.. _how_to_configure_sip_trunks: SIP Trunks ----------- This section describes how to add, edit, and delete SIP trunks, and how to reset or restart SIP trunks. Add and Edit SIP Trunks ............................ This procedure adds new SIP trunks and edits existing SIP trunks. **Perform these steps**: 1. Log in as provider, reseller, or customer administrator. 2. Set the hierarchy path to the node where the Cisco Unified Communications Manager (CUCM) is configured. 3. Choose an option: * If you're logged in as a Provider or Reseller admin, go to (default menus) **Apps Management > CUCM > SIP Trunks**. * If you're logged in as a Customer admin, go to (default menus) **Apps Management > Advanced > SIP Trunks**. 4. Do you want to ... * Add a new SIP trunk? Click **Add**, then go to Step 5. * Edit an existing SIP trunk? Click on the relevant SIP trunk in the list of SIP trunks; then, go to step 6. 5. From the **CUCM** drop-down menu, select the hostname, domain name, or IP address of the Unified CM where you want to add the SIP trunk. .. note:: The **CUCM** drop-down displays only when you're adding a new SIP trunk (not when editing). This drop-down menu displays the Unified CM located at the node, and all the Unified CM nodes in the hierarchies above the node where you're adding the SIP trunk. To provision a Unified CM server, see the Installation Tasks section of Installing Cisco Unified Communications Manager. 6. In the **Device Name** field, enter a unique name for the new SIP trunk (or modify the existing device name, as applicable). 7. On the **Device Information** tab, complete at minimum, the mandatory :ref:`device_information_fields`. 8. On the **Call Routing General** tab, complete at minimum, the mandatory :ref:`call_routing_general_fields`. 9. On the **Call Routing Inbound** tab, complete the required :ref:`call_routing_inbound_fields`. 10. On the **Call Routing Outbound** tab, complete the required :ref:`call_routing_outbound_fields`. 11. On the **SP Info** tab, complete the required :ref:`sp_info_fields`. 12. On the **GeoLocation** tab, complete at minimum, the mandatory :ref:`geolocation_fields`. 13. Click **Save** to save a new or updated SIP trunk. The SIP trunk appears in the SIP trunk list. The SIP trunk is automatically reset on the Unified CM as soon as it's added. To reset the SIP trunk at any other time, see "Reset SIP Trunk". To view the SIP trunk and its properties, log in to the Unified CM where you added the SIP trunk, select Device Trunk, and perform the "Find" operation. Clicking on the SIP trunk name in the list displays its characteristics. SIP Trunks Field Reference ............................... .. _device_information_fields: Device Information Tab '''''''''''''''''''''''''''' .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | Enter a unique identifier for the trunk using | | Device Name \* | up to 50 alphanumeric characters: A-Z, a-z, | | | numbers, hyphens (-) and underscores (_) | | | only. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | Choose one of: | | | | | | * **None** - Choose this option if the trunk | | | is not used for call control discovery, | | | Extension Mobility Cross Cluster, or | | | Cisco Intercompany Media Engine | | | * **Call Control Discovery** - Choose this | | | option to enable the trunk to support | | | call control discovery. | | | * **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 clear and become | | Trunk Service Type | 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. | | | * **Cisco Intercompany Media Engine** - Ensure| | | that the Cisco IME server is installed and | | | available before you configure this field. | | | * **IP Multimedia Subsystem Service Control | | | (ISC)** - Choose this option to enable the | | | trunk to support IP multimedia subsystem | | | service control. | | | | | | Default value: None (Default) | +------------------------+-----------------------------------------------+ | | Enter a descriptive name for the trunk using | | | up to 114 characters in any language, but not | | | including double-quotes (\"), percentage sign | | Description (Optional) | (%), ampersand (&), backslash (\\), or angle | | | brackets (<>). | | | | | | Default value: empty | +------------------------+-----------------------------------------------+ | | Choose the appropriate device pool for the | | | trunk. For trunks, device pools specify a | | | list of Cisco Unified Communications Managers | | | (Unified CMs) that the trunk uses to | | | distribute the call load dynamically. | | | | | | Note: | | | | | | Calls that are initiated from a phone that is | | | registered to a Unified CM that does not | | Device Pool \* | belong to the device pool of the trunk use | | | different Unified CMs of this device pool for | | | different outgoing calls. Selection of | | | Unified CM nodes occurs in a random order. A | | | call that is initiated from a phone that is | | | registered to a Unified CM that does belong | | | to the device pool of the trunk uses the same | | | Unified CM node for outgoing calls if the | | | Unified CM is up and running. | | | | | | Default value: Default | +------------------------+-----------------------------------------------+ | | Choose the common device configuration to | | Common Device | which you want this trunk assigned. The | | Configuration | common device configuration includes the | | | attributes (services or features) that are | | | associated with a particular user. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | 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 | | Call Classification | of the Unified CM 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. | | | | | | Default value: Use System Default | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | This list provides a prioritized grouping of | | | media resource groups. An application chooses | | Media Resource Group | the required media resource, such as a Music | | List | On Hold server, from among the available | | | media resources according to the priority | | | order that a Media Resource Group List | | | defines. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | 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. | | | | | | Choose the appropriate location for this | | Location \* | trunk: | | | | | | * Hub_None - Specifies that the locations | | | feature does not keep track of the | | | bandwidth that this trunk consumes. | | | * Phantom - Specifies a location that | | | enables successful CAC across | | | intercluster trunks that use H.323 | | | protocol or SIP. | | | * Shadow - Specifies a location for | | | intercluster enhanced location CAC. Valid | | | for SIP intercluster trunks (ICT) only. | | | | | | Default value: Hub_None | +------------------------+-----------------------------------------------+ | | Choose the automated alternate routing (AAR) | | | group for this device. The AAR group provides | | | the prefix digits that are used to route | | AAR Group | calls that are otherwise blocked due to | | | insufficient bandwidth. An AAR group setting | | | of None specifies that no rerouting of | | | blocked calls is attempted. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | Choose the QSIG option if you want to use SIP | | | trunks or SIP gateways to transport (tunnel) | | | QSI messages from Unified CM to other PINXs. | | | QSIG tunneling supports the following | | Tunneled Protocol | 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. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | To display the options in the **QSIG Variant**| | | drop-down list, choose QSIG from the | | | **Tunneled Protocol** drop-down menu. This | | | parameter specifies the protocol profile that | | | is sent in outbound QSIG facility information | | | elements. | | | | | | From the drop-down menu, select one of: | | | | | | * No Changes - Default. Keep this parameter | | QSIG Variant | set to the default value unless a VOSS | | | 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 | | | | | | Default value: No Changes | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | To display the options in the ASN.1 ROSE OID | | | Encoding drop-down menu, choose QSIG from the | | | **Tunneled Protocol** drop-down menu. This | | | parameter specifies how to encode the Invoke | | | Object ID (OID) for remote operations service | | | element (ROSE) operations. | | | | | | From the drop-down menu, select one of | | | | | | * No Changes - Keep this parameter set to | | | the default value unless a VOSS support | | | engineer instructs otherwise. | | ASN.1 ROSE OID | * Not Selected | | Encoding | | | | * Use Global Value ECMA - If you selected | | | the ECMA option from the QSIG Variant | | | drop-down menu, select this option. | | | * Use Global Value ISO - If you selected | | | the ISO option from the QSIG Variant | | | drop-down menu, select this option. | | | * Use Local Value | | | | | | Default value: No Changes | +------------------------+-----------------------------------------------+ | | This setting exists for troubleshooting | | | encryption only; packet capturing may cause | | | high CPU usage or call-processing | | | interruptions. | | | | | | From the drop-down menu, select one of: | | | | | | * 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 - Unified CM 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. Unified CM, which stores the file | | Packet Capture Mode | for seven days, also stores the keys that | | | encrypt the file in a secure location. | | | Unified CM 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. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | This setting exists for troubleshooting | | | encryption only; packet capturing may cause | | | high CPU usage or call-processing | | | interruptions. This field specifies the | | Packet Capture | maximum number of minutes that is allotted | | Duration | for one session of packet capturing. | | | | | | To initiate packet capturing, enter a value | | | other than 0 in the field. After packet | | | capturing completes, the value, 0, displays. | | | | | | Default value: 0 (zero), Range is from 0 to | | | 300 minutes | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | You can configure Unified CM SIP trunks to | | | always use an Media Termination Point (MTP). | | | Select this box to provide media channel | | | information in the outgoing INVITE request. | | | When this check box is selected, all media | | | channels must terminate and reoriginate on | | | the MTP device. If you clear the check box, | | | the Unified CM can decide whether calls are | | | to go through the MTP device or be connected | | | directly between the endpoints. | | | | | | Note: | | | | | Media Termination | If the check box remains clear, Unified CM | | Point Required | attempts 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 Unified CM 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, Unified CM | | | does not allocate an MTP because both phones | | | support RFC2833. So, by having the same type | | | of DTMF method supported on each phone, there | | | is no need for MTP. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | This check box pertains to outgoing SIP trunk | | | calls and does not impact incoming calls. By | | | default, the system selects 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 | | Retry Video Call as | sending the call to call control for | | Audio | rerouting. If you clear 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 using Automatic | | | Alternate Routing (AAR) and route list or | | | hunt list. | | | | | | Default value: True (Selected) | +------------------------+-----------------------------------------------+ | | This check box is relevant when you select | | | QSIG from the **Tunneled Protocol** drop-down | | Path Replacement | menu. This setting works with QSIG tunneling | | Support | to ensure that non-SIP information gets sent | | | on the leg of the call that uses path | | | replacement. | | | | | | Default value: False (Clear) | +------------------------+-----------------------------------------------+ | | 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 select 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 | | Transmit UTF-8 for | characters based on the user locale setting | | Calling Party Name | 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 are configured with | | | user locales that do not belong to the same | | | language group. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | 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 select this check box and the user | | | locale setting in the device pool matches the | | | terminating phone user locale, the device | | Transmit UTF-8 Names | sends unicode and encodes in UTF-8 format. If | | for QSIG APDU | 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. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | Select this check box if calls can be | | Unattended Port | redirected and transferred to an unattended | | | port, such as a voice mail port. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | Select this check box if you want Unified CM | | | to allow secure and nonsecure media calls | | | over the trunk. Selecting 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 select this check box, Unified | | | CM prevents SRTP negotiation with the trunk | | | and uses RTP negotiation instead. | | | | | SRTP Allowed | Caution: | | | | | | If you select this check box, we strongly | | | recommend 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 still works but the keys get | | | exposed in signaling and traces. In that | | | case, you must ensure the security of the | | | network between Unified CM and the | | | destination side of the trunk. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | 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. | | Consider Traffic on | | | This Trunk Secure | From the drop-down menu, select one of: | | | | | | * When using both sRTP and TLS | | | * When using sRTP Only - Displays when you | | | select the **SRTP Allowed** check box. | | | | | | Default value: When using both sRTP and TLS | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | From the drop-down menu, enable or disable | | | route class signaling for the port. Route | | | class signaling communicates special routing | | | or termination requirements to receiving | | | devices. It must be enabled for the port to | | | support the Hotline feature. | | | | | | From the drop-down menu, select one of: | | | | | | * Default - The device uses the setting | | Route Class Signaling | from the Route Class Signaling service | | Enabled | parameter | | | * Off - Enables route class signaling. This | | | setting overrides the Route Class | | | Signaling service parameter | | | * On - Disables route class signaling. This | | | setting overrides the Route Class | | | Signaling service parameter. | | | | | | Default value: Default | +------------------------+-----------------------------------------------+ | | From the drop-down menu, enable or disable | | | whether Unified CM inserts a trusted relay | | | point (TRP) device with this media endpoint. | | | A Trusted Relay Point (TRP) device designates | | | an MTP or transcoder device that is labeled | | | as Trusted Relay Point. Unified CM 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. If both TRP and RSVPAgent are | | | needed for the endpoint, Unified CM first | | | tries to find an RSVPAgent that can also be | | | used as a TRP. If both TRP and transcoder are | | | needed for the endpoint, Unified CM first | | | tries to find a transcoder that is also | | Use Trusted Relay | designated as a TRP. | | Point | | | | Select one of: | | | | | | * Default - The device uses the Use Trusted | | | Relay Point setting from the common | | | device configuration with which this | | | device associates | | | * Off - Disables 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 - Enables 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. | | | | | | Default value: Default | +------------------------+-----------------------------------------------+ | | If you use the Cisco Intercompany Media | | | Engine feature, select this check box to | | | indicate that calls made through this trunk | | | might reach the PSTN. Select this check box | | | even if all calls through this trunk device | | PSTN Access | do not reach the PSTN. For example, select | | | this check box for tandem trunks or an H.323 | | | gatekeeper routed trunk if calls might go to | | | the PSTN. When selected, this check box causes| | | the system to create upload voice call | | | records (VCRs) to validate calls made through | | | this trunk device. | | | | | | Default value: True (Selected) | +------------------------+-----------------------------------------------+ | Run On All Active | Select this check box to enable the trunk to | | Unified CM Nodes | run on every node. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ .. _call_routing_general_fields: Call Routing General Tab '''''''''''''''''''''''''''' .. tabularcolumns:: |p{3.5cm}|p{12cm}| +-------------------+---------------------------------------------------+ | Option | Description | +===================+===================================================+ | | Use this check box to allow or disallow the SIP | | | trunk to send the Remote-Party-ID (RPID) header | | | in outgoing SIP messages from Unified CM to the | | | remote destination. If you select this box, the | | | SIP trunk always sends the RPID header. If you do | | | not select this check box, the SIP trunk does not | | | send the RPID header. | | | | | | Note: | | | | | | Be aware that Calling Name Presentation, | | | Connected Line ID, and Connected Name | | | Presentation are not available when QSIG | | | tunneling is enabled. | | | | | | 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 Unified CM. If | | | either option is set to Allowed or Restricted, | | Remote-Party-ID | 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 Unified | | | CM 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 Unified CM. If either | | | option is set to Allowed or Restricted, the | | | corresponding information in the RPID header | | | comes from the SIP trunk configuration window. | | | | | | Note: | | | | | | The Remote-party ID and Asserted Identity options | | | represent independent mechanisms for | | | communication of display-identity information. | | | | | | Default value: True (Selected) | +-------------------+---------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +-------------------+---------------------------------------------------+ | Option | Description | +===================+===================================================+ | | 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 select this check | | | box, the SIP trunk always sends the Asserted-Type | | | header; whether or not the SIP trunk sends the | | | SIP Privacy header depends on the SIP Privacy | | | configuration. | | | | | | 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 | | | Unified CM Call Control. If the Asserted-Type | | | option is set to Default, the value of Screening | | | Identification that the SIP trunk receives from | | | Unified CM Call Control dictates the type of | | | Asserted-Identity. | | | | | | Outgoing SIP Trunk Calls - SIP Privacy Header | | | | | | The SIP Privacy header gets used only when you | | | select the **Asserted-Identity** check box and | | | when the SIP trunk sends either a Privacy-Asserted| | | Identity (PAI) or Privacy Preferred Identity | | | (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 Unified | | | CM 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 | | Asserted-Identity | Unified CM 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 | | | Unified CM Call Control. If the Asserted-Type | | | option is set to Default, the value of Screening | | | Identification that the SIP trunk receives from | | | Unified CM Call Control dictates the type of | | | Asserted-Identity. | | | | | | Incoming SIP Trunk Calls - SIP Privacy Header | | | | | | The SIP Privacy header gets used only when you | | | select 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 Unified CM 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 | | | Unified CM Call Control determine the SIP Privacy | | | header. | | | | | | Note: | | | | | | The Remote-party ID and Asserted Identity options | | | represent independent mechanisms for | | | communication of display-identity information. | | | | | | Default value: True (Selected) | +-------------------+---------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +-------------------+---------------------------------------------------+ | Option | Description | +===================+===================================================+ | | From the drop-down menu, select one of the | | | following values to specify the type of Asserted | | | Identity header that SIP trunk messages should | | | include: | | | | | | * Default - Screening information that the SIP | | | trunk receives from Unified CM Call Control | | | determines the type of header that the SIP | | | trunk sends. | | | * PAI - The Privacy-Asserted Identity header | | | gets sent in outgoing SIP trunk messages; | | Asserted-Type | this value overrides the Screening indication | | | value that comes from Unified CM. | | | * PPI - The Privacy Preferred Identity header | | | gets sent in outgoing SIP trunk messages; | | | this value overrides the Screening indication | | | value that comes from Unified CM. | | | | | | Note: | | | | | | These headers get sent only if the **Asserted- | | | Identity** check box is selected. | | | | | | Default value: Default | +-------------------+---------------------------------------------------+ | | From the drop-down menu, select 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 Unified CM | | | 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 | | | Unified CM. | | | | | SIP Privacy | * 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 Unified CM. | | | * 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 Unified CM. | | | | | | Note: | | | | | | These headers get sent only if the **Asserted | | | Identity** check box is selected. | | | | | | Default value: Default | +-------------------+---------------------------------------------------+ .. _call_routing_inbound_fields: Call Routing Inbound Tab '''''''''''''''''''''''''''' .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+------------------------------------------------+ | Option | Description | +========================+================================================+ | | 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. | | | | | Significant Digits | Choose the number of significant digits to | | | collect, from 0 to 32, or choose 99 to | | | indicate all digits. | | | | | | Note: | | | | | | Unified CM counts significant digits from the | | | right (last digit) of the number that is | | | called. | | | | | | Default value: 99 | +------------------------+------------------------------------------------+ | | Unified CM 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. | | | | | | Select one of | | | | | | * Default - Allowed. Choose Default if you | | | want Unified CM to send connected line | | | information. If a call that originates | | | from an IP phone on Unified CM encounters | | Connected Line ID | a device, such as a trunk, gateway, or | | Presentation | route pattern, that has the Connected | | | Line ID Presentation set to Default, the | | | presentation value is automatically set | | | to Allowed. | | | * Restricted - Choose Restricted if you do | | | not want Unified CM to send connected | | | line information. | | | | | | Note: | | | | | | Be aware that this service is not available | | | when QSIG tunneling is enabled. | | | | | | Default value: Default | +------------------------+------------------------------------------------+ | | Unified CM 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. | | | | | | Select one of | | | | | | * Default - Allowed. Choose Default if you | | Connected Name | want Unified CM to send connected name | | Presentation | information. | | | * Restricted - Choose Restricted if you do | | | not want Unified CM to send connected | | | name information. | | | | | | Note: | | | | | | Be aware that this service is not available | | | when QSIG tunneling is enabled. | | | | | | Default value: Default | +------------------------+------------------------------------------------+ | | From the drop-down menu, 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 menu by using the | | | Max List Box Items enterprise parameter. If | | | more calling search spaces exist than the Max | | Calling Search Space | 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. | | | | | | Default value: None | +------------------------+------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+------------------------------------------------+ | Option | Description | +========================+================================================+ | | Choose the appropriate calling search space | | | for the device to use when performing | | | automated alternate routing (AAR). The AAR | | AAR Calling Search | calling search space specifies the collection | | Space | of route partitions that are searched to | | | determine how to route a collected | | | (originating) number that is otherwise | | | blocked due to insufficient bandwidth. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | Enter the prefix digits that are appended to | | | the called party number on incoming calls. | | Prefix DN | Unified CM adds prefix digits after first | | | truncating the number in accordance with the | | | Significant Digits setting. You can enter the | | | international escape character \+. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | Select this check box to accept the | | | Redirecting Number in the incoming INVITE | | | message to the Unified CM. | | | | | | Clear the check box to exclude the | | Redirecting Diversion | Redirecting Number in the incoming INVITE | | Header - Delivery | message to the Unified CM. | | Inbound | | | | You use Redirecting Number for voice | | | messaging integration only. If your | | | configured voice-messaging system supports | | | Redirecting Number, you should select the | | | check box. | | | | | | Default value: False (Cleared) | +------------------------+------------------------------------------------+ | | Unified CM applies the prefix that you enter | | | in this field to calling party numbers that | | | use Unknown for the Calling Party Numbering | | | Type. You can enter up to 8 characters, which | | | include digits, the international escape | | | field, you cannot configure the Strip Digits | | | field. In this case, Unified CM takes the | | Incoming Calling Party | configuration for the Prefix and Strip Digits | | - Prefix | fields from the device pool that is applied | | | to the device. If the word, Default, displays | | | in the Prefix field in the Device Pool | | | Configuration window, Unified CM applies the | | | service parameter configuration for the | | | incoming calling party prefix, which supports | | | both the prefix and strip digit | | | functionality. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | Enter the number of digits, up to the number | | Incoming Calling Party | 24, that you want Unified CM to strip from | | - Strip Digits | the calling party number of Unknown type | | | before it applies the prefixes. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | This setting allows you to globalize the | | | calling party number of Unknown calling party | | | number type on the device. Make sure that the | | | calling party transformation CSS that you | | | choose contains the calling party | | | transformation pattern that you want to | | Incoming Calling Party | assign to this device. Before the call | | - Calling Search Space | occurs, the device must apply the | | | transformation by using digit analysis. If | | | you configure the 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. | | | | | | Default value: None | +------------------------+------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+------------------------------------------------+ | Option | Description | +========================+================================================+ | | Select this check box to use the calling | | Incoming Calling Party | search space for the Unknown Number field | | - Use Device Pool CSS | that is configured in the device pool that is | | | applied to the device. | | | | | | Default value: True (Selected) | +------------------------+------------------------------------------------+ | | Unified CM applies the prefix that you enter | | | in this field to called numbers that use | | | Unknown for the Called Party Number Type. You | | | can enter up to 16 characters, which include | | | digits, the international escape character | | | (\+), asterisk (*), or the pound sign (#). You | | | can enter the word, Default, instead of | | | entering a prefix. | | | | | Incoming Called Party | Tip: | | - Prefix | | | | If the word Default displays in the Prefix | | | field, you cannot configure the Strip Digits | | | field. In this case, Unified CM takes the | | | configuration for the Prefix and Strip Digits | | | fields from the device pool that is applied | | | to the device. If the word Default displays | | | in the Prefix field in the Device Pool | | | Configuration window, Unified CM does not | | | apply any prefix or strip digit | | | functionality. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | Enter the number of digits that you want | | | Unified CM to strip from the called party | | | number of Unknown type before it applies the | | | prefixes. | | | | | Incoming Called Party | Tip: | | - Strip Digits | | | | To configure the Strip Digits field, you must | | | leave the Prefix field blank or enter a valid | | | configuration in the Prefix field. To | | | configure the Strip Digits fields in these | | | windows, do not enter the word, Default, in | | | the Prefix field. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | This setting allows you to transform the | | | called party number of Unknown called party | | | number type on the device. If you choose | | Incoming Called Party | None, no transformation occurs for the | | - Calling Search Space | incoming called party number. Make sure that | | | the calling search space that you choose | | | contains the called party transformation | | | pattern that you want to assign to this | | | device. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | Select this check box to use the calling | | Incoming Called Party | search space for the Unknown Number field | | - Use Device Pool CSS | that is configured in the device pool that is | | | applied to the device. | | | | | | Default value: True (Selected) | +------------------------+------------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+------------------------------------------------+ | Option | Description | +========================+================================================+ | | 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. Unified | | | CM includes the transformed number in the | | | headers of various SIP messages, including | | | 200 OK and mid-call update and reinvite | | | messages. Make sure that the Connected Party | | | Transformation CSS that you choose contains | | Connected Party | the connected party transformation pattern | | Transformation CSS | 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. | | | | | | Default value: None | +------------------------+------------------------------------------------+ | | To use the Connected Party Transformation CSS | | | that is configured in the device pool that is | | Use Device Pool | assigned to this device, select this check | | Connected Party | box. If you do not select this check box, the | | Transformation CSS | device uses the Connected Party | | | Transformation CSS that you configured for | | | this device in the Trunk Configuration | | | window. | | | | | | Default value: True (Selected) | +------------------------+------------------------------------------------+ .. _call_routing_outbound_fields: Call Routing Outbound Tab '''''''''''''''''''''''''''' .. tabularcolumns:: |p{3.5cm}|p{12cm}| +-------------------------+----------------------------------------------+ | Option | Description | +=========================+==============================================+ | | This setting allows you to send the | | | transformed called party number in an 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. | | Called Party | | | Transformation CSS | 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. | | | | | | Default value: None | +-------------------------+----------------------------------------------+ | | To use the Called Party Transformation CSS | | | that is configured in the device pool that | | Use Device Pool Called | is assigned to this device, select this check| | Party Transformation | box. If you do not select this check box, the| | CSS | device uses the Called Party Transformation | | | CSS that you configured for this device in | | | the Trunk Configuration window. | | | | | | Default value: True (Selected) | +-------------------------+----------------------------------------------+ | | This setting allows you to send the | | | transformed calling party number in an | | | INVITE message for outgoing calls made over | | | a SIP Trunk. Also when redirection occurs | | | for outbound calls, this CSS is used to | | | transform the connected number that is sent | | | from Unified CM side in outgoing reINVITE / | | | UPDATE messages. Make sure that the Calling | | | Party Transformation CSS that you choose | | | contains the calling party transformation | | Calling Party | pattern that you want to assign to this | | Transformation CSS | 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. | | | | | | Default value: None | +-------------------------+----------------------------------------------+ | | To use the Calling Party Transformation CSS | | | that is configured in the device pool that | | Use Device Pool Calling | is assigned to this device, select this check| | Party Transformation | box. If you do not select this check box, the| | CSS | device uses the Calling Party Transformation | | | CSS that you configured in the Trunk | | | Configuration window. | | | | | | Default value: True (Selected) | +-------------------------+----------------------------------------------+ | | Choose the directory number that is sent on | | | an outbound call. Select one of the | | | following options to 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. | | Calling Party Selection | | | | * 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. | | | | | | Default value: Originator | +-------------------------+----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +-------------------------+----------------------------------------------+ | Option | Description | +=========================+==============================================+ | | Unified CM 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. | | | | | Calling Line ID | Select one of | | Presentation | | | | * Default - Allowed. Choose Default if you | | | want Unified CM to send calling number | | | information. | | | * Restricted - Choose Restricted if you do | | | not want Unified CM to send the calling | | | number information. | | | | | | Default value: Default | +-------------------------+----------------------------------------------+ | | Unified CM used 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. | | | | | | Select one of | | | | | | * Default - Allowed. Choose Default if you | | Calling Name | want Unified CM to send calling name | | Presentation | information. | | | * Restricted - Choose Restricted if you do | | | not want Unified CM to send the calling | | | name information. | | | | | | Note: | | | | | | This service is not available when QSIG | | | tunneling is enabled. | | | | | | Default value: Default | +-------------------------+----------------------------------------------+ | | This option allows you to configure whether | | | Unified CM 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 menu, select one of: | | | | | | * Deliver DN only in connected party - In | | | outgoing SIP messages, Unified CM | | | inserts the calling party - s directory | | | number in the SIP contact header | | | information. | | | * Deliver URI only in connected party, if | | | available - In outgoing SIP messages, | | | Unified CM inserts the sending party - s | | | directory URI in the SIP contact header. | | | If a directory URI is not available, | | | Unified CM inserts the directory number | | | instead. | | Calling and Connected | * Deliver URI and DN in connected party, | | Party Info Format \* | if available - In outgoing SIP messages, | | | Unified CM 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, Unified CM | | | 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 Unified CM | | | systems of Release 9.0 or greater, or | | | between a Cisco Unified Communications | | | Manager 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. | | | | | | Default value: Deliver DN only in connected | | | party | +-------------------------+----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +-------------------------+----------------------------------------------+ | Option | Description | +=========================+==============================================+ | | Select this check box to include the | | | Redirecting Number in the outgoing INVITE | | | message from the Unified CM to indicate the | | | original called party number and the | | | redirecting reason of the call when the call | | Redirecting Diversion | is forwarded. | | Header Delivery - | | | Outbound | Clear the check box to exclude the first | | | Redirecting Number and the redirecting | | | reason from the outgoing INVITE message. Use | | | Redirecting Number for voice-messaging | | | integration only. If your configured voice | | | messaging system supports Redirecting | | | Number, select the check box. | | | | | | Default value: False (Cleared) | +-------------------------+----------------------------------------------+ | | Select this check box to use the Redirecting | | Use Device Pool | Party Transformation CSS that is configured | | Redirecting Party | in the device pool that is assigned to this | | Transformation CSS | device. | | | | | | If you do not select this check box, the | | | device uses the Redirecting Party | | | Transformation CSS that you configured for | | | this device (see field below). | +-------------------------+----------------------------------------------+ | | Allows you to localize the redirecting | | Redirecting Party | party number on the device. | | Transformation CSS | | | | Make sure that the Redirecting Party | | | Transformation CSS that you enter contains | | | the redirecting party transformation pattern | | | that you want to assign to this device. | +-------------------------+----------------------------------------------+ | | Enter the pattern, from 0 to 24 digits that | | | you want to use to format the Called ID on | | | outbound calls from the trunk. For example, | | | in North America: | | | | | | * 555XXXX = Variable Caller ID, where X | | | represents an extension number. The | | | Central Office (CO) appends the number | | | with the area code if you do not specify | | Caller Information | it. | | Caller ID DN | | | | * 5555000 = Fixed Caller ID. Use this form | | | when you want the Corporate number to be | | | sent instead of the exact extension from | | | which the call is placed. The CO appends | | | the number with the area code if you do | | | not specify it. | | | | | | You can also enter the international escape | | | character \+. | | | | | | Default value: None | +-------------------------+----------------------------------------------+ | Caller Information - | Enter a caller name to override the caller | | Caller Name | name that is received from the originating | | | SIP Device. | | | | | | Default value: None | +-------------------------+----------------------------------------------+ | | This check box is used to specify whether | | Caller Information - | you will use the caller ID and caller name | | Maintain Original | in the URI outgoing request. If you select | | Caller ID DN and Caller | this check box, the caller ID and caller | | Name in Identity | name is used in the URI outgoing request. If | | Headers | you do not select this check box, the caller | | | ID and caller name is not used in the URI | | | outgoing request. | | | | | | Default value: False (Cleared) | +-------------------------+----------------------------------------------+ .. _sp_info_fields: SP Info Tab '''''''''''''' .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | Destination Address is | This field specifies that the configured | | an SRV | Destination Address is an SRV record. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | The Destination Address IPv4 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 | | | selected. | | | | | | Tip: | | | | | | For SIP trunks that can support IPv6 or IPv6 | | | and IPv4 (dual stack mode), configure the | | | Destination Address IPv6 field in addition to | | | the Destination Address field. | | | | | | Note: | | | | | Destination - | SIP trunks only accept incoming requests from | | Destination Address | the configured Destination Address and the | | IPv4 | specified incoming port that is specified in | | | the SIP Trunk Security Profile that is | | | associated with this trunk. | | | | | | Note: | | | | | | For configuring SIP trunks when you have | | | multiple device pools in a cluster, you must | | | configure a destination address that is a DNS | | | SRV destination port. Enter the name of a DNS | | | SRV port for the Destination Address and | | | select the Destination Address is an SRV | | | Destination Port check box. | | | | | | If the remote end is a Unified CM cluster, | | | DNS SRV represents the recommended choice for | | | this field. The DNS SRV record should include | | | all Unified CMs within the cluster. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | 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: | | | | | | * A fully qualified domain name (FQDN) | | | * A DNS SRV record, but only if the | | | Destination Address is an SRV field is | | | selected. | | | | | | SIP trunks only accept incoming requests from | | Destination - | the configured Destination IPv6 Address and | | Destination Address | the specified incoming port that is specified | | IPv6 | in the SIP Trunk Security Profile that is | | | associated with this trunk. | | | | | | If the remote end is a Unified CM cluster, | | | consider entering the DNS SRV record in this | | | field. The DNS SRV record should include all | | | Unified CMs within the cluster. | | | | | | Tip: | | | | | | For SIP trunks that run in dual-stack mode or | | | that support an IP Addressing Mode of IPv6 | | | Only, configure this field. If the SIP trunk | | | runs in dual-stack mode, you must also | | | configure the Destination Address field. | | | | | | Default value: None. If IPv4 field above is | | | completed, this field can be left blank. | +------------------------+-----------------------------------------------+ | | Choose the destination port. Ensure that the | | | value that you enter specifies any port from | | | 1024 to 65535, or 0. | | | | | | Note: | | Destination - | | | Destination port | You can now have the same port number that is | | | specified for multiple trunks. | | | | | | You do not need to enter a value if the | | | destination address is a DNS SRV port. The | | | default 5060 indicates the SIP port. | | | | | | Default value: 5060 | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | Indicate the order in which the prioritize | | Sort Order \* | multiple destinations. A lower sort order | | | indicates higher priority. This field | | | requires an integer value. | | | | | | Default value: Empty | +------------------------+-----------------------------------------------+ | Destination Address is | This field specifies that the configured | | an SRV | Destination Address is an SRV record. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | The Destination Address IPv4 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 | | | selected. | | | | | | Tip: | | | | | | For SIP trunks that can support IPv6 or IPv6 | | | and IPv4 (dual stack mode), configure the | | | Destination Address IPv6 field in addition to | | | the Destination Address field. | | | | | | Note: | | | | | Destination - | SIP trunks only accept incoming requests from | | Destination Address | the configured Destination Address and the | | IPv4 | specified incoming port that is specified in | | | the SIP Trunk Security Profile that is | | | associated with this trunk. | | | | | | Note: | | | | | | For configuring SIP trunks when you have | | | multiple device pools in a cluster, you must | | | configure a destination address that is a DNS | | | SRV destination port. Enter the name of a DNS | | | SRV port for the Destination Address and | | | select the Destination Address is an SRV | | | Destination Port check box. | | | | | | If the remote end is a Unified CM cluster, | | | DNS SRV represents the recommended choice for | | | this field. The DNS SRV record should include | | | all Unified CMs within the cluster. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | 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: | | | | | | * A fully qualified domain name (FQDN) | | | * A DNS SRV record, but only if the | | | Destination Address is an SRV field is | | | selected. | | | | | | SIP trunks only accept incoming requests from | | Destination - | the configured Destination IPv6 Address and | | Destination Address | the specified incoming port that is specified | | IPv6 | in the SIP Trunk Security Profile that is | | | associated with this trunk. | | | | | | If the remote end is a Unified CM cluster, | | | consider entering the DNS SRV record in this | | | field. The DNS SRV record should include all | | | Unified CMs within the cluster. | | | | | | Tip: | | | | | | For SIP trunks that run in dual-stack mode or | | | that support an IP Addressing Mode of IPv6 | | | Only, configure this field. If the SIP trunk | | | runs in dual-stack mode, you must also | | | configure the Destination Address field. | | | | | | Default value: None. If IPv4 field above is | | | completed, this field can be left blank. | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | Choose the destination port. Ensure that the | | | value that you enter specifies any port from | | | 1024 to 65535, or 0. | | | | | | Note: | | Destination - | | | Destination port | You can now have the same port number that is | | | specified for multiple trunks. | | | | | | You do not need to enter a value if the | | | destination address is a DNS SRV port. The | | | default 5060 indicates the SIP port. | | | | | | Default value: 5060 | +------------------------+-----------------------------------------------+ | | Indicate the order in which the prioritize | | Sort Order \* | multiple destinations. A lower sort order | | | indicates higher priority. This field | | | requires an integer value. | | | | | | Default value: Empty | +------------------------+-----------------------------------------------+ | | Indicate the preferred outgoing codec by | | | selecting one of: | | | | | | * 711ulaw | | | * 711alaw | | | * G729/G729a | | MTP Preferred | * G729b/G729ab | | Originating Codec | | | | 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. | | | | | | This field is used only when the **Media | | | Termination Point Required** check box is | | | selected on the **Device Information** tab. | | | | | | Default value: 711ulaw | +------------------------+-----------------------------------------------+ | | Configure this field with the Presence | | | feature. From the drop-down menu, select 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. | | | | | | * Standard Presence group is configured | | | with installation. Presence groups that | | | are configured in Unified CM | | | Administration also appear in the | | | drop-down menu. | | BLF Presence Group \* | * Presence authorization works with | | | presence groups to allow or block | | | presence requests between groups. | | | | | | Tip: | | | | | | You can apply a presence group to the SIP | | | trunk or to the application that is connected | | | to the SIP trunk. If a presence group is | | | configured for both a SIP trunk and SIP trunk | | | application, the presence group that is | | | applied to the application overrides the | | | presence group that is applied to the trunk. | | | | | | Default value: Standard Presence Group | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | Select the security profile to apply to the | | | SIP trunk. | | | | | | You must apply a security profile to all SIP | | | trunks that are configured in Unified CM | | SIP Trunk Security | Administration. Installing Cisco Unified | | Profile \* | 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. | | | | | | Default value: Non Secure SIP Trunk Profile | +------------------------+-----------------------------------------------+ | | 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 | | Rerouting Calling | the refer is completed, B and C connect. In | | Search Space | this case, the rerouting calling search space | | | that is used is that of the initial SIP user | | | (A). | | | | | | Calling Search Space also applies to 3xx | | | redirection and INVITE with Replaces | | | features. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | Calling search spaces determine the | | | partitions that calling devices can search | | | when they attempt to complete a call. The | | Out-Of-Dialog Refer | out-of-dialog calling search space gets used | | Calling Search Space | when a Unified CM 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). | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | Supported with the Presence feature, the | | | SUBSCRIBE calling search space determines how | | | Unified CM 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 menu, choose the SUBSCRIBE | | | calling search space to use for presence | | SUBSCRIBE Calling | requests for the SIP trunk. All calling | | Search Space | search spaces that you configure in Unified | | | CM Administration display in the SUBSCRIBE | | | Calling Search Space drop-down menu. | | | | | | If you do not select a different calling | | | search space for the SIP trunk from the | | | drop-down menu, the SUBSCRIBE calling search | | | space defaults to None. | | | | | | To configure a SUBSCRIBE calling search space | | | specifically for this purpose, configure a | | | calling search space as you do all calling | | | search spaces. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | From the drop-down list box, select the SIP | | SIP Profile \* | profile that is to be used for this SIP | | | trunk. | | | | | | Default value: Standard SIP Profile | +------------------------+-----------------------------------------------+ .. tabularcolumns:: |p{3.5cm}|p{12cm}| +------------------------+-----------------------------------------------+ | Option | Description | +========================+===============================================+ | | Select one of: | | | | | | * No Preference - Unified CM picks 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 | | | selected on the **Device Information** tab),| | | SIP trunk negotiates DTMF to RFC2833. | | | * RFC 2833 - Choose this configuration if | | | the preferred DTMF method to be used | | | across the trunk is RFC2833. Unified CM | | | makes every effort to negotiate RFC2833, | | DTMF Signaling Method | regardless of MTP usage. Out of band | | | (OOB) 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, Unified CM negotiates both | | | out-of-band and RFC2833 DTMF methods. As a | | | result, two DTMF events are sent for the same | | | DTMF keypress (one out of band and the other, | | | RFC2833). | | | | | | Default value: No Preference | +------------------------+-----------------------------------------------+ | | From the drop-down menu, choose the script | | | that you want to apply to this trunk. | | | | | Normalization Script | To import another script, on Unified CM go to | | | the SIP Normalization Script Configuration | | | window (Device > Device Settings > SIP | | | Normalization Script), and import a new | | | script file. | | | | | | Default value: None | +------------------------+-----------------------------------------------+ | | Select this check box to enable tracing within| | | the script or clear the check box to disable | | | tracing. When selected, the trace.output API | | | provided to the Lua scripter produces SDI | | | trace. | | Normalization Script - | | | Enable Trace | Note: | | | | | | We recommend that you only enable tracing | | | while debugging a script. Tracing impacts | | | performance and should not be enabled under | | | normal operating conditions. | | | | | | Default value: False (Cleared) | +------------------------+-----------------------------------------------+ | | Enter parameter names and values in the | | | format Param1Name=Param1Value; | | | Param2Name=Param2Value where Param1Name is | | | the name of the first script parameter and | | Script Parameters | Param1Value is the value of the first script | | | parameter. Multiple parameters can be | | | specified by putting semicolon after each | | | name and value pair . Valid values include | | | all characters except equal signs (=), | | | semi-colons (;); and non-printable | | | characters, such as tabs. You can enter a | | | parameter name with no value. | +------------------------+-----------------------------------------------+ | | Enter one of | | | | | | * 0 - None (default) | | Recording Information | | | | * 1 - This trunk connects to a | | | recording-enabled gateway | | | * 2 - This trunk connects to other | | | clusters with recording-enabled gateways | +------------------------+-----------------------------------------------+ .. _geolocation_fields: GeoLocation Tab '''''''''''''''''''' .. tabularcolumns:: |p{3.5cm}|p{12cm}| +--------------------+--------------------------------------------------+ | Option | Description | +====================+==================================================+ | | From the drop-down list box, choose a | | | geolocation. | | | | | | You can choose the Unspecified geolocation, | | Geolocation | which designates that this device does not | | | associate with a geolocation. | | | | | | On Unified CM, you can also choose a geolocation | | | that has been configured with the System > | | | Geolocation Configuration menu option. | | | | | | Default value: None | +--------------------+--------------------------------------------------+ | | From the drop-down menu, choose a geolocation | | | filter. | | | | | | If you leave the setting, no geolocation | | Geolocation Filter | filter gets applied for this device. | | | | | | On Unified CM, you can also choose a geolocation | | | filter that has been configured with the System | | | > Geolocation Filter menu option. | | | | | | Default value: None | +--------------------+--------------------------------------------------+ | Send Geolocation | Select this check box to send geolocation | | Information | information for this device. | | | | | | Default value: False (Cleared) | +--------------------+--------------------------------------------------+ .. _delete_sip_trunks: Delete a SIP Trunk ................... To delete a SIP trunk: 1. Log in as provider, reseller or customer administrator. 2. Choose an option: * If you logged in as Provider or Reseller administrator, go to **Apps Management > CUCM > SIP Trunks**. * If you logged in as Customer administrator, go to **Apps Management > Advanced > SIP Trunks**. 3. From the list of trunks, choose the SIP trunk to be deleted. 4. Click **Delete** to delete the SIP trunk. 5. Click **Yes** to confirm the deletion. .. _reset_sip_trunks: Reset a SIP Trunk .................... This procedure shuts down a SIP trunk and brings it back into service. .. note:: This procedure does not physically reset the hardware; it only re-initializes the configuration that is loaded by the Cisco Unified Communications Manager (CUCM) cluster. To restart a SIP trunk without shutting it down, use **Restart SIP Trunks**. **Perform these steps**: 1. Log in as provider, reseller or customer administrator. 2. Perform one of: * If you logged in as Provider or Reseller administrator, go to **Apps Management > CUCM > SIP Trunks**. * If you logged in as Customer administrator, go to **Apps Management > Advanced > SIP Trunks**. 3. From the list of SIP trunks, click the SIP trunk to be reset, then choose **Action > Reset**. .. _restart_sip_trunks: Restart SIP Trunks ------------------ This procedure restarts a SIP trunk without shutting it down first. .. note:: * To shut down a SIP trunk prior to the reset, see :ref:`reset_sip_trunks`. * If the SIP trunk is not registered with Cisco Unified Communications Manager, you cannot restart it. .. warning:: Restarting a SIP trunk drops all active calls that are using the trunk. **Perform these steps**: 1. Log in as provider, reseller or customer administrator. 2. Choose an option: * If you logged in as provider or reseller administrator, choose **Apps Management > CUCM > SIP Trunks**. * If you logged in as customer administrator, choose **Apps Management > Advanced > SIP Trunks**. 3. From the list of trunks, click the SIP trunk to be restarted, then click **Action > Restart**.