Provisioning Profile

Profile rule setting

Default setting: Empty

Provision Status

Indicate the status of last provisioning

Default setting: Empty

Provisioning Failure Reason

Reason for failure

Default setting: Empty

Dial Tone

Prompts you to enter a phone number. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.

Default setting: 350@-19,440@-19;10(*/0/1+2)

Second Dial Tone

Alternative to the Dial Tone when you dial a three-way call.

Default setting: 420@-19,520@-19;10(*/0/1+2)

Outside Dial Tone

Alternative to the Dial Tone. It prompts you to enter an external phone number, as opposed to an internal extension. A comma character in the dial plan triggers it.

Default setting: 420@-16;10(*/0/1)

Prompt Tone

Prompts you to enter a call forwarding phone number.

Default setting: 520@-19,620@-19;10(*/0/1+2)

Busy Tone

Played when a 486 RSC is received for an outbound call.

Default setting: 480@-19,620@-19;10(.5/.5/1+2)

Reorder Tone

Played when an outbound call has failed, or after the far end hangs up during an established call. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.

Default setting: 480@-19,620@-19;10(.25/.25/1+2)

Off Hook Warning Tone

Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when the Reorder Tone times out.

Default setting: 480@-10,620@0;10(.125/.125/1+2)

Ring Back Tone

Played during an outbound call when the far end is ringing.

Default setting: 440@-19,480@-19;*(2/4/1+2)

Ring Back 2 Tone

Your ATA plays this tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request.

Default setting: the same as Ring Back Tone, except the cadence is 1s on and 1s off.

Default setting: 440@-19,480@-19;*(1/1/1+2)

Confirm Tone

Brief tone to notify you that the last input value has been accepted.

Default setting: 600@-16;1(.25/.25/1)

SIT1 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT2 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)

SIT3 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT4 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)

MWI Dial Tone

Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.

Default setting: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Cfwd Dial Tone

Played when all calls are forwarded.

Default setting: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)

Holding Tone

Informs the local caller that the far end has placed the call on hold.

Default setting: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)

Conference Tone

Played to all parties when a three-way conference call is in progress.

Default setting: 350@-19;20(.1/.1/1,.1/9.7/1)

Secure Call Indication Tone

Played when a call has been successfully switched to secure mode. Play it for a short period - less than 30 seconds - and at a reduced level - less than 19 dBm - so it doesn't interfere with the call.

Default setting: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

Feature Invocation Tone

Played when a feature is implemented.

Default setting: 350@-16;*(.1/.1/1)

Call Remind Tone

The holding tone is played on the Phone ports during the active call to remind you of the held call.

Default setting: blank

Ring1 Cadence

Cadence script for distinctive ring 1.

Default setting: 60(2/4)

Ring2 Cadence

Cadence script for distinctive ring 2.

Default setting: 60(.8/.4,.8/4)

Ring3 Cadence

Cadence script for distinctive ring 3.

Default setting: 60(.4/.2,.4/.2,.8/4)

Ring4 Cadence

Cadence script for distinctive ring 4.

Default setting: 60(.3/.2,1/.2,.3/4)

Ring5 Cadence

Cadence script for distinctive ring 5.

Default setting: 1(.5/.5)

Ring6 Cadence

Cadence script for distinctive ring 6.

Default setting: 60(.2/.4,.2/.4,.2/4)

Ring7 Cadence

Cadence script for distinctive ring 7.

Default setting: 60(.4/.2,.4/.2,.4/4)

Ring8 Cadence

Cadence script for distinctive ring 8.

Default setting: 60(0.25/9.75)

CWT1 Cadence

Cadence script for distinctive CWT 1.

Default setting: *(.3/9.7)

CWT2 Cadence

Cadence script for distinctive CWT 2.

Default setting: 30(.1/.1, .1/9.7)

CWT3 Cadence

Cadence script for distinctive CWT 3.

Default setting: 30(.1/.1, .1/.1, .1/9.7)

CWT4 Cadence

Cadence script for distinctive CWT 4.

Default setting: 30(.1/.1, .3/.1, .1/9.3)

CWT5 Cadence

Cadence script for distinctive CWT 5.

Default setting: 1(.5/.5)

CWT6 Cadence

Cadence script for distinctive CWT 6.

Default setting: 30(.1/.1,.3/.2,.3/9.1)

CWT7 Cadence

Cadence script for distinctive CWT 7.

Default setting: 30(.3/.1,.3/.1,.1/9.1)

CWT8 Cadence

Cadence script for distinctive CWT 8.

Default setting: 2.3(.3/2)

Ring1 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call.

Default setting: Bellcore-r1

Ring2 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call.

Default setting: Bellcore-r2

Ring3 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.

Default setting: Bellcore-r3

Ring4 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call.

Default setting: Bellcore-r4

Ring5 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call.

Default setting: Bellcore-r5

Ring6 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call.

Default setting: Bellcore-r6

Ring7 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call.

Default setting: Bellcore-r7

Ring8 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call.

Default setting: Bellcore-r8

Ring Waveform

Waveform for the ringing signal. Choices are Sinusoid or Trapezoid.

Default setting: Trapezoid

Ring Frequency

Frequency of the ringing signal. Valid values are 15–50 (Hz)

Default setting: 20

Ring Voltage

Ringing voltage. Choices are 30–90 (V)

Default setting: 85

CWT Frequency

Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.

Default setting: 440@-10

Synchronized Ring

If this is set to yes, when the ATA is called, all lines ring at the same time (similar to a regular PSTN line). After one line answers, the others stop ringing.

Default setting: no

Hook Flash Timer Min.

Minimum on-hook time before off-hook qualifies as hook flash. Less than this value, and the on-hook event is ignored. Range: 0.1–0.4 seconds.

Default setting: 0.1

Hook Flash Timer.

Max Maximum on-hook time before off-hook qualifies as hook flash. More than this value, and the on-hook event is treated as on hook (no hook-flash event).

Range: 0.4–1.6 seconds.

Default setting: 0.9

Callee On Hook Delay.

Phone must be on-hook for time before the ATA tears down the current inbound call. It does not apply to outbound calls.

Range: 0–255 seconds.

Default setting: 0

Reorder Delay.

Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds.

Default setting: 5.

Call Back Expires.

Expiration time in seconds of a call back activation. Range: 0–65535 seconds.

Default setting: 1800

Call Back Retry Intvl.

Call back retry interval in seconds. Range: 0–255 seconds.

Default setting: 30

Call Back Delay.

Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the ATA still considers the call as failed and keeps on retrying. Range: 0–65 seconds.

Default setting: 0.5

VMWI Refresh Intvl.

Interval between VMWI refresh to the device. Range: 0–65535 seconds.

Default setting: 0

Interdigit Long Timer.

Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds.

Default setting: 10

Interdigit Short Timer.

Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.

Default setting: 3

CPC Delay.

Delay in seconds after caller hangs up when the ATA starts removing the tip-and-ring voltage to the attached equipment of the called party. The range is 0–255 seconds. This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected (remote end has answered) or disconnected (remote end has hung up) This feature should be disabled for the called party (in other words, by using the same polarity for connected and idle state) and the CPC feature should be used instead.

Without CPC enabled, reorder tone is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored. Resolution is 1 second.

Default setting: 2

CPC Duration.

Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and the dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second.

Default setting: 0.5

Call Return Code.

Call Return Code This code calls the last caller.

Default setting: *69

Call Redial Code.

Redials the last number called.

Default setting: *07

Blind Transfer Code.

Begins a blind transfer of the active call to the extension specified after the activation code.

Default setting: *98

Call Back Act Code.

Starts a callback when the last outbound call is not busy.

Default setting: *66

Call Back Deact Code.

Cancels a callback.

Default setting: *86

Call Back Busy Act Code.

Starts a callback when the last outbound call is busy.

Default setting: *05

Cfwd All Act Code.

Forwards all calls to the extension specified after the activation code.

Default setting: *72

Cfwd All Deact Code.

Cancels call forwarding of all calls.

Default setting: *73

Cfwd Busy Act Code.

Forwards busy calls to the extension specified after the activation code.

Default setting: *90

Cfwd Busy Deact Code.

Cancels call forwarding of busy calls.

Default setting: *91

Cfwd No Ans Act Code.

Forwards no-answer calls to the extension specified after the activation code.

Default setting: *92

Cfwd No Ans Deact Code.

Cancels call forwarding of no-answer calls.

Default setting: *93

Cfwd Last Act Code.

Forwards the last inbound or outbound call to the number that you specify after entering the activation code.

Default setting: *63

Cfwd Last Deact Code.

Cancels call forwarding of the last inbound or outbound call.

Default setting: *83

Block Last Act Code.

Blocks the last inbound call.

Default setting: *60

Block Last Deact Code.

Cancels blocking of the last inbound call.

Default setting: *80

Accept Last Act Code.

Accepts the last outbound call. It lets the call ring through when Do Not Disturb or call forwarding of all calls are enabled.

Default setting: *64

Accept Last Deact Code.

Cancels the code to accept the last outbound call.

Default setting: *84

CW Act Code.

Enables call waiting on all calls.

Default setting: *56

CW Deact Code.

Disables call waiting on all calls.

Default setting: *57

CW Per Call Act Code.

Enables call waiting for the next call.

Default setting: *71

CW Per Call Deact Code.

Disables call waiting for the next call.

Default setting: *70

Block CID Act Code.

Blocks caller ID on all outbound calls.

Default setting: *67

Block CID Deact Code.

Removes caller ID blocking on all outbound calls.

Default setting: *68

Block CID Per Call Act Code.

Blocks caller ID on the next outbound call.

Default setting: *81

Block CID Per Call Deact Code.

Removes caller ID blocking on the next inbound call.

Default setting: *82

Block ANC Act Code.

Blocks all anonymous calls.

Default setting: *77

Block ANC Deact Code.

Removes blocking of all anonymous calls.

Default setting: *87

DND Act Code.

Enables the Do Not Disturb feature.

Default setting: *78

DND Deact Code.

Disables the Do Not Disturb feature.

Default setting: *79

CID Act Code.

Enables caller ID generation.

Default setting: *65

CID Deact Code.

Disables caller ID generation.

Default setting: *85

CWCID Act Code.

Enables call waiting, caller ID generation.

Default setting: *25

CWCID Deact Code.

Disables call waiting, caller ID generation.

Default setting: *45

Dist Ring Act Code.

Enables the distinctive ringing feature.

Default setting: *26

Dist Ring Deact Code.

Disables the distinctive ringing feature.

Default setting: *46

Speed Dial Act Code.

Assigns a speed dial number.

Default setting: *74

Paging Code.

Used for paging other clients in the group.

Default setting: *96

Secure All Call Act Code.

Makes all outbound calls secure.

Default setting: *16

Secure No Call Act Code.

Makes all outbound calls not secure.

Default setting: *17

Secure One Call Act Code.

Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default).

Default setting: *18

Secure One Call Deact Code.

Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default).

Default setting: *19

Conference Act Code.

If this code is specified, you must enter it before dialing the third party for a conference call. Enter the code for a conference call.

Default setting: blank

Attn-Xfer Act Code.

If the code is specified, you must enter it before dialing the third party for a call transfer. Enter the code for a call transfer.

Default setting: blank

Modem Line Toggle Code.

Toggles the line to a modem. Modem passthrough mode can be triggered only by pre-dialing this code.

Default setting: *99

FAX Line Toggle Code.

Toggles the line to a fax machine.

Default setting: #99

Media Loopback Code.

Use for media loopback.

Default setting: *03

Referral Services Codes.

These codes tell the ATA what to do when you place the active call on hold and is listening to the second dial tone. One or more *codes can be configured into this parameter, such as *98, or *97|*98|*123, and so on. The maximum length is 79 characters. This parameter applies when you place the active call on hold by pressing the hook flash button. Each *code (and the following valid target number according to current dial plan) triggers the ATA to perform a blind transfer to a target number that is prepended by the service *code.

For example, after you dial *98, the ATA plays the Prompt Tone while waiting for you to enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the ATA sends a blind REFER to the holding party with the Refer-To target equal to *98 target_number. This feature allows the ATA to hand off a call to an application server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can empty the corresponding *code that you do not want the ATA to process.

Default setting: blank

Feature Dial Services Codes.

These codes let the ATA know what to do when you are listening to the first or second dial tone.

One or more *codes can be configured into this parameter, such as *72, or *72|*74|*67|*82, and so on. The maximum length is 79 characters. This parameter applies when you have a dial tone (first or second dial tone).

After receiving dial tone, you enters the *code and the target number according to current dial plan. For example, after you dial *72, the ATA plays a special tone called a Prompt tone while awaiting you to enter a valid target number. When a complete number is entered, the ATA sends a INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features like call forward (*72) or Block Caller ID (*67).

The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can remove a corresponding *code that you do not want to the ATA to process.

You can add a parameter to indicate which tone plays after the *code is entered, such as *72‘c‘|*67‘p‘. Below is a list of allowed tone parameters (note the use of open quotes surrounding the parameter, without spaces).

• 'c' = <Cfwd Dial Tone>
• 'd' = <Dial Tone>
• 'm' = <MWI Dial Tone>
• 'o' = <Outside Dial Tone>
• 'p' = <Prompt Dial Tone>
• 's' = <Second Dial Tone>
• 'x' = No tones are placed, x is any digit not used above.

If no tone parameter is specified, the ATA plays Prompt tone by default.

If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include this parameter. Instead, add the *code in the dial plan and the ATA send INVITE *73@..... as usual when you dial *73.

Default setting: blank

Service Annc Base Number

Base number for service announcements.

Default setting: blank

Service Annc Extension Codes

Extension codes for service announcements.

Default setting: blank

Prefer G711u Code.

Dial prefix to make G.711u the preferred codec for the call.

Default setting: *017110

Force G711u Code.

Dial prefix to make G.711u the only codec that can be used for the call.

Default setting: *027110

Prefer G711a Code.

Dial prefix to make G.711a the preferred codec for the call.

Default setting: *017111

Force G711a Code.

Dial prefix to make G.711a the only codec that can be used for the call.

Default setting: *027111

Prefer G726r32 Code.

Dial prefix to make G.726r32 the preferred codec for the call.

Default setting: *0172632

Force G726r32 Code.

Dial prefix to make G.726r32 the only codec that can be used for the call.

Default setting: *0272632

Prefer G729a Code.

Dial prefix to make G.729a the preferred codec for the call.

Default setting: *01729

Force G729a Code.

Dial prefix to make G.729a the only codec that can be used for the call.

Default setting: *02729

Line Enable

To enable this line for service, select yes. Otherwise, select no.

Default setting: yes

SAS Enable

To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller.

Default setting: no

SAS DLG Refresh Intvl

A non-zero value is the interval in which the streaming audio server sends out session refresh (SIP re-INVITE) messages to determine if the connection is active. If the caller does not respond to the refresh message, the ATA ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled).

Default setting: 30

SAS Inbound RTP Sink

This parameter works around devices that do not play inbound RTP if the SAS line declares itself as a send-only device and tells the client not to stream out audio. This parameter is an FQDN or IP address of an RTP sink to be used by the SAS line in the SDP of its 200 response to inbound INVITE from a client. It appears in the c = line and the port number appears in the m = line of the SDP.

If this value is not specified or is equal to 0, then c = 0.0.0.0 and a=sendonly are used in the SDP to tell the SAS client not to send any RTP to this SAS line. If a non-zero value is specified, then a=sendrecv and the SAS client streams audio to the given address.

Special case: If the value is $IP, then the SAS line’s own IP address is used in the c = line and a=sendrecv. In that case, the SAS client streams RTP packets to the SAS line.

Default setting: blank

NAT Mapping Enable

To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no.

Default setting: no

NAT Keep Alive Enable

To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.

Default setting: no

NAT Keep Alive Msg

Enter the keep alive message sent periodically to maintain the current NAT mapping.

Valid values are: $NOTIFY, $REGISTER, and $OPTIONS.

• $NOTIFY: A NOTIFY message is sent to keep NAT alive.

• $REGISTER: A REGISTER message without contact is sent.

• $OPTIONS: An OPTIONS message is sent.

Default setting: $NOTIFY

NAT Keep Alive Dest

Destination receiving the NAT keep alive messages. If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server.

Default setting: $PROXY

SIP ToS/DiffServ Value

TOS/DiffServ field value in UDP IP packets carrying a SIP message.

Default setting: 0x68

SIP CoS Value [0-7]

CoS value for SIP messages. Valid values are 0 through 7.

Default setting: 3

RTP ToS/DiffServ Value

ToS/DiffServ field value in UDP IP packets carrying RTP data.

Default setting: 0xb8

RTP CoS Value [0- 7]

CoS value for RTP data. Valid values are 0 through 7.

Default setting: 6

Network Jitter Level

Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high.

Default setting: high

Jitter Buffer Adjustment

Choose yes to enable or no to disable this feature.

Default setting: yes

SIP Transport

Select the protocol for SIP messages:

• UDP

• TCP

• TLS

• AUTO

The TCP choice provides “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports don't need to be opened or packets dropped for activities such as Internet browsing or e-commerce.

AUTO allows the ATA to select the appropriate protocol automatically, based on the NAPTR records on the DNS server.

SIP Port

Port number of the SIP message listening and transmission port.

Default setting: 5060 for PHONE1 and 5061 for PHONE2

SIP 100REL Enable

To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select No.

Default setting: No

EXT SIP Port

The external SIP port number.

Default setting: blank

Auth Resync-Reboot

If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select Yes. Otherwise, select No.

When the ATA works as a User Agent Server (UAS) and receives NOTIFY request from a User Agent Client (UAC), you can enable the 401 challenge for the NOTIFY request by doing the following:

1. Set the field to Yes.

2. Configure the fields User ID and Password (under the section Subscriber Information from Voice > Line (n)).

   Note	You can replace User ID with Auth ID for the authentication. To achieve this, see Subscriber information for details.

Default setting: Yes

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided.

Default setting: blank

SIP Remote-Party-ID

To use the Remote-Party-ID header instead of the From header, select Yes. Otherwise, select No.

Default setting: Yes

SIP GUID

This feature limits the registration of SIP accounts. The Global Unique ID is generated for each line for each ATA. When it is enabled, the ATA adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset.

Default setting: No

RTP Log Intvl

The interval for the RTP log.

Default setting: 0

Restrict Source IP

If configured, the ATA drops all packets sent to its SIP Ports from an untrusted IP address. A source IP address is untrusted if it doesn't match the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use Outbound Proxy is Yes).

Default setting: No

Referor Bye Delay

The number of seconds to wait before sending a BYE to the referrer to terminate a stale call leg after a call transfer).

Default setting: 4

Refer Target Bye Delay

The number of seconds to wait before sending a BYE to the refer target to terminate a stale call leg after a call transfer.

Default setting: 0

Referee Bye Delay

The number of seconds to wait before sending a BYE to the referee to terminate a stale call leg after a call transfer.

Default setting: 0

Refer-To Target Contact

To contact the refer-to target, select yes. Otherwise, select No.

Default setting: no

Sticky 183

If this feature is enabled, the ATA ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select Yes. Otherwise, select No.

Default setting: No

Auth INVITE

When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.

When the ATA works as a User Agent Server (UAS) and receives INVITE request from a User Agent Client (UAC), you can enable the 401 challenge for the INVITE request by doing the following:

1. Set the field to Yes.

2. Configure the fields User ID and Password (under the section Subscriber Information from Voice > Line (n)).

   Note	You can replace User ID with Auth ID for the authentication. To achieve this, see Subscriber information for details.

Default setting: No

Reply 182 On Call Waiting

When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature, select Yes.

Default setting: No

Use Anonymous With RPID

Determine whether the ATA uses “Anonymous” when Remote Party ID is requested in the SIP message.

Default setting: Yes

Use Local Addr In From

Use the local ATA IP address in the SIP FROM message.

Default setting: No

Broadsoft ALTC

Set whether the SIP is the Broadsoft ALTC.

Default setting: No

Auth Support RFC8760

Determine whether the ATA authorization supports RFC-8760.

• If set to Yes, ATA authorization supports the digest algorithms SHA256, SHA-512/256, and MD5.

  When ATA works as a User Agent Client (UAC), it sends SIP REGISTER or INVITE or SUBSCRIBE requests without authorization header field. SIP server responses 401/407 status code with www-authenticate or proxy-authenticate header field. A SIP server responds with multiple www-authenticate headers. If multiple headers are sent, each must have a different algorithm, with the most preferred one first.

  When the field, Auth Resync-Reboot, and Auth INVITE are set to Yes, ATA will send 401 status code with multiple www-authenticate header fields. The algorithm SHA-256 is top prioritized.

• If set to No, ATA authorization only supports the digest algorithm MD5.

  When ATA works as a User Agent Client (UAC), it sends SIP requests without authorization header field. SIP server responses 401 status code. ATA retries to send request and add an authorization header with MD5 algorithm for server to validate.

Default setting: No

MediaSec Request

Determine whether the ATA initiates media plane security negotiations with the server.

• If set to Yes, ATA supports the client-initiated mode. The ATA phone can initiate media plane security negotiations.

• If set to No, ATA only supports the server-initiated mode. In this case, the server initiates media plane security negotiations. The ATA doesn't initiate negotiations, but can handle negotiation requests from the server to establish secure calls.

To use this parameter, make sure that the following conditions are satisfied:

• Set Secure Call Serv (under the section Supplementary Service Subscription) to Yes.

• Set Secure Call Option (under the section Call Feature Settings) set to Optional.

Default setting: No

MediaSec Over TLS Only

Determine how the ATA initiates or handles the media plane security negotiations. This parameter works with MediaSec Request.

• If set to Yes, ATA initiates or handles media plane security negotiations only when SIP over TLS.

• If set to No, ATA initiates or handles media plane security negotiations regardless of the protocol (UDP/TCP/TLS) for SIP messages.

To use this parameter, make sure the following conditions are satisfied:

• Set Secure Call Serv (under the section Supplementary Service Subscription) to Yes.

• Set Secure Call Option (under the section Call Feature Settings) set to Optional.

Default setting: No

Blind Attn-Xfer Enable

Enables the ATA to perform an attended transfer operation by ending the active call leg and performing a blind transfer of the other call leg. If this feature is disabled, the ATA performs an attended transfer operation by referring the other call leg to the active call leg, while maintaining both call legs. To use this feature, select yes. Otherwise, select no.

Default setting: no

MOH Server

User ID or URL of the auto-answering streaming audio server. When only a user ID is specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the MOH Server is not specified.

Default setting: blank

Xfer When Hangup Conf

Makes the ATA perform a transfer when a conference call has ended. Select yes or no from the drop-down menu.

Default setting: yes

Conference Bridge URL

This feature supports external conference bridging for n-way conference calls (n>2), instead of mixing audio locally. To use this feature, set this parameter to that of the server's name. For example: conf@mysefver.com:12345 or conf (which uses the Proxy value as the domain).

Default setting: blank

Conference Bridge Ports

Select the maximum number of conference call participants. The range is 3 to 10.

Default setting: 3

Enable IP Dialing.

Enable or disable IP dialing. If IP dialing is enabled, you can dial [userid@] a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by entering *, user-id must be numeric and a, b, c, d must be between 0 and 255; the port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled.

Default setting: no

Emergency Number

Comma separated list of emergency number patterns. If outbound call matches one of the patterns, the ATA disables hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters.

Default setting: blank

Mailbox ID

Enter the ID number of the mailbox for this line.

Default setting: blank

Feature Key Sync

Allows the phone to synchronize with the call server. If Do Not Disturb or Call Forwarding settings are changed on the phone, the changes are also made on the server. If changes are made on the server, they are propagated to the phone.

Default setting: no

Secure Call Option

Configures a line to only accept secure calls. Select any of the options:

Optional: Retains the current secure call option for the phone adapter.

Strict: Allows SRTP only when SIP transport is set to TLS and if the ATA receives an unsecure call, the call fails. Allows RTP only when SIP transport is UDP/TCP and if the ATA receives an unsecure call, the call fails.

Default setting: Optional

Company UUID

The Universally Unique Identifier (UUID) assigned to the customer by the emergency call services provider.

For example:

19c8168c-a366-44b5-853c-960fcaa19592

Allowed values: Maximum identifier length is 128 characters.

Default setting: Blank

Primary Request URL

URL of the primary location server that provides the emergency call services.

The location server returns an HELD response to the phone with the requested location URI that is tied to the user phone IP address.

This parameter must be in the form of a valid HTTP or HTTPS URL.

Allowed values: A valid URL not exceeding 255 characters.

Default setting: Blank

Secondary Request URL

URL of the backup server to obtain the user's phone location.

If the primary request URL fails, ATA tries to send the secondary request URL to the emergency call services provider.

This parameter must be in the form of a valid HTTP or HTTPS URL.

Allowed values: A valid URL not exceeding 255 characters.

Default setting: Blank

Proxy

SIP proxy server for all outbound requests.

Default setting: blank

Outbound Proxy

SIP Outbound Proxy Server where all outbound requests are sent as the first hop.

Default setting: blank

Survivability Proxy

Specifies a DNS A record of the Local Survivability Gateway (LSG) nodes. This allows the ATA to perform a failover to a survivability gateway.

String syntax:

hostname[:port][:p=priority][:w=weight][:A=ip-list]
[| hostname2[:port][:p=priority][:w=weight][:A=ip-list]]

ip-list: ip-addr[,ip-addr[,ip-addr…]]

Default: port=0

webex-sgw.example.com:8933:A=10.10.10.10

Where,

• webex-sgw.example.com=Provisioned LSG hostname. It is used for TLS certificate validation when connecting to LSG nodes.

• 8933=LSG port

• 10.10.10.10=Provisioned LSG address

Compared to LSG nodes that have the lowest priority, SSE nodes always have high priority. If there are multiple LSG nodes, try one after the other.

Default setting: blank

Use Outbound Proxy

Enables the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored.

Default setting: no

Use OB Proxy In Dialog

Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the parameter Use Outbound Proxy is no, or the Outbound Proxy parameter is empty.

Default setting: yes

Register

Enable periodic registration with the Proxy parameter. This parameter is ignored if Proxy is not specified.

Default setting: yes

Make Call Without Reg

Allow making outbound calls without successful (dynamic) registration by the unit. If No, dial tone will not play unless registration is successful.

Default setting: no

Register Expires

Expires value in sec in a REGISTER request. The ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec.

Default setting: 3600

Ans Call Without Reg

Allow answering inbound calls without successful (dynamic) registration by the unit.

Default setting: no

Use DNS SRV

Whether to use DNS SRV lookup for Proxy and Outbound Proxy.

Default setting: no

DNS SRV Auto Prefix

If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.

Default setting: no

Proxy Fallback Intvl

After failing over to a lower priority server, the ATA waits for the specified Proxy Fallback Interval, in seconds, before retrying the highest priority proxy (or outbound proxy) servers. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name.

Using multiple DNS A records per server name does not allow the notion of priority, so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a failover.

If the value is 0, the SIP proxy fallback feature is disabled.

Range: 0 –65535 sec

Default setting: 3600

Survivability Proxy Fallback Intvl

After ATA is registered to an LSG node, it waits for the specified interval before it attempts to fallback to a Webex Calling SSE node when any SSE node is reachable.

If there's an active call on the ATA, the call still remains during the fallback process.

Default setting: 30

Proxy Redundancy Method

The method that the ATA uses to create a list of proxies returned in the DNS SRV records. If you select Normal, the list will contain proxiesranked by weight and priority. If you select Based on SRV port, the ATA also inspects the port number based on 1st proxy’s port.

Default setting: Normal

Mailbox Subscribe URL

The URL or IP address of the voicemail server.

Default setting: blank

Mailbox Subscribe Expires

Sets subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server.

Default setting: 2147483647

Auto Register When Failover

Controls the fallback duration.

• no: the fallback happens immediately and automatically. If the Proxy Fallback Intvl is exceeded, all the new SIP messages go to the primary proxy.

• yes: the fallback happens only when current registration expires, which means only a REGISTER message can trigger fallback.

For example, when the value for Register Expires is 3600 seconds and Proxy Fallback Intvl is 600 seconds, the fallback is triggered 3600 seconds later and not 600 seconds later. When the value for Register Expires is 600 seconds and Proxy Fallback Intvl is 1000 seconds, the fallback is triggered at 1200 seconds. After successfully registering back to primary server, all the SIP messages go to primary server.

Default setting: yes

Display Name

Display name for caller ID.

Default setting: blank

User ID

User ID for this line.

Default setting: blank

Password

Password for this line.

Default setting: blank

Use Auth ID

To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password.

Default setting: no

Auth ID

Authentication ID for SIP authentication.

Default setting: blank

Reserved Auth Realm

The value of an authentication realm, which is used in the www-authenticate/proxy-authenticate header field for the INVITE and NOTIFY requests.

Default setting: Blank. The proxy IP address is used as the authentication realm.

Resident Online Number

This setting allows you to associate a "local" telephone number with this line using a valid Skype Online Number from Skype. Calls made to that number will ring your phone. Enter the number without spaces or special characters.

Default setting: blank

SIP URI

The parameter by which the user agent will identify itself for this line. If this field is blank, the actual URI used in the SIP signaling should be automatically formed as: sip:UserName@Domain

Where UserName is the username given for this line in the User ID, and Domain is the domain given for this profile in the User Agent Domain.

If the User Agent Domain is an empty string, then the IP address of the phone should be used for the domain.

If the URI field is not empty, but if a SIP or SIPS URI that contains no @ character, then the actual URI used in the SIP signaling should be automatically formed by appending this parameter with an @ character followed by the IP address of the device.

Call Waiting Serv

Enable Call Waiting Service.

Default setting: yes

Block CID Serv

Enable Block Caller ID Service.

Default setting: yes

Block ANC Serv

Enable Block Anonymous Calls Service

Default setting: yes

Dist Ring Serv

Enable Distinctive Ringing Service

Default setting: yes

Cfwd All Serv

Enable Call Forward All Service

Default setting: yes

Cfwd Busy Serv

Enable Call Forward Busy Service

Default setting: yes

Cfwd No Ans Serv

Enable Call Forward No Answer Service

Default setting: yes

Cfwd Sel Serv

Enable Call Forward Selective Service. Configure this service in the Selective Call Forward Settings section.

Default setting: yes

Cfwd Last Serv

Enable Forward Last Call Service

Default setting: yes

Block Last Serv

Enable Block Last Call Service

Default setting: yes

Accept Last Serv

Enable Accept Last Call Service

Default setting: yes

DND Serv

Enable Do Not Disturb Service

Default setting: yes

CID–Serv

Enable Caller ID Service

Default setting: yes

CWCID Serv

Enable Call Waiting Caller ID Service

Default setting: yes

Call Return Serv

Enable Call Return Service

Default setting: yes

Call Redial Serv

Enable Call Redial Service.

Default setting: yes

Call Back Serv

Enable Call Back Service.

Default setting: yes

Three Way Call Serv

Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer.

Default setting: yes

Three Way Conf Serv

Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer.

Default setting: yes

Attn Transfer Serv

Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer.

Default setting: yes

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer Service.

Default setting: yes

MWI Serv

Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment.

Default setting: yes

VMWI Serv

Enable VMWI Service (FSK)

Default setting: yes

Speed Dial Serv

Enable Speed Dial Service.

Default setting: yes

Secure Call Serv

Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call.

Default setting: yes

Star codes are set in Vertical Service Activation Codes. To enable secure calling by default, without requiring a star code, set the user’s Secure Call Setting to yes. See User 1 and User 2.

Referral Serv

Enable Referral Service. See the Referral Services Codes parameter in Vertical Service Activation Codes for more information.

Default setting: yes

Feature Dial Serv

Enable Feature Dial Service. See the Feature Dial Services Codes parameter in Vertical Service Activation Codes for more information.

Default setting: yes

Service Announcement Serv

Enable Service Announcement Service.

Default setting: no

Reuse CID Number As Name

Use the Caller ID number as the caller name.

Default settings: yes

CONFCID Serv

Enable Caller ID during conference call.

Default settings: yes

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

Int’l Operator

[2-9]xxxxxx

US local number

1xxx[2-9]xxxxxx

US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else

Idle Polarity

Polarity before a call is connected: Forward or Reverse.

Default setting: Forward

Caller Conn Polarity

Polarity after an outbound call is connected: Forward or Reverse.

Default setting: Forward.

Callee Conn Polarity

Polarity after an inbound call is connected: Forward or Reverse.

Default setting: Forward

Cfwd All Dest

Forward number for Call Forward All Service.

Default setting: blank

Cfwd Busy Dest

Forward number for Call Forward Busy Service. Same as Cfwd All Dest.

Default setting: blank

Cfwd No Ans Dest

Forward number for Call Forward No Answer Service. Same as Cfwd All Dest.

Default setting: blank

Cfwd No Ans Delay

Delay in sec before Call Forward No Answer triggers.

Default setting: 20

Cfwd Sel1-8 Caller

Caller number pattern to trigger Call Forward Selective service. When the caller’s phone number matches the entry, the call is forwarded to the corresponding Cfwd Selective Destination (Cfwd Sel1-8 Dest).

• • Use ? to match any single digit.

• • Use * to match any number of digits.

Example: 1408*, 1512???1234

In the above example, a call is forwarded to the corresponding destination if the caller ID either starts with 1408 or is an 11-digit numbering starting with 1512 and ending with 1234.

Default setting: blank

Cfwd Sel1-8 Dest

The destination for the corresponding Call Forward Selective caller pattern (Cfwd Sel1-8 Caller).

Default setting: blank

Cfwd Last Caller

The number of the last caller; this caller is actively forwarded to the Cfwd Last Dest via the Call Forward Last service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Cfwd Last Dest

The destination for the Cfwd Last Caller.

Block Last Caller

The number of the last caller; this caller is blocked via the Block Last Caller Service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Accept Last Caller

The number of the last caller; this caller is accepted via the Accept Last Caller Service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Speed Dial 2-9

Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9.

Default setting: blank

CW Setting

Call Waiting on/off for all calls.

Default setting: yes

Block CID

Setting Block Caller ID on/off for all calls.

Default setting: no

Block ANC

Setting Block Anonymous Calls on or off.

Default setting: no

DND

Setting DND on or off.

Default setting: no

CID Setting

Caller ID Generation on or off.

Default setting: yes

CWCID Setting

Call Waiting Caller ID Generation on or off.

Default setting: yes

Dist Ring

Setting Distinctive Ring on or off.

Default setting: yes

Secure Call Setting

If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first.

Default setting: no

• If Secure Call Setting is set to yes, all outbound calls are secure. However, a user can disable security for a call by dialing *19 before dialing the target number.

• If Secure Call Setting is set to No, the user can make a secure outbound call by dialing *18 before dialing the target number.

• A user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.

Message Waiting

Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle.

Default setting: no

Accept Media Loopback Request

Controls how to handle incoming requests for loopback operation.

• never—Never accepts loopback calls; replies 486 to the caller.

• automatic—Automatically accepts the call without ringing.

• manual—Rings the phone first, and the call must be picked up manually before loopback starts.

Default setting: Automatic

Media Loopback Mode

The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror.

Default setting: source

Media Loopback Type

The loopback type to use when making call to request media loopback operation. Choices are Media and Packet.

Default setting: media

Note that if the ATA answers the call, then the loopback type is determined by the caller (the ATA always picks the first loopback type in the offer if it contains multiple type)

CONFCID Setting

Enables or disables the CONFCID.

Default setting: yes

Ring1 - 8 Caller

Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, or 8. Caller number patterns are matched from Ring 1 to Ring 8. The first match (not the closest match) will be used for alerting the subscriber. The distinctive rings are set on the Regional page. See Regional.

Default setting: blank

Default Ring

Default ringing pattern, 1–8, for all callers.

Default setting: 1

Default CWT

Default CWT pattern, 1–8, for all callers.

Default setting: 1

Hold Reminder Ring

Ring pattern for reminder of a holding call when the phone is on-hook.

Default setting: 8

Call Back Ring

Ring pattern for call back notification.

Default setting: 7

Cfwd Ring Splash Len

Duration of ring splash when a call is forwarded (0 – 10.0s)

Default setting: 0

Cblk Ring Splash Len

Duration of ring splash when a call is blocked (0 – 10.0s)

Default setting: 0

VMWI Ring Policy

The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the ATA indicating the status of the subscriber’s mail box. Three settings are available.

Default setting: New VM Available

• New VM Available—Ring as long as there new voicemail messages.

• New VM Becomes Available—Ring at the point when the first new voicemail message is received.

• New VM Arrives—Ring when the number of new voicemail messages increases.

VMWI Ring Splash Len

Duration of ring splash when new messages arrive before the VMWI signal is applied (0 – 10.0s)

Default setting: 0

Ring On No New VM

If enabled, the ATA plays a ring splash when the voicemail server sends SIP NOTIFY message to the ATA indicating that there are no more unread voice mails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp.

Default setting: no