n].networkHoldMohAudioSourceId Type: ["String", "Null", "Integer"] Target: device/cucm/MohAudioSource Target attr: sourceId Cardinality: [0..1] Format: uri Choices: ["0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "10", "11", "12", "13", "14", "15", "16", "17", "18", "19", "20", "21", "22", "23", "24", "25", "26", "27", "28", "29", "30", "31", "32", "33", "34", "35", "36", "37", "38", "39", "40", "41", "42", "43", "44", "45", "46", "47", "48", "49", "50", "null", ""]
		Call Classification				This parameter determines whether an incoming call through this trunk is considered off the network (OffNet) or on the network (OnNet). The default value for Call Classification is Use System Default. When the Call Classification field is configured as Use System Default, the setting of the Cisco Unified Communications Manager clusterwide service parameter, Call Classification, determines whether the trunk is OnNet or OffNet. This field provides an OnNet or OffNet alerting tone when the call is OnNet or OffNet, respectively. Use this parameter in conjunction with the settings on the Route Pattern Configuration window to classify an outgoing call as OnNet or OffNet. Default: Use System Default	Field Name: sipTrunk.[n].networkLocation Type: String Cardinality: [0..1] Default: Use System Default Choices: ["OnNet", "OffNet", "Use System Default"]
		Unattended Port				Check this check box if calls can be redirected and transferred to an unattended port, such as a voice mail port. The default value for this check box leaves it unchecked.	Field Name: sipTrunk.[n].unattendedPort Type: Boolean Cardinality: [0..1]
		Calling Party Transformation CSS				This settings allows you to send the transformed calling party number in INVITE message for outgoing calls made over SIP Trunk. Also when redirection occurs for outbound calls, this CSS will be used to transform the connected number that is sent from Cisco Unified Communications Manager side in outgoing reINVITE / UPDATE messages. Make sure that the Calling Party Transformation CSS that you choose contains the calling party transformation pattern that you want to assign to this device. Tip If you configure the Calling Party Transformation CSS as None, the transformation does not match and does not get applied. Ensure that you configure the Calling Party Transformation Pattern in a non-null partition that is not used for routing.	Field Name: sipTrunk.[n].cgpnTransformationCssName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Connected Name Presentation				Default: Default	Field Name: sipTrunk.[n].connectedNamePresentation Type: String Cardinality: [0..1] Default: Default Choices: ["Default", "Allowed", "Restricted"]
		External Presentation Info					Field Name: externalPresentationInfo Type: Object Cardinality: [0..1]
			Presentation Info				Field Name: presentationInfo Type: ["Object", "Null"] Cardinality: [0..1]
				External Presentation Number			Field Name: sipTrunk.[n].externalPresentationInfo.presentationInfo.externalPresentationNumber Type: ["String", "Null"] Cardinality: [0..1]
				External Presentation Name			Field Name: sipTrunk.[n].externalPresentationInfo.presentationInfo.externalPresentationName Type: ["String", "Null"] Cardinality: [0..1] MaxLength: 50
			Is Anonymous				Field Name: sipTrunk.[n].externalPresentationInfo.isAnonymous Type: Boolean Cardinality: [0..1]
		Use Device Pool Calling Party Transformation CSS				To use the Calling Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Calling Party Transformation CSS that you configured in the Trunk Configuration window. Default: True	Field Name: sipTrunk.[n].useDevicePoolCgpnTransformCssUnkn Type: Boolean Cardinality: [0..1] Default: True
		Called Party Unknown Transformation Css Name					Field Name: sipTrunk.[n].calledPartyUnknownTransformationCssName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Use Ime Public Ip Port					Field Name: sipTrunk.[n].useImePublicIpPort Type: Boolean Cardinality: [0..1]
		Asserted-Type				From the drop-down list, choose one of the following values to specify the type of Asserted Identity header that SIP trunk messages should include: Default—This option represents the default value; Screening indication information that the SIP trunk receives from Cisco Unified Communications Manager Call Control determines the type of header that the SIP trunk sends. PAI—The Privacy-Asserted Identity (PAI) header gets sent in outgoing SIP trunk messages; this value overrides the Screening indication value that comes from Cisco Unified Communications Manager. PPI—The Privacy Preferred Identity (PPI) header gets sent in outgoing SIP trunk messages; this value overrides the Screening indication value that comes from Cisco Unified Communications Manager. Note These headers get sent only if the Asserted Identity check box is checked. Default: Default	Field Name: sipTrunk.[n].sipAssertedType Type: String Cardinality: [0..1] Default: Default Choices: ["Default", "PAI", "PPI"]
		Unknown Strip Digits					Field Name: sipTrunk.[n].unknownStripDigits Type: Integer Cardinality: [0..1]
		Called Party Unknown Strip Digits					Field Name: sipTrunk.[n].calledPartyUnknownStripDigits Type: ["Integer", "Null"] Cardinality: [0..1]
		Normalization Script				From the drop-down list box, choose the script that you want to apply to this trunk. To import another script, go to the SIP Normalization Script Configuration window (Device > Device Settings > SIP Normalization Script), and import a new script file.	Field Name: sipTrunk.[n].sipNormalizationScriptName Type: ["String", "Null"] Target: device/cucm/SIPNormalizationScript Target attr: name Cardinality: [0..1] Format: uri
		Packet Capture Duration				This setting exists for troubleshooting encryption only; packet capturing may cause high CPU usage or call-processing interruptions. This field specifies the maximum number of minutes that is allotted for one session of packet capturing. The default setting equals 0, although the range exists from 0 to 300 minutes. To initiate packet capturing, enter a value other than 0 in the field. After packet capturing completes, the value, 0, displays. For more information on capturing packets, see the Cisco Unified Communications Manager Troubleshooting Guide.	Field Name: sipTrunk.[n].packetCaptureDuration Type: ["Integer", "Null"] Cardinality: [0..1]
		Remote-Party-Id				Default: True	Field Name: sipTrunk.[n].isRpidEnabled Type: Boolean Cardinality: [0..1] Default: True
		MLPP Indication				This setting only affects devices that support MLPP. Default: Off	Field Name: sipTrunk.[n].mlppIndicationStatus Type: String Cardinality: [0..1] Default: Off Choices: ["Off", "On", "Default"]
		Packet Capture Mode				This setting exists for troubleshooting encryption only; packet capturing may cause high CPU usage or call-processing interruptions. Choose one of the following options from the drop-down list box: None—This option, which serves as the default setting, indicates that no packet capturing is occurring. After you complete packet capturing, configure this setting. Batch Processing Mode— Cisco Unified Communications Manager writes the decrypted or nonencrypted messages to a file, and the system encrypts each file. On a daily basis, the system creates a new file with a new encryption key. Cisco Unified Communications Manager, which stores the file for seven days, also stores the keys that encrypt the file in a secure location. Cisco Unified Communications Manager stores the file in the PktCap virtual directory. A single file contains the time stamp, source IP address, source IP port, destination IP address, packet protocol, message length, and the message. The TAC debugging tool uses HTTPS, administrator username and password, and the specified day to request a single encrypted file that contains the captured packets. Likewise, the tool requests the key information to decrypt the encrypted file. Before you contact TAC, you must capture the SRTP packets by using a sniffer trace between the affected devices. For more information on capturing packets, see the Troubleshooting Guide for Cisco Unified Communications Manager. Default: None	Field Name: sipTrunk.[n].packetCaptureMode Type: String Cardinality: [0..1] Default: None Choices: ["None", "Batch Processing Mode"]
		Asserted-Identity				Default: True	Field Name: sipTrunk.[n].isPaiEnabled Type: Boolean Cardinality: [0..1] Default: True
		Load Information				For devices with load information, if any special load information is specified the special attribute is set to TRUE.Otherwise,the load information is default for the product.	Field Name: sipTrunk.[n].loadInformation Type: ["String", "Null"] Cardinality: [0..1]
		Trunk Type *				Product ID string. read-only except when creating a device.	Field Name: sipTrunk.[n].product Type: String Cardinality: [1..1] Choices: ["7914 14-Button Line Expansion Module", "7915 12-Button Line Expansion Module", "7915 24-Button Line Expansion Module", "7916 12-Button Line Expansion Module", "7916 24-Button Line Expansion Module", "AIM-VOICE-30", "Analog Phone", "Annunciator", "BEKEM 36-Button Line Expansion Module", "C881V", "C887VA-V", "CKEM 36-Button Line Expansion Module", "CP-8800-Audio 28-Button Key Expansion Module", "CP-8800-Video 28-Button Key Expansion Module", "CTI Port", "CTI Remote Device", "CTI Route Point", "Carrier-integrated Mobile", "Cisco IAD2400", "Cisco 12 S", "Cisco 12 SP", "Cisco 12 SP+", "Cisco 1751", "Cisco 1760", "Cisco 1861", "Cisco 269X", "Cisco 26XX", "Cisco 2801", "Cisco 2811", "Cisco 2821", "Cisco 2851", "Cisco 2901", "Cisco 2911", "Cisco 2921", "Cisco 2951", "Cisco 30 SP+", "Cisco 30 VIP", "Cisco 362X", "Cisco 364X", "Cisco 366X", "Cisco 3725", "Cisco 3745", "Cisco 3825", "Cisco 3845", "Cisco 3905", "Cisco 3911", "Cisco 3925", "Cisco 3925E", "Cisco 3945", "Cisco 3945E", "Cisco 3951", "Cisco 6901", "Cisco 6911", "Cisco 6921", "Cisco 6941", "Cisco 6945", "Cisco 6961", "Cisco 7811", "Cisco 7821", "Cisco 7832", "Cisco 7841", "Cisco 7861", "Cisco 7902", "Cisco 7905", "Cisco 7906", "Cisco 7910", "Cisco 7911", "Cisco 7912", "Cisco 7920", "Cisco 7921", "Cisco 7925", "Cisco 7926", "Cisco 7931", "Cisco 7935", "Cisco 7936", "Cisco 7937", "Cisco 7940", "Cisco 7941", "Cisco 7941G-GE", "Cisco 7942", "Cisco 7945", "Cisco 7960", "Cisco 7961", "Cisco 7961G-GE", "Cisco 7962", "Cisco 7965", "Cisco 7970", "Cisco 7971", "Cisco 7975", "Cisco 7985", "Cisco 860", "Cisco 881", "Cisco 8811", "Cisco 8821", "Cisco 8831", "Cisco 8832", "Cisco 8832NR", "Cisco 8841", "Cisco 8845", "Cisco 8851", "Cisco 8851NR", "Cisco 8861", "Cisco 8865", "Cisco 8865NR", "Cisco 888/887/886", "Cisco 8941", "Cisco 8945", "Cisco 8961", "Cisco 9951", "Cisco 9971", "Cisco ATA 186", "Cisco ATA 187", "Cisco ATA 190", "Cisco ATA 191", "Cisco C8200/L-1N-4T", "Cisco C8300-1N1S-4T2X", "Cisco C8300-1N1S-6T", "Cisco C8300-2N2S-4T2X/6T", "Cisco Catalyst 4000 Access Gateway Module", "Cisco Catalyst 4224 Voice Gateway Switch", "Cisco Catalyst 6000 12 port FXO Gateway", "Cisco Catalyst 6000 24 port FXS Gateway", "Cisco Catalyst 6000 E1 VoIP Gateway", "Cisco Catalyst 6000 T1 VoIP Gateway", "Cisco Cius", "Cisco Cius SP", "Cisco Collaboration Mobile Convergence", "Cisco Conference Bridge (WS-SVC-CMM)", "Cisco Conference Bridge Hardware", "Cisco Conference Bridge Software", "Cisco DX650", "Cisco DX70", "Cisco DX80", "Cisco Dual Mode for Android", "Cisco Dual Mode for iPhone", "Cisco E20", "Cisco ENCS 5400 ISRV", "Cisco IOS Conference Bridge", "Cisco IOS Enhanced Conference Bridge", "Cisco IOS Enhanced Media Termination Point", "Cisco IOS Enhanced Software Media Termination Point", "Cisco IOS Guaranteed Audio Video Conference Bridge", "Cisco IOS Heterogeneous Video Conference Bridge", "Cisco IOS Homogeneous Video Conference Bridge", "Cisco IOS Media Termination Point", "Cisco IP Communicator", "Cisco ISR 4321", "Cisco ISR 4331", "Cisco ISR 4351", "Cisco ISR 4431", "Cisco ISR 4451", "Cisco ISR 4461", "Cisco Jabber for Tablet", "Cisco MGCP BRI Port", "Cisco MGCP E1 Port", "Cisco MGCP FXO Port", "Cisco MGCP FXS Port", "Cisco MGCP T1 Port", "Cisco Media Server (WS-SVC-CMM-MS)", "Cisco Media Termination Point (WS-SVC-CMM)", "Cisco Media Termination Point Hardware", "Cisco Media Termination Point Software", "Cisco Meeting Server", "Cisco SIP FXS Port", "Cisco Spark Remote Device", "Cisco TelePresence", "Cisco TelePresence 1000", "Cisco TelePresence 1100", "Cisco TelePresence 1300-47", "Cisco TelePresence 1300-65", "Cisco TelePresence 200", "Cisco TelePresence 3000", "Cisco TelePresence 3200", "Cisco TelePresence 400", "Cisco TelePresence 500-32", "Cisco TelePresence 500-37", "Cisco TelePresence Codec C40", "Cisco TelePresence Codec C60", "Cisco TelePresence Codec C90", "Cisco TelePresence Conductor", "Cisco TelePresence DX70", "Cisco TelePresence EX60", "Cisco TelePresence EX90", "Cisco TelePresence Exchange System", "Cisco TelePresence IX5000", "Cisco TelePresence MCU", "Cisco TelePresence MX200", "Cisco TelePresence MX200 G2", "Cisco TelePresence MX300", "Cisco TelePresence MX300 G2", "Cisco TelePresence MX700", "Cisco TelePresence MX800", "Cisco TelePresence MX800 Dual", "Cisco TelePresence Profile 42 (C20)", "Cisco TelePresence Profile 42 (C40)", "Cisco TelePresence Profile 42 (C60)", "Cisco TelePresence Profile 52 (C40)", "Cisco TelePresence Profile 52 (C60)", "Cisco TelePresence Profile 52 Dual (C60)", "Cisco TelePresence Profile 65 (C60)", "Cisco TelePresence Profile 65 Dual (C90)", "Cisco TelePresence Quick Set C20", "Cisco TelePresence SX10", "Cisco TelePresence SX20", "Cisco TelePresence SX80", "Cisco TelePresence TX1310-65", "Cisco TelePresence TX9000", "Cisco TelePresence TX9200", "Cisco Unified Client Services Framework", "Cisco Unified Communications for RTX", "Cisco Unified Mobile Communicator", "Cisco Unified Personal Communicator", "Cisco VG200", "Cisco VG248 Gateway", "Cisco VGC Phone", "Cisco VGC Virtual Phone", "Cisco VGD-1T3", "Cisco VXC 6215", "Cisco Video Conference Bridge(IPVC-35xx)", "Cisco Voice Mail Port", "Cisco Webex Board 55", "Cisco Webex Board 70", "Cisco Webex Board 85", "Cisco Webex DX80", "Cisco Webex Desk LE", "Cisco Webex Desk Pro", "Cisco Webex Room 55", "Cisco Webex Room 55 Dual", "Cisco Webex Room 70 Dual", "Cisco Webex Room 70 Dual G2", "Cisco Webex Room 70 Panorama", "Cisco Webex Room 70 Single", "Cisco Webex Room 70 Single G2", "Cisco Webex Room Kit", "Cisco Webex Room Kit Mini", "Cisco Webex Room Kit Plus", "Cisco Webex Room Kit Pro", "Cisco Webex Room Panorama", "Cisco Webex Room Phone", "Cisco Webex VDI Svc Framework", "Communication Media Module", "EMCC Base Phone", "FLEX_SLOT", "Gatekeeper", "Generic Desktop Video Endpoint", "Generic Multiple Screen Room System", "Generic Single Screen Room System", "H.225 Trunk (Gatekeeper Controlled)", "H.323 Client", "H.323 Gateway", "Hunt List", "IAD2400_ANALOG", "IAD2400_DIGITAL", "IMS-integrated Mobile (Basic)", "IP-STE", "ISDN BRI Phone", "Inter-Cluster Trunk (Gatekeeper Controlled)", "Inter-Cluster Trunk (Non-Gatekeeper Controlled)", "Interactive Voice Response", "Load Simulator", "Motorola CN622", "Music On Hold", "NM-1V", "NM-2V", "NM-4VWIC-MBRD", "NM-HD-1V", "NM-HD-2V", "NM-HD-2VE", "NM-HDA", "NM-HDV", "NM-HDV2-0PORT", "NM-HDV2-1PORT", "NM-HDV2-2PORT", "Nokia S60", "PA-MCX", "PA-VXA", "PA-VXB", "PA-VXC", "Pilot", "Remote Destination Profile", "Route List", "SCCP Device", "SCCP gateway virtual phone", "SIP Trunk", "SIP WSM Connection", "SPA8800", "Third-party AS-SIP Endpoint", "Third-party SIP Device (Advanced)", "Third-party SIP Device (Basic)", "Transnova S3", "Universal Device Template", "Unknown", "VG202", "VG204", "VG224", "VG310", "VG320", "VG350", "VG400", "VG420", "VG450", "VGC Port", "VIC_SLOT", "VKEM 36-Button Line Expansion Module", "VNM-HDA", "VWIC_SLOT", "WS-SVC-CMM-MS", "WS-X6600"]
		Description				Enter a descriptive name for the trunk. The description can include up to 114 characters in any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), back-slash (\), or angle brackets (<>).	Field Name: sipTrunk.[n].description Type: ["String", "Null"] Cardinality: [0..1] MaxLength: 128
		Send Geolocation Information				Check this check box to send geolocation information for this device. For an overview and details of how logical partitioning uses geolocation information, see the the Cisco Unified Communications Manager Features and Services Guide.	Field Name: sipTrunk.[n].sendGeoLocation Type: Boolean Cardinality: [0..1]
		ASN.1 ROSE OID Encoding				To display the options in the ASN.1 ROSE OID Encoding drop-down list box, choose QSIG from the Tunneled Protocol drop-down list box. This parameter specifies how to encode the Invoke Object ID (OID) for remote operations service element (ROSE) operations. From the drop-down list box, select one of the following options: No Changes—Default. Keep this parameter set to the default value unless a Cisco support engineer instructs otherwise. Not Selected Use Global Value ECMA—If you selected the ECMA option from the QSIG Variant drop-down list box, select this option. Use Global Value ISO—If you selected the ISO option from the QSIG Variant drop-down list box, select this option. Use Local Value For more information, see the following information: Be aware that ASN.1 ROSE OID Encoding can also be defined as a clusterwide parameter. For information on QSIG support with Cisco Unified Communications Manager, see the Cisco Unified Communications Manager System Guide. Default: No Changes	Field Name: sipTrunk.[n].asn1RoseOidEncoding Type: String Cardinality: [0..1] Default: No Changes Choices: ["No Changes", "Use Local Value", "Use Global Value ISO", "Use Global Value ECMA"]
		Redirecting Party Transformation CSS					Field Name: sipTrunk.[n].rdnTransformationCssName Type: ["String", "Null"] Cardinality: [0..1]
		Class *				Class ID string. Class information is read-only except when creating a device.	Field Name: sipTrunk.[n].class Type: String Cardinality: [1..1] Choices: ["Phone", "Gateway", "Conference Bridge", "Media Termination Point", "Route List", "Voice Mail", "CTI Route Point", "Music On Hold", "Simulation", "Pilot", "GateKeeper", "Add-on modules", "Hidden Phone", "Trunk", "Tone Announcement Player", "Remote Destination Profile", "EMCC Base Phone Template", "EMCC Base Phone", "Remote Destination Profile Template", "Gateway Template", "UDP Template", "Phone Template", "Device Profile", "Invalid", "Interactive Voice Response"]
		SRTP Allowed - When this flag is checked, IPSec needs to be configured in the network to provide end to end security. Failure to do so will expose keys and other information.				Check this check box if you want Cisco Unified Communications Manager to allow secure and nonsecure media calls over the trunk. Checking this check box enables Secure Real-Time Protocol (SRTP) SIP Trunk connections and also allows the SIP trunk to fall back to Real-Time Protocol (RTP) if the endpoints do not support SRTP. If you do not check this check box, Cisco Unified Communications Manager prevents SRTP negotiation with the trunk and uses RTP negotiation instead. The default value for this check box leaves it unchecked. Caution If you check this check box, Cisco strongly recommends that you use an encrypted TLS profile, so that keys and other security-related information do not get exposed during call negotiations. If you use a non-secure profile, SRTP will still work but the keys will get exposed in signaling and traces. In that case, you must ensure the security of the network between Cisco Unified Communications Manager and the destination side of the trunk. For more information on encryption for trunks, see the Cisco Unified Communications Manager Security Guide.	Field Name: sipTrunk.[n].srtpAllowed Type: Boolean Cardinality: [0..1]
		BLF Presence Group *					Field Name: sipTrunk.[n].presenceGroupName Type: String Target: device/cucm/PresenceGroup Target attr: name Cardinality: [1..1] Format: uri
		Recording Information				This field can have values 0,1 or 2 Default: 0	Field Name: sipTrunk.[n].recordingInformation Type: String Cardinality: [0..1] Default: 0 Pattern: ^[0-2]$
		Rerouting Calling Search Space				Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The rerouting calling search space gets used to determine where a SIP user (A) can refer another user (B) to a third party (C). After the refer is completed, B and C connect. In this case, the rerouting calling search space that is used is that of the initial SIP user (A). Note Calling Search Space also applies to 3xx redirection and INVITE with Replaces features. The default value for Rerouting Calling Search Space specifies None.	Field Name: sipTrunk.[n].rerouteCallingSearchSpaceName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Transmit UTF-8 for Calling Party Name				This device uses the user locale setting of the device pool to determine whether to send unicode and whether to translate received Unicode information. For the sending device, if you check this check box and the user locale setting in the device pool matches the terminating phone user locale, the device sends unicode. If the user locale settings do not match, the device sends ASCII. The receiving device translates incoming unicode characters based on the user locale setting of the sending device pool. If the user locale setting matches the terminating phone user locale, the phone displays the characters. Note The phone may display malformed characters if the two ends of the trunk configure user locales that do not belong to the same language group. The default value for Transmit UTF-8 for Calling Party Name leaves the check box unchecked.	Field Name: sipTrunk.[n].transmitUtf8 Type: Boolean Cardinality: [0..1]
		Out-Of-Dialog Refer Calling Search Space					Field Name: sipTrunk.[n].referCallingSearchSpaceName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		E.164 Transformation Profile				Check this check box if you want to use the Cisco Intercompany Media Engine and calls might reach the PSTN. For more information, see the Cisco Intercompany Media Engine Installation and Configuration Guide. From the drop-down list box, choose the appropriate E.164 transformation that you created on the Intercompany Media Services E.164 Transformation Configuration window (Advanced Features > Intercompany Media Services > E.164 Transformation). For more information on Cisco Intercompany Media Engine, see the Cisco Intercompany Media Engine Installation and Configuration Guide.	Field Name: sipTrunk.[n].imeE164TransformationName Type: ["String", "Null"] Target: device/cucm/ImeE164Transformation Target attr: name Cardinality: [0..1] Format: uri
		AAR Calling Search Space				Choose the appropriate calling search space for the device to use when performing automated alternate routing (AAR). The AAR calling search space specifies the collection of route partitions that are searched to determine how to route a collected (originating) number that is otherwise blocked due to insufficient bandwidth. The default value for AAR Calling Search Space specifies None.	Field Name: sipTrunk.[n].automatedAlternateRoutingCssName Type: ["String", "Null"] Target: device/cucm/Css Target attr: name Cardinality: [0..1] Format: uri
		Calling and Connected Party Info Format *				This option allows you to configure whether Cisco Unified Communications Manager inserts a directory number, a directory URI, or a blended address that includes both the directory number and directory URI in the SIP identity headers for outgoing SIP messages. From the drop-down list box, choose one of the following options: Deliver DN only in connected party—In outgoing SIP messages, Cisco Unified Communications Manager inserts the calling party’s directory number in the SIP contact header information. This is the default setting. Deliver URI only in connected party, if available—In outgoing SIP messages, Cisco Unified Communications Manager inserts the sending party’s directory URI in the SIP contact header. If a directory URI is not available, Cisco Unified Communications Manager inserts the directory number instead. Deliver URI and DN in connected party, if available—In outgoing SIP messages, Cisco Unified Communications Manager inserts a blended address that includes the calling party's directory URI and directory number in the SIP contact headers. If a directory URI is not available, Cisco Unified Communications Manager includes the directory number only. Note You should set this field to Deliver URI only in connected party or Deliver URI and DN in connected party only if you are setting up URI dialing between Cisco Unified CM systems of release 9.0 or greater, or between a Cisco Unified CM system of release 9. 0 or greater and a third party solution that supports URI dialing. Otherwise, you must set this field to Deliver DN only in connected party. For more information on URI dialing, see the URI dialing chapter in the Cisco Unified Communications Manager System Guide. Default: Deliver DN only in connected party	Field Name: sipTrunk.[n].callingAndCalledPartyInfoFormat Type: String Cardinality: [1..1] Default: Deliver DN only in connected party Choices: ["Deliver DN only in connected party", "Deliver URI only in connected party, if available", "Deliver URI and DN in connected party, if available"]

Model: relation/HcsCCServerREL

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