CUBE Commands
The following table lists the commands that are supported by Cisco Catalyst SD-WAN CLI templates for CUBE configuration. Click a command name in the Command column to view information about the command, its syntax, and its use.
Command |
Description |
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Hides signaling and media peer addresses from endpoints other than the gateway. |
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Enables Alternative Network Address Types (ANAT) on a SIP trunk. |
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Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call. |
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Enters application configuration mode to configure applications. |
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Enables support for the asserted ID header in incoming SIP requests or response messages, and to send the asserted ID privacy information in outgoing SIP requests or response messages. |
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Configures SIP asymmetric payload support. |
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Allows only audio and image (for T.38 Fax) media types, and drops all other media types). |
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Enables SIP digest authentication. |
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Binds the source address for signaling and media packets to the IPv4 or IPv6 address of a specific interface. |
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Configures global settings to drop (not pass) specific incoming SIP provisional response messages on a CUBE. |
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Configures the limit on the number of incoming calls received in a short period (a call spike). |
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Enables the global resources of a gateway. |
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Configures the action that the router takes when local resources are unavailable. |
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Specifies the reason for the disconnection to the caller when local resources are unavailable. |
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Specifies the rejection cause code for ISDN calls when all ISDN trunks are busied out, but the switch ignores the busyout trunks and still sends ISDN calls into the gateway. |
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Enables call treatment to process calls when local resources are unavailable. |
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Enables the call monitoring messaging functionality on a SIP endpoint in a VoIP network. |
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Enables header-based routing at the global configuration level. |
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Configures the cipher setting, and associates it to a TLS profile. |
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Passes the network-provided ISDN numbers in an ISDN calling party information element screening indicator field, and removes the calling party name and number from the calling-line identifier in voice service voip configuration mode. Alternatively, allows the presentation of the calling number by substituting for the missing Display Name field in the Remote-Party-ID and From headers. |
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Specifies a list of preferred codecs to use on a dial peer. |
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Defines audio and video capabilities that are needed for video endpoints. |
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Enables codec capabilities to be passed transparently between endpoints in a CUBE. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Reuses the TCP connection of a SIP registration for an endpoint behind a firewall. |
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Uses global listener port for sending requests over UDP. |
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Configures pass-through of the contact header from one leg to the other leg for 302 pass-through. |
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Enables the call progress analysis (CPA) algorithm for outbound VoIP calls and to set CPA parameters. |
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Configures a SIP TDM gateway or CUBE to send a SIP registration message when in the UP state. |
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Identifies the trustpoint trustpoint-name keyword and argument that is used during the Transport Layer Security (TLS) handshake that corresponds to the remote device address. |
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Specifies that named class of restrictions (COR) apply to dial peers. |
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Defines a class of restrictions (COR) list name. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Specifies which call treatment, early media or local ringback, is provided for 180 responses with 180 responses with Session Description Protocol (SDP). |
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Enters DSP farm profile configuration mode and defines a profile for DSP farm services. |
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Enables a delay between the dtmf-digit begin and dtmf-digit end events in the RFC 2833 packets sent from CUBE, and generates RFC 4733 compliance RTP Named Telephony Event (NTE) packets from CUBE. |
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Blocks the UPDATE requests with the Session Description Protocol (SDP) in an early dialog. |
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Forces CUBE to send a SIP invite with Early Offer on the Out Leg. |
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Configures a list of emergency numbers. |
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Configures the SIP error code to be used at the dial peer. |
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Enables the passage of error messages from the incoming SIP leg to the outgoing SIP leg. |
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Configures the settings for G.729 codec interoperability and overrides the default value if the annexb attribute is not present. |
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Enables Global Call ID (GCID) for every call on an outbound leg of a VoIP dial peer for a SIP endpoint. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Enables an accounting method for collecting call detail records (CDRs). |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Configures a Cisco IOS device to handle SIP INVITE with Replaces header messages at the SIP protocol level. |
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Enables the passing of headers to and from SIP INVITE, SUBSCRIBE, and NOTIFY messages. |
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Populates the sip-ua registrar domain name or IP address value in the host portion of the diversion header and redirects the contact header of the 302 response. |
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Sets the number of seconds for which the HTTP client waits before terminating an idle connection. |
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Enables HTTP persistent connections so that multiple files can be loaded by using the same connection. |
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Sets the number of seconds for which the HTTP client waits for a server to establish a connection before abandoning its connection attempt. |
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Configures the DSCP value for QoS. |
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Globally configures CUBE to substitute a DNS hostname or domain as the localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers in outgoing messages. |
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Specifies the maximum number of incoming or outgoing connections for a particular VoIP dial peer. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Globally sets the maximum number of hops, that is, proxy or redirect servers that can forward the SIP request. |
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Enables media packets to pass directly between endpoints without the intervention of CUBE, and enables signaling services. |
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Disables the collection of detailed call statistics. |
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Creates a media profile to configure acoustic shock-protection parameters. |
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Creates a media profile to configure noise-reduction parameters. |
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Enables stream service on CUBE. |
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Creates a media profile video. |
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Associates an RTP port range with VRF. |
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Specifies the mechanism for detecting media inactivity (silence) on a voice call. |
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Configures the method that is used for signaling messages. |
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Changes the minimum session expiration (Min-SE) header value for all the calls that use the SIP session timer. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Uses SIP Network Address Translation (NAT) global configuration. |
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Enables media keepalive packet transmission for the specified interval of time. |
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Enables application handling of redirect requests for all VoIP dial peers. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Specifies Ignoring the Subscription-State header in a Notify message. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Configures the maximum interval between two consecutive NOTIFY messages for a particular telephone event. |
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Defines how to expand a telephone extension number into a particular destination pattern. |
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Enables in-dialog options. |
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Configures a SIP outbound proxy for outgoing SIP messages globally. |
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Enables the pass-through of SDP from in-leg to the out-leg. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Stores hostnames used during validation of initial incoming INVITE messages. |
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Sets privacy support at the global level as defined in RFC 3323. |
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Configures the privacy header policy options at the global level. |
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Configures an outbound dial peer on a CUBE to override and remove or replace the default progress indicator in specified call messages. |
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Configures the Cisco IOS SIP stack. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Populates an outgoing INVITE message with random-contact information instead of clear-contact information. |
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Enables cause code passing from one SIP leg to another. |
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Redirects SIP phone calls to SIP phone calls globally on a gateway. |
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Enables the handling of 3xx redirect messages |
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Disables dial peer lookup and modification of the Refer-To header when the CUBE passes across a REFER message during a call transfer. |
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Enables SIP gateways to register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and SCCP phones with an external SIP proxy or SIP registrar. |
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Enables SIP provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint. |
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Enables translation of the Remote-Party-ID SIP header. |
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Enables pass-through of the host part of the Request-URI and To SIP headers. |
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Configures the number of times that a BYE request is retransmitted to the other user agent. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Configures the number of times that a SIP INVITE request is retransmitted to the other user agent. |
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Passes through all the RTCP packets in the datapath. |
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Configures RTCP keepalive report generation and generates RTCP keepalive packets. |
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Identifies the payload type of an RTP packet. |
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Configures the number of media loops before RTP voice and video media packets are dropped. |
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Configures the real-time protocol range. |
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Multiplexes RTCP packets with RTP packets and sends multiple synchronization source in RTP headers (SSRCs) in an RTP session. |
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Configures the cipher suites (encryption algorithms) to be used for encryption over HTTPS for a WebSocket connection in CUBE. |
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Enables SIP session refresh globally. |
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Configures a VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages. |
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Maps an incoming PSTN cause code to a SIP error status code. |
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Maps an incoming SIP error status code to a PSTN cause code. |
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Configures global settings for transparent tunneling of QSIG, Q.931, H.225, and ISUP messages. |
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Discards SIP requests from untrusted sources in an incoming SIP trunk. |
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Configures a network address for the SIP server interface. |
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Specifies that SRTP be used to enable secure calls and call fallback. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Enables the Cisco IOS Session Initiation Protocol (SIP) gateway to accept and send a Real-Time Transport Protocol (RTP) Audio/Video Profile (AVP) at the global configuration level. |
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Enters STUN configuration mode for configuring firewall traversal parameters. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Configures a secret shared on a call control agent. |
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Enables firewall traversal using STUN. |
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Globally enables midcall media renegotiation for supplementary services. |
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Configures SIP-signaling timers. |
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Configures the SIP user agent (gateway) for SIP signaling messages in inbound calls through the SIP TCP, TLS over TCP, or UDP socket. This command supports TLS version 1.3 and all associated ciphers. |
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Configures a secure Cisco Unified Communication IOS services environment for a specific application. |
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Configures a nonsecure Cisco Unified Communication IOS services environment for a specific application. |
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Enables sending updates for caller IDs. |
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Configures URLs to either the SIP, SIP secure (SIPS), or telephone (TEL) format for your VoIP SIP calls. |
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Enables VAD for calls using a specific dial peer. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Specifies a video codec for a voice class. |
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Sets the internal Q850 cause code mapping for, voice and enters voice cause configuration mode. |
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Enters voice-class configuration mode and assigns an identification tag number for a codec voice class. |
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Creates a dial-peer group for grouping multiple outbound dial peers. |
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Creates an E.164 pattern map that specifies multiple destination E.164 patterns in a dial peer. |
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Configures media control parameters for voice. |
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Enters voice-class configuration mode and configures server groups (groups of IPv4 and IPv6 addresses) that can be referenced from an outbound SIP dial peer. |
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Monitors connectivity between CUBE VoIP dial peers and SIP servers. |
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Configures a list of entities to be sent to the peer call leg. |
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Configures a list of SIP events to be passed through. |
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Configures a list of headers to be passed through the route string. |
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Configures media keepalive to enable media keepalive packets to be transmitted for the interval specified. |
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Configures SIP profiles for a voice class. |
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Enters voice class configuration mode and assigns an identification tag for an srtp-crypto voice class command. |
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Creates or modifies a voice class for matching dial peers to a SIP or TEL URI. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Configures an ordered set of TLS cipher suites. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Enables voice class configuration mode, and assigns an identification tag for a TLS profile. |
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Enables viewing of internal error codes as they are encountered in real time. |
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Enables collection of internal error code statistics. |
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Minimum supported releases: Cisco vManage Release 20.10.1 and Cisco IOS XE Catalyst SD-WAN Release 17.10.1a. Routes the INVITE to the refer-to destination in the REFER consume case. The routing decision is made based on the xfer target destination. |