About Trunk Setup
Use a trunk device to configure a logical route to a gatekeeper (that is, the wholesale network or an intercluster trunk with gatekeeper control), to an intercluster trunk without a gatekeeper, or to a SIP network. Choose from the following available trunk types:
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H.225 trunk (gatekeeper controlled)
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Intercluster trunk (gatekeeper controlled)
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Intercluster trunk (non-gatekeeper controlled)
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SIP trunk
Tip |
Configure SIP Trunk Security Profiles and SIP Profiles before you configure a SIP Trunk. For more information, see the Cisco Unified Communications Manager Security Guide. |
Tip |
Resetting a trunk drops any calls in progress that are using that trunk. Restarting a gateway tries to preserve the calls in progress that are using that gateway, if possible. Other devices wait until calls complete before restarting or resetting. Resetting/restarting an H.323 or SIP device does not physically reset/restart the hardware; it only reinitializes the configuration that is loaded by Cisco Unified Communications Manager. For SIP trunks, Restart and Reset behave the same way, so all active calls will disconnect when either choice is pressed. Trunks do not have to undergo a Restart or Reset when Packet Capture is enabled or disabled. |
H.225 and Intercluster Trunks Settings
The following table describes the trunk settings for gatekeeper-controlled H.225 trunks, gatekeeper-controlled intercluster trunks, and non-gatekeeper-controlled intercluster trunks.
Field |
Description |
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Device Information |
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Device Name |
Enter a unique identifier for the trunk. |
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Description |
Enter a descriptive name for the trunk. The description can include up to 50 characters in any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), back-slash (\), or angle brackets (<>). |
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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.
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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. |
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Call Classification |
This parameter determines whether an incoming call through this trunk is considered off the network (OffNet) or on the network (OnNet). 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. The alerting tones are provided by Cisco Unified Communications Manager Annunciators. Use this parameter in conjunction with the settings on the Route Pattern Configuration window to classify an outgoing call as OnNet or OffNet. |
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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. |
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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 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. |
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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. |
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Tunneled Protocol |
This drop-down list box displays for H.225 trunks, gatekeeper-controlled trunks, and non-gatekeeper-controlled trunks. Choose the QSIG option if you want to use trunks to transport (tunnel) non-H.323 protocol information in H.323 signaling messages from Cisco Unified Communications Manager to other Annex M.1-compliant H.323 PINXs. QSIG tunneling supports the following features: Call Completion, Call Diversion, Call Transfer, Identification Services, and Message Waiting Indication. |
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QSIG Variant |
To display the options in the QSIG Variant drop-down list box, choose 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, choose one of the following options:
For more information, see the following information:
|
||||
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, choose one of the following options:
For more information, see the following information:
|
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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:
For more information on capturing packets, see the Troubleshooting Guide for Cisco Unified Communications Manager. |
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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. |
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Media Termination Point Required |
This check box is used to indicate whether a media termination point (MTP) is used to implement features that H.323 does not support (such as hold and transfer). Check the Media Termination Point Required check box if you want to use a media termination point to implement features. Uncheck the Media Termination Point Required check box if you do not want to use a media termination point to implement features. Use this check box only for H.323 clients and those H.323 devices that do not support the H.245 Empty Capabilities Set or if you want media streaming to terminate through a single source. If you check this check box to require an MTP and one or both parties are a video endpoint, the call operates as audio only. |
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Retry Video Call as Audio |
This check box applies only to video endpoints that receive a call. For trunks, this check box pertains to calls that are received from Cisco Unified Communications Manager but not to calls that are received from the wide-area network (WAN). 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. |
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Wait for Far-End H.245 Terminal Capability Set |
This field applies only to H.323 devices. This check box specifies that Cisco Unified Communications Manager waits to receive the far-end H.245 Terminal Capability Set before it sends its H.245 Terminal Capability Set. By default, the system checks this check box. To specify that Cisco Unified Communications Manager should initiate capabilities exchange, uncheck this check box. |
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Path Replacement Support |
If you choose the QSIG option from the Tunneled Protocol drop-down list box, this check box displays for H.225 trunks, gatekeeper-controlled trunks, and non-gatekeeper-controlled trunks. This setting works with QSIG tunneling (Annex M.1) to ensure that non-H.323 information gets sent on the leg of the call that uses path replacement.
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Transmit UTF-8 for Calling Party Name |
If the Transmit UTF-8 for Calling Party Name is checked to obtain the locale, the SIP trunk attempts to obtain the locale from the device. If that fails, the SIP trunk attempts to obtain the user locale from the Common Device Configuration and if that fails the SIP trunk obtains the user locale used for the Enterprise Parameters. |
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Unattended Port |
Check this check box if calls can be redirected, transferred and forwarded to an unattended port, such as a voice mail port. The default value for this check box leaves it unchecked. |
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SRTP Allowed |
Check the SRTP Allowed check box if you want Cisco Unified Communications Manager to allow secure and nonsecure calls over the trunk. If you do not check this check box, Cisco Unified Communications Manager prevents SRTP negotiation with the trunk and uses RTP.
For more information on encryption for trunks, see the Cisco Unified Communications Manager Security Guide. |
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H.235 Pass Through Allowed |
This feature allows Cisco Unified Communications Manager to transparently pass through the shared secret (Diffie-Hellman key) and other H.235 data between two H.235 endpoints, so the two endpoints can establish a secure media channel. To allow H.235 pass through, check the check box. |
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Enable SAF |
Check this check box if you want to enable this intercluster (non-gatekeeper controlled) trunk for SAF. When a trunk is enabled for SAF, the trunk can support the call control discovery feature. SAF-enabled trunks that are assigned to the CCD advertising service in the Advertising Service window handle inbound calls from remote call-control entities that use the SAF network. ( ) SAF-enabled trunks that are assigned to the CCD requesting service handle outgoing calls to learned patterns. ( )For more information on the call control discovery feature, see the Cisco Unified Communications Manager Features and Services Guide. |
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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:
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. |
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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. |
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Intercompany Media Engine (IME) |
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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 ( ).For more information on Cisco Intercompany Media Engine, see the Cisco Intercompany Media Engine Installation and Configuration Guide. |
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Incoming Calling Party Settings |
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Clear Prefix Setting |
To delete all prefixes for all calling party number types, click Clear Prefix Settings. |
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Default Prefix Setting |
To enter the default value for all prefix fields at the same time, click Default Prefix Settings. |
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National Number |
Configure the following settings to globalize calling party numbers that use National for the Calling Party Number Type.
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International Number |
Configure the following settings to globalize calling party numbers that use International for the Calling Party Number Type.
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Subscriber Number |
Configure the following settings to globalize calling party numbers that use Subscriber for the Calling Party Number Type.
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Unknown Number |
Configure the following settings to globalize calling party numbers that use Unknown for the Calling Party Number Type.
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Incoming Called Party Settings The H.323 protocol does not support the international escape character +. To ensure the correct prefixes, including the +, get applied to inbound calls over H.323 trunks, configure the incoming called party settings; that is, configuring the incoming called party settings ensures that when an inbound call comes from a H.323 trunk, Cisco Unified Communications Manager transforms the called party number back to the value that was originally sent over the trunk. |
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Clear Prefix Settings |
To delete all prefixes for all called party number types, click Clear Prefix Settings. |
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Default Prefix Settings |
To enter the default value for all prefix fields at the same time, click Default Prefix Settings. |
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National Number |
Configure the following settings to transform incoming called party numbers that use National for the Called Party Number Type.
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International Number |
Configure the following settings to transform incoming called party numbers that use International for the Called Party Number Type.
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Unknown Number |
Configure the following settings to transform incoming called party numbers that use Unknown for the Called Party Number Type.
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Subscriber Number |
Configure the following settings to transform incoming called party numbers that use Subscriber for the Called Party Number Type.
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Multilevel Precedence and Preemption (MLPP) Information |
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MLPP Domain |
From the drop-down list box, 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 was 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 was set for the MLPP Domain Identifier enterprise parameter. |
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MLPP Indication |
If available, this setting specifies whether a device that is capable of playing precedence tones will use the capability when it places an MLPP precedence call. From the drop-down list box, choose a setting to assign to this device from the following options:
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Call Routing Information |
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Inbound Calls |
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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 H.323 device. Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called. |
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Calling Search Space |
From the drop-down list box, select 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.
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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. |
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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 +. |
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Redirecting Number IE Delivery - Inbound |
Check this check box to accept the Redirecting Number IE in the incoming SETUP message to the Cisco Unified Communications Manager. (The UUIE part of the SETUP message includes the Redirecting Number IE.) Uncheck the check box to exclude the Redirecting Number IE. You use Redirecting Number IE for voice-messaging integration only. If your configured voice-messaging system supports Redirecting Number IE, you should check the check box.
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Enable Inbound FastStart |
Check this check box to enable the H.323 FastStart call connections on incoming calls. By default, the check box remains unchecked for the H.323 gateway. For intercluster calls, you must check the Enable Inbound FastStart check box on Cisco Unified Communications Manager servers in other clusters for the outbound FastStart feature to work. |
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Connected Party Settings |
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Connected Party Transformation CSS |
This setting is applicable only for inbound Calls. This setting allows you to transform the connected party number that Cisco Unified Communications Manager sends in another format, such as a DID or E.164 number. This setting is applicable while sending connected number for basic call as well as sending connected number after inbound call is redirected. Cisco Unified Communications Manager includes the transformed number in the Connected Number Information Element (IE) of CONNECT and NOTIFY 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.
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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. |
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Outbound Calls |
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Called Party Transformation CSS |
This setting allows you to send transformed called party number in SETUP message for outgoing calls. 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.
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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. |
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Calling Party Transformation CSS |
This setting allows you to send transformed calling party number in SETUP message for outgoing calls. Also when redirection occurs for outbound calls, this CSS will be used to transform the connected number sent from Cisco Unified Communications Manager side in outgoing NOTIFY 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.
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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. |
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Calling Party Selection |
Choose the directory number that is sent on an outbound call on a gateway. The following options specify which directory number is sent:
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Calling Line ID Presentation |
Cisco Unified Communications Manager uses calling line ID presentation (CLIP) as a supplementary service to control the display of the calling party number on the called party phone display screen. Choose Default if you do not want to change the presentation setting. Choose Allowed if you want calling number information to display. Choose Restricted if you do not want the calling number information to display. |
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Called Party IE Number Type Unknown |
Choose the format for the type of number in called party directory numbers. Cisco Unified Communications Manager sets the called directory number (DN) type. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans, such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the called directory number to be encoded to a non-national numbering plan type. Choose one of the following options:
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Calling Party IE Number Type Unknown |
Choose the format for the type of number in calling party directory numbers. Cisco Unified Communications Manager sets the calling directory number (DN) type. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans, such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the calling directory number to be encoded to a non-national numbering plan type. Choose one of the following options:
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Called Numbering Plan |
Choose the format for the numbering plan in called party directory numbers. Cisco Unified Communications Manager sets the called DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans, such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the called numbering plan to be encoded to a non-national numbering plan. Choose one of the following options:
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Calling Numbering Plan |
Choose the format for the numbering plan in calling party directory numbers. Cisco Unified Communications Manager sets the calling DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans, such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the calling numbering plan to be encoded to a non-national numbering plan. Choose one of the following options:
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Caller ID DN |
Enter the pattern, from 0 to 24 digits, that you want to use to format the caller ID on outbound calls from the trunk. For example, in North America
You can enter the international escape character +. |
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Display IE Delivery |
Check this check box to enable delivery of the display information element (IE) in SETUP and CONNECT messages for the calling and called party name delivery service.
|
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Redirecting Number IE Delivery - Outbound |
Check this check box to indicate the first redirecting number and the redirecting reason of the call when the call is forwarded. (The UUIE part of the outgoing SETUP message from the Cisco Unified Communications Manager includes the Redirecting Number IE.) Uncheck the check box to exclude the first Redirecting Number and the redirecting reason. You use Redirecting Number IE for voice-messaging integration only. If your configured voice-messaging system supports Redirecting Number IE, you should check the check box.
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Enable Outbound FastStart |
Check this check box to enable the H.323 FastStart feature on outgoing calls. By default, the check box remains unchecked for the H.323 gateway or trunk. When you check the Enable Outbound FastStart check box, you must set the Media Termination Point Required, Media Resource Group Lists, and Codec for Outbound FastStart. |
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Codec For Outbound FastStart |
Choose the codec for use with the H.323 device for an outbound FastStart call:
When you check the Enable Outbound FastStart check box, you must choose the codec for supporting outbound FastStart calls. |
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Gatekeeper Information (for gatekeeper-controlled H.225 trunks and intercluster trunks) |
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Gatekeeper Name |
Choose the gatekeeper that controls this trunk.
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Terminal Type |
Use the Terminal Type field to designate the type for all devices that this trunk controls. Always set this field to Gateway for normal trunk call admission control. |
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Technology Prefix |
Use this optional field to eliminate the need for entering the IP address of every Cisco Unified Communications Manager when configuring the gw-type-prefix on the gatekeeper:
For example, if you leave this field blank and you have two Cisco Unified Communications Managers with IP addresses of 10.1.1.2 and 11.1.1.3, enter the following gw-type-prefix command on the gatekeeper: gw-type-prefix 1#* default-technology gw ip 10.1.1.2 gw ip 11.1.1.3 If you enter 1#* in this field, enter the following gw-type-prefix command on the gatekeeper: gw-type-prefix 1#* default-technology |
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Zone |
Use this optional field to request a specific zone on the gatekeeper with which Cisco Unified Communications Manager will register. The zone specifies the total bandwidth that is available for calls between this zone and another zone:
See the command reference documentation for your gatekeeper for more information. |
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Remote Cisco Unified Communications Manager Information (for non-gatekeeper-controlled intercluster trunks) |
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Server 1 IP Address/Host Name |
Enter the IP address or host name of the first remote Cisco Unified Communications Manager that this trunk accesses. |
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Server 2 IP Address/Host Name |
Enter the IP address or host name of the second remote Cisco Unified Communications Manager that this trunk accesses.
|
||||
Server 3 IP Address/Host Name |
Enter the IP address or host name of the third remote Cisco Unified Communications Manager that this trunk accesses.
|
||||
UUIE Configuration |
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Passing Precedence Level Through UUIE |
Check this check box to enable passing MLPP information through the PRI 4ESS UUIE field. The system uses this box for interworking with DRSN switch. The system makes this check box available only if the PRI Protocol Type value of PRI 4ESS is specified for this trunk. The default value specifies unchecked. |
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Security Access Level |
Enter the value for the security access level. Valid values include 00 through 99. The system makes this field available only if the Passing Precedence Level Through UUIE check box is checked. The default value specifies 2. |
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Geolocation Configuration |
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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 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. |
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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 menu 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. |
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Send Geolocation Information |
Check this box to send geolocation information for this device. For an overview and details of how logical partitioning uses geolocation information, see the Cisco Unified Communications Manager Features and Services Guide. |
SIP Trunk Settings
The following table describes the settings for SIP trunks.
Field |
Description |
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Trunk Service Type |
Choose one of the following options from the Trunk Service Type drop-down list box:
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Device Information |
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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. |
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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 (<>). |
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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.
The default value for Device Pool specifies Not Selected. |
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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. |
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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. |
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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. |
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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 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. |
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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. |
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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).
|
||||||
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:
For more information, see the following information:
|
||||||
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:
For more information, see the following information:
|
||||||
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:
For more information on capturing packets, see the Troubleshooting Guide for Cisco Unified Communications Manager. |
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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. |
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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.
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. |
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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. |
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Path Replacement Support |
This check box displays when you select QSIG from the Tunneled Protocol drop-down list box. This setting works with QSIG tunneling to ensure that non-SIP information gets sent on the leg of the call that uses path replacement.
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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.
The default value for Transmit UTF-8 for Calling Party Name leaves the check box unchecked. |
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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. |
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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. |
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SRTP Allowed |
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.
For more information on encryption for trunks, see the Cisco Unified Communications Manager Security Guide. |
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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:
For more information on security and trunks, see the Cisco Unified Communications Manager Security Guide. |
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Route Class Signaling Enabled |
From the drop-down list, enable or disable route class signaling for the port. Choose one of the following values:
Route class signaling communicates special routing or termination requirements to receiving devices. It must be enabled for the port to support the Hotline feature. |
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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:
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. |
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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. |
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Run On All Active Unified CM Nodes |
To enable the trunk to run on every node, check this check box. |
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Intercompany Media Engine (IME) |
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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 ( ).For more information on Cisco Intercompany Media Engine, see the Cisco Intercompany Media Engine Installation and Configuration Guide. |
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Multilevel Precedence and Preemption (MLPP) Information |
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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. |
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Confidential Access Level |
Select the appropriate CAL value from the drop-down list box. |
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Confidential Access Mode |
From the drop-down list box, select one of the following options to set the CAL mode:
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Call Routing Information |
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Remote-Party-ID |
Use this check box to allow or disallow the SIP trunk to send the Remote-Party-ID (RPID) header in outgoing SIP messages from Cisco Unified Communications Manager to the remote destination. If you check this box, the SIP trunk always sends the RPID header. If you do not check this box, the SIP trunk does not send the RPID header.
Outgoing SIP Trunk Calls The configured values of the Calling Line ID Presentation and Calling Name Presentation provide the basis for the construction of the Privacy field of the RPID header. Each of these two options can have the values of Default, Allowed, or Restricted. If either option is set to Default, the corresponding information (Calling Line ID Presentation and/or Calling Name Presentation) in the RPID header comes from the Call Control layer (which is based on call-by-call configuration) within Cisco Unified Communications Manager. If either option is set to Allowed or Restricted, the corresponding information in the RPID header comes from the SIP trunk configuration window. Incoming SIP Trunk Calls The configured values of the Connected Line ID Presentation and Connected Name Presentation provide the basis for the construction of the Privacy field of the RPID header. Each of these two options can have the values of Default, Allowed, or Restricted. Be aware that the Connected Line ID Presentation and Connected Name Presentation options are relevant for 180/200 messages that the SIP trunk sends in response to INVITE messages that Cisco Unified Communications Manager receives. If either option is set to Default, the corresponding information (Connected Line ID Presentation and/or Connected Name Presentation) in the RPID header comes from the Call Control layer (which is based on call-by-call configuration) within Cisco Unified Communications Manager. If either option is set to Allowed or Restricted, the corresponding information in the RPID header comes from the SIP trunk configuration window.
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Asserted Identity |
Use this check box to allow or disallow the SIP trunk to send the Asserted-Type and SIP Privacy headers in SIP messages. If you check this check box, the SIP trunk always sends the Asserted-Type header; whether the SIP trunk sends the SIP Privacy header depends on the SIP Privacy configuration. If the check box is not selected, the SIP trunk does not include any Asserted-Type or SIP Privacy headers in its SIP messages. For more information, see the descriptions of Asserted-Type and SIP Privacy in this table. Outgoing SIP Trunk Calls—P Headers The decision of which Asserted Identity (either P-Asserted-Identity or P-Preferred-Identity) header gets sent depends on the configured value of the Asserted-Type option. A non-default value for Asserted-Type overrides values that come from Cisco Unified Communications Manager Call Control. If the Asserted-Type option is set to Default, the value of Screening Identification that the SIP trunk receives from Cisco Unified Communications Manager Call Control dictates the type of Asserted-Identity. Outgoing SIP Trunk Calls—SIP Privacy Header The SIP Privacy header gets used only when you check the Asserted Identity check box and when the SIP trunk sends either a PAI or PPI header. (Otherwise the SIP Privacy header neither gets sent nor processed in incoming SIP messages). The value of the SIP Privacy headers depends on the configured value of the SIP Privacy option. A non-default value for SIP Privacy overrides values that come from Cisco Unified Communications Manager Call Control. If the SIP Privacy option is set to Default, the Calling Line ID Presentation and Calling Name Presentation that the SIP trunk receives from Cisco Unified Communications Manager Call Control determines the SIP Privacy header. Incoming SIP Trunk Calls—P Headers The decision of which Asserted Identity (either P-Asserted-Identity or P-Preferred-Identity) header gets sent depends on the configured value of the Asserted-Type option. A non-default value for Asserted-Type overrides values that come from Cisco Unified Communications Manager Call Control. If the Asserted-Type option is set to Default, the value of Screening Identification that the SIP trunk receives from Cisco Unified Communications Manager Call Control dictates the type of Asserted-Identity. Incoming SIP Trunk Calls—SIP Privacy Header The SIP Privacy header gets used only when you check the Asserted Identity check box and when the SIP trunk sends either a PAI or PPI header. (Otherwise the SIP Privacy header neither gets sent nor processed in incoming SIP messages.) The value of the SIP Privacy headers depends on the configured value of the SIP Privacy option. A non-default value for SIP Privacy overrides values that come from Cisco Unified Communications Manager Call Control. If the SIP Privacy option is set to Default, the Connected Line ID Presentation and Connected Name Presentation that the SIP trunk receives from Cisco Unified Communications Manager Call Control determine the SIP Privacy header.
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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:
|
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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:
|
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Inbound Calls |
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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.
The default value for Significant Digits specifies All. |
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Connected Line ID Presentation |
Cisco Unified Communications Manager uses connected line ID presentation (COLP) as a supplementary service to provide the calling party with the connected party number. The SIP trunk level configuration takes precedence over the call-by-call configuration. The default value for Connected Line ID Presentation specifies Default, which translates to Allowed. Choose Default if you want Cisco Unified Communications Manager to send connected line information. Choose Restricted if you do not want Cisco Unified Communications Manager to send connected line information. If a call that originates from an IP phone on Cisco Unified Communications Manager encounters a device, such as a trunk, gateway, or route pattern, that has the Connected Line ID Presentation set to Default, the presentation value is automatically set to Allowed.
For more information about this field, see the Cisco Unified Communications Manager System Guide. |
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Connected Name Presentations |
Cisco Unified Communications Manager uses connected name ID presentation (CONP) as a supplementary service to provide the calling party with the connected party name. The SIP trunk level configuration takes precedence over the call-by-call configuration. The default value for Connected Name Presentation specifies Default, which translates to Allowed. Choose Default if you want Cisco Unified Communications Manager to send connected name information. Choose Restricted if you do not want Cisco Unified Communications Manager to send connected name information.
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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.
The default value for Calling Search Space specifies None. |
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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. |
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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 +. |
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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. |
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Incoming Calling Party Settings |
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Clear Prefix Setting |
To delete all prefixes for all calling party number types, click Clear Prefix Settings. |
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Default Prefix Setting |
To enter the default value for all prefix fields at the same time, click Default Prefix Settings. |
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Incoming Number |
Configure the following settings to globalize calling party numbers that use Unknown for the Calling Party Number Type.
|
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Incoming Called Party Settings |
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Clear Prefix Settings |
To delete the prefix for unknown number type for the called party, click Clear Prefix Settings. |
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Default Prefix Settings |
To enter the default value for the Prefix field for unknown number type, click Default Prefix Settings. |
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Unknown Number |
Configure the following settings to transform incoming called party numbers that use Unknown for the Called Party Number Type.
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Connected Party Settings |
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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.
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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. |
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Outbound Calls |
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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.
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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. |
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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.
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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. |
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Calling Party Selection |
Choose the directory number that is sent on an outbound call. The following options specify which directory number is sent:
The default value for Calling Party Selection specifies Originator. |
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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. |
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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.
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Caller ID DN |
Enter the pattern, from 0 to 24 digits, that you want to use to format the caller ID on outbound calls from the trunk. For example, in North America
You can enter the international escape character +. |
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Caller Name |
Enter a caller name to override the caller name that is received from the originating SIP Device. |
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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:
For more information on URI dialing, see the URI dialing chapter in the Cisco Unified Communications Manager System Guide. |
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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. |
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SIP Information |
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Destination Address |
The Destination Address represents the remote SIP peer with which this trunk will communicate. The allowed values for this field are an IP address, a fully qualified domain name (FQDN), or DNS SRV record only if the Destination Address is an SRV field is checked.
If the remote end is a Cisco Unified Communications Manager cluster, DNS SRV represents the recommended choice for this field. The DNS SRV record should include all Cisco Unified Communications Managers within the cluster. |
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Destination Address IPv6 |
The Destination IPv6 Address represents the remote SIP peer with which this trunk will communicate. You can enter one of the following values in this field:
SIP trunks only accept incoming requests from the configured Destination IPv6 Address and the specified incoming port that is specified in the SIP Trunk Security Profile that is associated with this trunk. If the remote end is a Cisco Unified Communications Manager cluster, consider entering the DNS SRV record in this field. The DNS SRV record should include all Cisco Unified Communications Managers within the cluster.
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Destination Address is an SRV |
This field specifies that the configured Destination Address is an SRV record. The default value specifies unchecked. |
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Destination Port |
Choose the destination port. Ensure that the value that you enter specifies any port from 1024 - 65535, or 0.
You need not enter a value if the destination address is an DNS SRV port. The default 5060 indicates the SIP port. The default value for Destination Port specifies 5060. |
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MTP Preferred Originating Codec |
Indicate the preferred outgoing codec:
This field gets used only when the MTP Termination Point Required check box is checked. |
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Presence Group |
Configure this field with the Presence feature. From the drop-down list box, choose a Presence group for the SIP trunk. The selected group specifies the destinations that the device/application/server that is connected to the SIP trunk can monitor. The default value for Presence Group specifies Standard Presence group, which gets configured with installation. Presence groups that are configured in Cisco Unified Communications Manager Administration also appear in the drop-down list box. Presence authorization works with presence groups to allow or block presence requests between groups. See the Cisco Unified Communications Manager Features and Services Guide for information about configuring permissions between groups.
|
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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 .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. |
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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).
The default value for Rerouting Calling Search Space specifies None. |
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Out-of-Dialog Refer Calling Search Space |
Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The out-of-dialog calling search space gets used when a Cisco Unified Communications Manager refers a call (B) that is coming into SIP user (A) to a third party (C) when no involvement of SIP user (A) exists. In this case, the system uses the out-of-dialog calling search space of SIP user (A). The default value for Out-of-Dialog Refer Calling Search Space specifies None. |
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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. |
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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. |
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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.
|
||||||
Normalization Script |
|||||||
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 ( ), and import a new script file. |
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Parameter Name/Parameter Value |
Optionally, enter parameter names and parameter values. Valid values include all characters except equals signs (=), semi-colons (;), and non-printable characters, such as tabs. You can enter a parameter name with no value. Example: Parameter Name Parameter Value CCA-ID 11223344 pbx location RTP You must choose a script from the Normalization Script drop-down list box before you can enter parameter names and values. To add another parameter line, click the + (plus) button. To delete a parameter line, click the - (minus) button. |
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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.
|
||||||
Recording Information |
|||||||
None |
To disable the trunk for call recording, click the Noneradio button. |
||||||
This trunk connects to a recording-enabled gateway |
To establish the recording session using the recording enabled Cisco gateway directly connected by this trunk, click This trunk connects to a recording-enabled gateway radio button.
|
||||||
This trunk connects to other clusters with trunk |
To establish recording session in the other cluster connected by this trunk, This trunk connects to other clusters with trunk radio button. |
||||||
Geolocation Configuration |
|||||||
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 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. |
||||||
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 menu 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. |
||||||
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. |