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:
Log in as provider, reseller, or customer administrator.
Set the hierarchy path to the node where the Cisco Unified Communications Manager (CUCM) is configured.
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.
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.
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.
In the Device Name field, enter a unique name for the new SIP trunk (or modify the existing device name, as applicable).
On the Device Information tab, complete at minimum, the mandatory Device Information Tab.
On the Call Routing General tab, complete at minimum, the mandatory Call Routing General Tab.
On the Call Routing Inbound tab, complete the required Call Routing Inbound Tab.
On the Call Routing Outbound tab, complete the required Call Routing Outbound Tab.
On the SP Info tab, complete the required SP Info Tab.
On the GeoLocation tab, complete at minimum, the mandatory GeoLocation Tab.
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 Tab¶
Option |
Description |
|---|---|
Device Name * |
Enter a unique identifier for the trunk using up to 50 alphanumeric characters: A-Z, a-z, numbers, hyphens (-) and underscores (_) only. Default value: None |
Trunk Service Type |
Choose one of:
Default value: None (Default) |
Description (Optional) |
Enter a descriptive name for the trunk using up to 114 characters in any language, but not including double-quotes ("), percentage sign (%), ampersand (&), backslash (\), or angle brackets (<>). Default value: empty |
Device Pool * |
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 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 |
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. Default value: None |
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 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 |
Option |
Description |
|---|---|
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. Default value: None |
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. Choose the appropriate location for this trunk:
Default value: Hub_None |
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 is attempted. Default value: None |
Tunneled Protocol |
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 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 |
QSIG Variant |
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:
Default value: No Changes |
Option |
Description |
|---|---|
ASN.1 ROSE OID Encoding |
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
Default value: No Changes |
Packet Capture Mode |
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:
Default value: None |
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. 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 |
Option |
Description |
|---|---|
Media Termination Point Required |
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: If the check box remains clear, Unified CM 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) |
Retry Video Call as Audio |
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 sending the call to call control for 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) |
Path Replacement Support |
This check box is relevant when you select QSIG from the Tunneled Protocol drop-down menu. This setting works with QSIG tunneling to ensure that non-SIP information gets sent on the leg of the call that uses path replacement. Default value: False (Clear) |
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 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 characters based on the user locale setting of the sending device pool. If the user locale setting matches the terminating phone user locale, the phone displays the characters. Note: The phone may display malformed characters if the two ends of the trunk are configured with user locales that do not belong to the same language group. Default value: False (Cleared) |
Option |
Description |
|---|---|
Transmit UTF-8 Names for 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 select 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. Default value: False (Cleared) |
Unattended Port |
Select this check box if calls can be redirected and transferred to an unattended port, such as a voice mail port. Default value: False (Cleared) |
SRTP Allowed |
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. 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) |
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. From the drop-down menu, select one of:
Default value: When using both sRTP and TLS |
Option |
Description |
|---|---|
Route Class Signaling Enabled |
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 value: Default |
Use Trusted Relay Point |
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 designated as a TRP. Select one of:
Default value: Default |
PSTN Access |
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 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 Unified CM Nodes |
Select this check box to enable the trunk to run on every node. Default value: False (Cleared) |
Call Routing General Tab¶
Option |
Description |
|---|---|
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 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, 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) |
Option |
Description |
|---|---|
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 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 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) |
Option |
Description |
|---|---|
Asserted-Type |
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:
Note: These headers get sent only if the Asserted- Identity check box is selected. Default value: Default |
SIP Privacy |
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:
Note: These headers get sent only if the Asserted Identity check box is selected. Default value: Default |
Call Routing Inbound Tab¶
Option |
Description |
|---|---|
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 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 |
Connected Line ID Presentation |
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
Note: Be aware that this service is not available when QSIG tunneling is enabled. Default value: Default |
Connected Name Presentation |
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
Note: Be aware that this service is not available when QSIG tunneling is enabled. Default value: Default |
Calling Search Space |
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 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 |
Option |
Description |
|---|---|
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. Default value: None |
Prefix DN |
Enter the prefix digits that are appended to the called party number on incoming calls. 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 |
Redirecting Diversion Header - Delivery Inbound |
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 Number in the incoming INVITE message to the Unified CM. 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) |
Incoming Calling Party - Prefix |
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 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 applies the service parameter configuration for the incoming calling party prefix, which supports both the prefix and strip digit functionality. Default value: None |
Incoming Calling Party - Strip Digits |
Enter the number of digits, up to the number 24, that you want Unified CM to strip from the calling party number of Unknown type before it applies the prefixes. Default value: None |
Incoming Calling Party - Calling Search Space |
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 assign to this device. Before the call 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 |
Option |
Description |
|---|---|
Incoming Calling Party - Use Device Pool CSS |
Select this check box to use the calling search space for the Unknown Number field that is configured in the device pool that is applied to the device. Default value: True (Selected) |
Incoming Called Party - Prefix |
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. Tip: 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 |
Incoming Called Party - Strip Digits |
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. Tip: 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 |
Incoming Called Party - Calling Search Space |
This setting allows you to transform the called party number of Unknown called party number type on the device. If you choose None, no transformation occurs for the 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 |
Incoming Called Party - Use Device Pool CSS |
Select this check box to use the calling search space for the Unknown Number field that is configured in the device pool that is applied to the device. Default value: True (Selected) |
Option |
Description |
|---|---|
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. 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 the connected party transformation pattern that you want to assign to this device. Note: If you configure the Connected Party Transformation CSS as None, the transformation does not match and does not get applied. Ensure that you configure the Calling Party Transformation pattern used for Connected Party Transformation in a non-null partition that is not used for routing. Default value: None |
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, select this check box. If you do not select this check box, the 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 Tab¶
Option |
Description |
|---|---|
Called Party Transformation CSS |
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. 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 |
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, select this check box. If you do not select this check box, the device uses the Called Party Transformation CSS that you configured for this device in the Trunk Configuration window. Default value: True (Selected) |
Calling Party Transformation CSS |
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 pattern that you want to assign to this device. Tip: If you configure the Calling Party Transformation CSS as None, the transformation does not match and does not get applied. Ensure that you configure the Calling Party Transformation Pattern in a non-null partition that is not used for routing. Default value: None |
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, select this check box. If you do not select this check box, the device uses the Calling Party Transformation CSS that you configured in the Trunk Configuration window. Default value: True (Selected) |
Calling Party Selection |
Choose the directory number that is sent on an outbound call. Select one of the following options to specify which directory number is sent:
Default value: Originator |
Option |
Description |
|---|---|
Calling Line ID Presentation |
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. Select one of
Default value: Default |
Calling Name Presentation |
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
Note: This service is not available when QSIG tunneling is enabled. Default value: Default |
Calling and Connected Party Info Format * |
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:
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 |
Option |
Description |
|---|---|
Redirecting Diversion Header Delivery - Outbound |
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 is forwarded. 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) |
Use Device Pool Redirecting Party Transformation CSS |
Select this check box to use the Redirecting Party Transformation CSS that is configured in the device pool that is assigned to this 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). |
Redirecting Party Transformation CSS |
Allows you to localize the redirecting party number on the device. 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. |
Caller Information Caller ID DN |
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:
You can also enter the international escape character +. Default value: None |
Caller Information - Caller Name |
Enter a caller name to override the caller name that is received from the originating SIP Device. Default value: None |
Caller Information - Maintain Original Caller ID DN and Caller Name in Identity Headers |
This check box is used to specify whether you will use the caller ID and caller name in the URI outgoing request. If you select this check box, the caller ID and caller name is used in the URI outgoing request. If 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 Tab¶
Option |
Description |
|---|---|
Destination Address is an SRV |
This field specifies that the configured Destination Address is an SRV record. Default value: False (Cleared) |
Destination - Destination Address IPv4 |
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: SIP trunks only accept incoming requests from the configured Destination Address and the 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 |
Destination - 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 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. |
Destination - Destination port |
Choose the destination port. Ensure that the value that you enter specifies any port from 1024 to 65535, or 0. Note: 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 |
Option |
Description |
|---|---|
Sort Order * |
Indicate the order in which the prioritize multiple destinations. A lower sort order indicates higher priority. This field requires an integer value. Default value: Empty |
Destination Address is an SRV |
This field specifies that the configured Destination Address is an SRV record. Default value: False (Cleared) |
Destination - Destination Address IPv4 |
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: SIP trunks only accept incoming requests from the configured Destination Address and the 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 |
Destination - 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 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. |
Option |
Description |
|---|---|
Destination - Destination port |
Choose the destination port. Ensure that the value that you enter specifies any port from 1024 to 65535, or 0. Note: 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 |
Sort Order * |
Indicate the order in which the prioritize multiple destinations. A lower sort order indicates higher priority. This field requires an integer value. Default value: Empty |
MTP Preferred Originating Codec |
Indicate the preferred outgoing codec by selecting one of:
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 |
BLF Presence Group * |
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.
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 |
Option |
Description |
|---|---|
SIP Trunk Security Profile * |
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 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. Default value: Non Secure SIP Trunk Profile |
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). Calling Search Space also applies to 3xx redirection and INVITE with Replaces features. Default value: None |
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 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 |
SUBSCRIBE Calling Search Space |
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 requests for the SIP trunk. All calling 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 |
SIP Profile * |
From the drop-down list box, select the SIP profile that is to be used for this SIP trunk. Default value: Standard SIP Profile |
Option |
Description |
|---|---|
DTMF Signaling Method |
Select one of:
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 |
Normalization Script |
From the drop-down menu, choose the script that you want to apply to this trunk. 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 |
Normalization Script - Enable Trace |
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. 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) |
Script Parameters |
Enter parameter names and values in the format Param1Name=Param1Value; Param2Name=Param2Value where Param1Name is the name of the first script parameter and 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. |
Recording Information |
Enter one of
|
GeoLocation Tab¶
Option |
Description |
|---|---|
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. On Unified CM, you can also choose a geolocation that has been configured with the System > Geolocation Configuration menu option. Default value: None |
Geolocation Filter |
From the drop-down menu, choose a geolocation filter. If you leave the <None> setting, no geolocation 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 Information |
Select this check box to send geolocation information for this device. Default value: False (Cleared) |
Delete a SIP Trunk¶
To delete a SIP trunk:
Log in as provider, reseller or customer administrator.
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.
From the list of trunks, choose the SIP trunk to be deleted.
Click Delete to delete the SIP trunk.
Click Yes to confirm the deletion.
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:
Log in as provider, reseller or customer administrator.
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.
From the list of SIP trunks, click the SIP trunk to be reset, then choose Action > Reset.
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 Reset a SIP Trunk.
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:
Log in as provider, reseller or customer administrator.
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.
From the list of trunks, click the SIP trunk to be restarted, then click Action > Restart.