settings

Defines the time between 100 response and 18x response.

Sends ALERT to ISDN upon 183 receive.

Defines the index of the first call ringback tone in the Call-Progress Tones file.

Enables use of Alternative Route Reasons Table for 3xx.

Defines the far disconnect silence detection method.

Defines the silence period detection time.

Defines if a Message Manipulation Set is run again on incoming authenticated SIP messages received after the device sends a SIP 401 response for challenging initial incoming SIP REGISTER requests.

Defines the Authentication, Authorization and Accounting indications to use.

Defines the RADIUS accounting port.

Defines the RADIUS accounting server IP.

Enables empty Authorization header to be added to Register request.

Defines the AMD beep detection mode.

Defines the AMD mode.

Defines the "anonymous" mode.

Defines the SIP transport type.

Defines the Application Profile.

Enables backward compatibility for using parameters that configure Rx payload types for media features.

Defines the duration the RTP connection should be broken before the Broken Connection event is issued [100ms].

Enables trunks to be taken out of service in case of LAN down.

Defines how the device handles SIP Call-Info headers with multiple values in outgoing SIP messages.

Defines the Calling Number Play Back ID.

Defines the key sequence for call pickup.

Enables Call Transfer using re-INVITEs.

Enables call connection without on-hook/off-hook process 'Cut-Through'.

Defines the CDR report timing.

Defines the Syslog server IP address for sending CDRs.

Defines which IP address to use for classifying the incoming SIP dialog message to an IP Group, based on Proxy Set.

Defines the coder priority in SDP negotiation.

Disables Crypto life time in SDP.

Enables disconnect call upon detection of current disconnect signal.

Defines the default record location URI used by Media Ctrl.

Defines the Gateway delay time after restart (seconds).

Defines the delay between off-hook detection and Wink generation (FXS).

Enables sending INVITE message with/without SDP offer.

Defines the release cause sent to IP or Tel when device initiates release.

Enables use of different RTP port after hold.

Enables DID lines using Wink.

Defines if the device includes or excludes URI parameters for the Digest URI in the SIP Proxy-Authorization or Authorization headers of the request that the device sends in reply to a received SIP 401 (Unauthorized) or 407 (Proxy Authentication Required) response.

Enables automatic digit delivery to IP after call is connected.

Enables automatic digit delivery to Tel after line is off-hooked or seized.

Enables Play Code string to Tel when connect message received from IP.

Defines the behavior when receiving RTP or MSRP broken notification.

Enables disconnect calls on a configured silence timeout.

Enables display name to be used as source number.

Enables default port 5060 shown in the headers.

Defines the maximum time for an E911 ELIN callback to be valid (minutes).

Enables E911 to NG911 gateway and ELIN handling.

Defines the regret time for Emergency calls.

Defines emergency numbers.

set configuration

Enables RADIUS.

Enables DID for all FXS ports (that are are not enabled for DID per FXS port - see enable-did).

Enables requirement of ptime parameter in SDP.

Enables SIP secured URI usage.

Enables sending 408 responses for non-INVITE transactions.

Defines the AES-256 encryption key for encrypting (and decrypting) the SIP header value.

Defines the ENUM domain for ENUM resolution.

Defines if the device, upon receiving a SIP 503 response without a Retry-After header, behaves as if the 503 response included a Retry-After header and with the period (in seconds) specified by the parameter.

Enables enforcement reuse of TCP/TLS connection.

Enables rerouting of fax calls to fax destination.

Defines fax signaling method.

Enables filtering of calls to IP.

Enables the device to generate the 'tag' parameter's value in the SIP To header for SBC calls.

Enables responses sent to the UDP port from where the Request was sent, even if RPORT parameter was not received in the Via header.

Defines the forking delay time (in seconds) to wait before sending INVITE of second forking call.

Defines the Graceful Busy Out timeout in seconds.

Limits the device call time duration (minutes).

Enables History-Info header support.

Enables the device to retry registering even if the last SIP 401\407 response included "stale=false".

Ignores MKI if present in the remote SDP

Enables immediate trying sent upon INVITE receive.

Defines the mode to handle calls based on ip-addr defined in ip2tel-rte-tbl.

Defines the name of the attribute which represents the user display name in the Microsoft AD database.

Defines the name of the attribute which represents the user Mobile number in the Microsoft AD database.

Defines the name of the attribute which represents the user OCS number in the Microsoft AD database.

Defines the name of the attribute which represents the user PBX number in the Microsoft AD database.

Defines the name of the query primary key in the Microsoft AD database.

Defines the name of the attribute which represents the user Private number in the Microsoft AD database.

Defines the name of the query secondary key in the Microsoft AD database.

Defines the maximum timer for next request transmission after 491 response.

Defines the limit of number of concurrent calls.

Defines the maximum number of characters allowed in the SDP body's "o=" (originator and session identifier) field for the session ID and session version values.

Defines the Media CDR reports,

Defines the response type returned when a message is rejected according to the Message Policy.

Enables Microsoft proprietary Extension to modify called-nb.

Defines the time (in seconds) in the SIP Min-SE header, which is the minimum time that the user agent refreshes the session for Gateway calls.

Defines the minimum call duration.

Defines the maximum record duration supported by Microsoft.

Defines the format of multiple ptime (ptime per coder) in outgoing SDP.

Defines the call time duration limit (minutes).

Defines the maximum duration for an IVR dialog.

Defines the Network Node ID.

Rejects ISDN transfer requests.

Defines the NoAudio payload type.

Enables CDR message for all non-call dialogs.

Defines the number of concurrent non-responded dialogs.

Defines the Out-Of-Service Behavior for FXS.

Defines the user part of the user url in the P-Asserted-Identity header.

Defines the number of P-Preferred-Identity SIP headers included in the outgoing SIP message when the header contains multiple values.

Enables the masking of DTMF and other digits in syslog messages generated by the device.

Enables the PII masking (with asterisks) of URI host parts (including IP addresses) in CDRs that the device sends to Web, CLI, Syslog, REST, RADIUS, and Local Storage (depending on pii-mask-private-info-in-cdrs), or to OVOC if pii-mask-private-info-for-ovoc is enabled.

Enables the PII masking (with asterisks) of phone numbers, URI user parts, and display names in CDRs that the device sends to OVOC.

Enables the masking of personally identifiable information (PII) in CDRs and SDRs generated by the device.

Defines the number of PII characters to mask.

Defines from where to apply the PII mask, when the [PIIMaskPrivateInfoInCDRs] parameter is enabled.

Enables play Busy Tone to Tel.

Enables ringback tone playing towards IP.

Enables ringback tone playing towards Tel side.

Enables FXO Connect/Disconnect call upon detection of polarity reversal signal. FXS: generates the signal.

Defines the PRACK mode for 1XX reliable responses.

When the SBC sends out a SIP message that has multiple bodies, it enables the device to preserve the value of the Content-Type header (type and boundary) in the outgoing message.

Defines the whether to send the Progress Indicator to IP.

Defines the max time (in seconds) to wait for connect from PSTN.

Defines the release cause for SIT IC.

Defines the release cause for SIT RO.

Defines the release cause for SIT VC.

Defines the QoS period - if during this period [in seconds], no updated QOS info received, the old QOS info is discarded. if QOS poor, and no calls allowed, after this period, calls will be allowed again

Defines the number of samples to average.

Defines whether to add statistics to call release.

Defines the when RADIUS Accounting messages are sent.

Defines the percentage of active calls to send 'Almost out of resources' RAI.

Defines the time period to check call resources (seconds).

Defines the percentage of active calls to send 'Resources OK' RAI.

Defines the time to wait between phone hang up and call termination.

Enables Reason header in outgoing messages.

Defines the type of default record URI used by Media Ctrl.

Enables the device to send a SIP re-INVITE message with the local IP address of the new active device after a High-Availability (HA) switchover for current calls.

Note: The parameter is applicable only to Mediant VE in HA mode that is deployed on the Azure cloud platform.

Defines whether or not reject Cancel request after connect.

If set to false (0), a 503 response will not be sent on overload.

Defines the release cause mapping format.

Defines the release cause for SIT NC.

If set to 1 - AllTCP/TLS connections are set as persistent and will not be released.

Enables the Remote-Party-ID header.

Removes to-tag in final reject response for setup INVITE transaction.

Replaces Calling Number with Redirect Number ISDN to IP.

Replaces the number sign (#) with the escape character %23 in outgoing SIP messages.

Indicates whether or not Require header is able to contain the resource-priority tag.

Defines the device’s behavior when it receives a SIP 503 (Service Unavailable) containing a Retry-After header, in response to a SIP message (e.g., REGISTER) sent to a proxy server.

Retry After time for the proxy to be in state Unavailable.

Enables transcoding call.

Enables Rport option in Via header.

Enables RCTP attribute in the SDP.

Defines the RTCP-XR server IP address.

0:rtcpxr is not sent over SIP at all{@}1:rtcpxr is sent over sip when call ended{@}2:rtcpxr is sent over sip when on periodic interval and when call ended{@}3:rtcpxr is sent over sip when media segment ended and when call ended

Defines the RtcpXrEsc transport type.

On RTP only mode there is no signaling protocol (for media parameters negotiation with the remote side). The channel is open immediately. 0 - regular call establishment. 1 - The RTP channel open for Rx & Tx. 2- The RTP channel open only for Tx 3 -The RTP channel open only for Rx

Enables RTP Redundancy negotiation.

0:rtcpxr is not sent over SIP at all,1:rtcpxr is sent over sip when call ended

Defines echo canceller format for outgoing SDP.

Defines the SDP owner string.

Handle SDP offer/answer if SDP version was increased, otherwise takes SDP offer/answer parameters from last agreement (derived from previous SDP negotiations).

Defines from where the second calling number is taken from (in an incoming INVITE request).

Defines if resources self-check audit is used.

Sends 180 for call waiting.

Enables the use of the Global Session ID in SIP messages (AC-Session-ID header), which is a unique identifier of the call session, even if it traverses multiple devices.

Defines the SIP session - refreshed (using INVITE) each time this timer expires (seconds).

Defines the minimum time factor before the session expires.

Determines the Method to refresh the SIP session.

Defines the signaling cpu usage threshold alarm (percentage)

Defines the Silk max average bitrate (bps).

Enables  single DSP for G.711 to LBR coder.

Defines the default SIP destination port (usually 5060).

if set to 1, handle re-INVITE with a=recvonly as a=inactive

Defines the maximum number of retransmissions.

If not set, the incoming request will be always processed as user NOT behind NAT

Enables remote management of device by receiving NOTIFY request with specific event type.

Defines the SIP T.38 Version.

Use Tel uri or Sip uri for Diversion header.

Defines the release cause for SIT.

0 (default): Disable, 1:Add special header with capabilities for Skype

Select source header for called number (IP->TEL), either from the user part of To header or the P-Called-Party-ID header.

if set to 1 Use source number as display name if empty.if set to 2 always use source number as display name .{@}if set to 3 use the source number before manipulation, if empty.

Defines from where the source number is taken (in an incoming INVITE request).

Defines the SIP T1 timeout for retransmission.

Defines the SIP T2 timeout for retransmission.

Defines the fax max buffer size in T.38 SDP negotiation.

Defines the T.38 coder max datagram size.

T.38 Fax Session Immediate Start (Fax behind NAT)

Defines the T.38 packets received on RTP port.

Defines the interval between subsequent keep-alive probes, regardless of what the connection has exchanged in the meantime.

Defines the number of unacknowledged probes to send before considering the connection down and notifying the application layer.

Defines the interval between the last data packet sent (simple ACKs are not considered data) and the first keepalive probe.

Defines the SIP TCP time out.

Defines the Tel-to-IP call forking mode.

Defines the time between first and second Wink generation (FXS).

Defines the TR-104 Voice Profile Name.

Defines the Transparent code presentation.

Defines the payload type of the Transparent coder for outgoing data calls (ISDN-to-IP).

Enables the device to unregister all user Accounts that were registered with the device, upon a device restart.

Enables use of Tel uri or Sip uri for P-Asserted or P-Preferred headers.

If enabled, we will use URI from To/From headers in Refer-To header. If disabled, we will take the URI from Contact

Enables use of destination as connected number.

Enables use of DTG parameter.

Enables use of Tgrp information.

Defines the string that is displayed in the SIP Header 'User-Agent' or 'Server'.

Adds User=Phone parameter to SIP URL.

Defines the SIP subject.

Defines the interworking between the SIP INVITE's User-to-User header.

Defines whether to verify Request URI Header in requests.

Defines whether to verify Source IP with IP in top-most Via.

Defines the period at which web socket PING messages are sent.

Enables X-Channel header.