address-hiding

Hides signaling and media peer addresses from endpoints other than the gateway.

anat

Enables Alternative Network Address Types (ANAT) on a SIP trunk.

answer-address

Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call.

application (global)

Enters application configuration mode to configure applications.

asserted-id

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.

asymmetric payload

Configures SIP asymmetric payload support.

audio forced

Allows only audio and image (for T.38 Fax) media types, and drops all other media types).

authentication

Enables SIP digest authentication.

bind

Binds the source address for signaling and media packets to the IPv4 or IPv6 address of a specific interface.

block

Configures global settings to drop (not pass) specific incoming SIP provisional response messages on a CUBE.

call spike

Configures the limit on the number of incoming calls received in a short period (a call spike).

call threshold global

Enables the global resources of a gateway.

call treatment action

Configures the action that the router takes when local resources are unavailable.

call treatment cause-code

Specifies the reason for the disconnection to the caller when local resources are unavailable.

call treatment isdn-reject

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.

call treatment on

Enables call treatment to process calls when local resources are unavailable.

callmonitor

Enables the call monitoring messaging functionality on a SIP endpoint in a VoIP network.

call-route

Enables header-based routing at the global configuration level.

cipher (voice class)

Configures the cipher setting, and associates it to a TLS profile.

clid

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.

codec preference

Specifies a list of preferred codecs to use on a dial peer.

codec profile

Defines audio and video capabilities that are needed for video endpoints.

codec transparent

Enables codec capabilities to be passed transparently between endpoints in a CUBE.

conn-reuse

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.

connection-reuse

Uses global listener port for sending requests over UDP.

contact-passing

Configures pass-through of the contact header from one leg to the other leg for 302 pass-through.

cpa

Enables the call progress analysis (CPA) algorithm for outbound VoIP calls and to set CPA parameters.

credentials

Configures a SIP TDM gateway or CUBE to send a SIP registration message when in the UP state.

crypto signaling

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.

dial-peer cor custom

Specifies that named class of restrictions (COR) apply to dial peers.

dial-peer cor list

Defines a class of restrictions (COR) list name.

disable-early-media 180

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).

dspfarm profile

Enters DSP farm profile configuration mode and defines a profile for DSP farm services.

dtmf-interworking

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.

early-media update block

Blocks the UPDATE requests with the Session Description Protocol (SDP) in an early dialog.

early-offer

Forces CUBE to send a SIP invite with Early Offer on the Out Leg.

emergency

Configures a list of emergency numbers.

error-code-override

Configures the SIP error code to be used at the dial peer.

error-passthru

Enables the passage of error messages from the incoming SIP leg to the outgoing SIP leg.

g729-annexb override

Configures the settings for G.729 codec interoperability and overrides the default value if the annexb attribute is not present.

gcid

Enables Global Call ID (GCID) for every call on an outbound leg of a VoIP dial peer for a SIP endpoint.

gw-accounting

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).

handle-replaces

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.

header-passing

Enables the passing of headers to and from SIP INVITE, SUBSCRIBE, and NOTIFY messages.

host-registrar

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.

http client connection idle timeout

Sets the number of seconds for which the HTTP client waits before terminating an idle connection.

http client connection persistent

Enables HTTP persistent connections so that multiple files can be loaded by using the same connection.

http client connection timeout

Sets the number of seconds for which the HTTP client waits for a server to establish a connection before abandoning its connection attempt.

ip qos dscp

Configures the DSCP value for QoS.

localhost

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.

max-conn

Specifies the maximum number of incoming or outgoing connections for a particular VoIP dial peer.

max-forwards

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.

media

Enables media packets to pass directly between endpoints without the intervention of CUBE, and enables signaling services.

media disable-detailed-stats

Disables the collection of detailed call statistics.

media profile asp

Creates a media profile to configure acoustic shock-protection parameters.

media profile nr

Creates a media profile to configure noise-reduction parameters.

media profile stream-service

Enables stream service on CUBE.

media profile video

Creates a media profile video.

media-address voice-vrf

Associates an RTP port range with VRF.

media-inactivity-criteria

Specifies the mechanism for detecting media inactivity (silence) on a voice call.

midcall-signaling

Configures the method that is used for signaling messages.

min-se

Changes the minimum session expiration (Min-SE) header value for all the calls that use the SIP session timer.

nat

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.

nat media-keepalive

Enables media keepalive packet transmission for the specified interval of time.

notify redirect

Enables application handling of redirect requests for all VoIP dial peers.

notify ignore substate

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.

notify telephone-event

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.

num-exp

Defines how to expand a telephone extension number into a particular destination pattern.

options-ping

Enables in-dialog options.

outbound-proxy

Configures a SIP outbound proxy for outgoing SIP messages globally.

pass-thru content

Enables the pass-through of SDP from in-leg to the out-leg.

permit hostname

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.

privacy

Sets privacy support at the global level as defined in RFC 3323.

privacy-policy

Configures the privacy header policy options at the global level.

progress_ind

Configures an outbound dial peer on a CUBE to override and remove or replace the default progress indicator in specified call messages.

protocol mode

Configures the Cisco IOS SIP stack.

random-contact

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.

reason-header override

Enables cause code passing from one SIP leg to another.

redirect ip2ip

Redirects SIP phone calls to SIP phone calls globally on a gateway.

redirection

Enables the handling of 3xx redirect messages

referto-passing

Disables dial peer lookup and modification of the Refer-To header when the CUBE passes across a REFER message during a call transfer.

registrar

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.

rel1xx

Enables SIP provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint.

remote-party-id

Enables translation of the Remote-Party-ID SIP header.

requri-passing

Enables pass-through of the host part of the Request-URI and To SIP headers.

retry bye

Configures the number of times that a BYE request is retransmitted to the other user agent.

retry invite

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.

rtcp all-pass-through

Passes through all the RTCP packets in the datapath.

rtcp keepalive

Configures RTCP keepalive report generation and generates RTCP keepalive packets.

rtp payload-type

Identifies the payload type of an RTP packet.

rtp-media-loop count

Configures the number of media loops before RTP voice and video media packets are dropped.

rtp-port

Configures the real-time protocol range.

rtp-ssrc multiplex

Multiplexes RTCP packets with RTP packets and sends multiple synchronization source in RTP headers (SSRCs) in an RTP session.

secure-ciphersuite

Configures the cipher suites (encryption algorithms) to be used for encryption over HTTPS for a WebSocket connection in CUBE.

session refresh

Enables SIP session refresh globally.

session transport

Configures a VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages.

set pstn-cause

Maps an incoming PSTN cause code to a SIP error status code.

set sip-status

Maps an incoming SIP error status code to a PSTN cause code.

signaling forward

Configures global settings for transparent tunneling of QSIG, Q.931, H.225, and ISUP messages.

silent discard untrusted

Discards SIP requests from untrusted sources in an incoming SIP trunk.

sip-server

Configures a network address for the SIP server interface.

srtp

Specifies that SRTP be used to enable secure calls and call fallback.

srtp negotiate

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.

stun

Enters STUN configuration mode for configuring firewall traversal parameters.

stun flowdata shared-secret

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.

stun usage firewall-traversal flowdata

Enables firewall traversal using STUN.

supplementary-service media-renegotiate

Globally enables midcall media renegotiation for supplementary services.

timers

Configures SIP-signaling timers.

transport

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.

uc secure-wsapi

Configures a secure Cisco Unified Communication IOS services environment for a specific application.

uc wsapi

Configures a nonsecure Cisco Unified Communication IOS services environment for a specific application.

update-callerid

Enables sending updates for caller IDs.

url (SIP)

Configures URLs to either the SIP, SIP secure (SIPS), or telephone (TEL) format for your VoIP SIP calls.

vad

Enables VAD for calls using a specific dial peer.

video codec

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.

voice cause code

Sets the internal Q850 cause code mapping for, voice and enters voice cause configuration mode.

voice class codec

Enters voice-class configuration mode and assigns an identification tag number for a codec voice class.

voice class dpg

Creates a dial-peer group for grouping multiple outbound dial peers.

voice class e164-pattern-map

Creates an E.164 pattern map that specifies multiple destination E.164 patterns in a dial peer.

voice class media

Configures media control parameters for voice.

voice class server-group

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.

voice-class sip options-keepalive

Monitors connectivity between CUBE VoIP dial peers and SIP servers.

voice class sip-copylist

Configures a list of entities to be sent to the peer call leg.

voice class sip-event-list

Configures a list of SIP events to be passed through.

voice class sip-hdr-passthrulist

Configures a list of headers to be passed through the route string.

voice-class sip nat media-keepalive

Configures media keepalive to enable media keepalive packets to be transmitted for the interval specified.

voice class sip-profiles

Configures SIP profiles for a voice class.

voice class srtp-crypto

Enters voice class configuration mode and assigns an identification tag for an srtp-crypto voice class command.

voice class uri

Creates or modifies a voice class for matching dial peers to a SIP or TEL URI.

voice class tls-cipher

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.

voice class tls-profile

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.

voice iec syslog

Enables viewing of internal error codes as they are encountered in real time.

voice statistics iec

Enables collection of internal error code statistics.

xfer target

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.