AudioCodes MediaPack 114 Manual do Utilizador descarregar pdf (Página 149)

Version 5.6 149 November 2008

SIP User's Manual 3. Web-Based Management

Parameter Description

of the destination phone number.

For both options, after the phone number is collected, it's sent to the

transferee in a SIP REFER request (without a Replaces header). The

call is then terminated and a confirmation tone is played to the phone.

If the phone number collection fails due to a mismatch, a reorder tone

is played to the phone.

Notes:

 This parameter is applicable to FXO and FXS interfaces, but for

FXO the Web interface does not display this parameter.

 It is possible to configure whether the KeyBlindTransfer code is

added as a prefix to the dialed destination number, by using the

parameter KeyBlindTransferAddPrefix (refer to ''Channel

Parameters'' on page 298).

Call Waiting (Note: The call waiting can be viewed in the 'Call Waiting' table - refer to ''Call Waiting''

on page 361.)

Activate

[KeyCallWaiting]

Keypad sequence that activates the Call Waiting option. After the

sequence is pressed, a confirmation tone is heard.

Deactivate

[KeyCallWaitingDeact]

Keypad sequence that deactivates the Call Waiting option. After the

sequence is pressed, a confirmation tone is heard.

Reject Anonymous Call

(Note: You can reject anonymous calls per device, using RejectAnonymousCallPerPort - refer to

''Analog Telephony Parameters'' on page 279.)

Activate

[KeyRejectAnonymousCal

Keypad sequence that activates the reject anonymous call option,

whereby the device rejects incoming anonymous calls. After the

sequence is pressed, a confirmation tone is heard.

Deactivate

[KeyRejectAnonymousCal

lDeact]

Keypad sequence that de-activates the reject anonymous call option.

After the sequence is pressed, a confirmation tone is heard.

3.4.4.2.6 Stand-Alone Survivability

The 'SAS Configuration' page allows you to configure the device's Stand-Alone Survivability

(SAS) feature. This feature is useful for providing a local backup via the PSTN in Small or

Medium Enterprises (SME) that are serviced by IP Centrex services. In such environments,

the enterprise's incoming and outgoing telephone calls (external and internal) are controlled

by the Proxy, which communicates with the enterprise through the WAN interface. SAS

ensures that incoming, outgoing, and internal calls service is maintained in case of a WAN

or Proxy failure, using a PSTN (or an alternate VoIP) backup connection and the device's

built-in internal routing. To utilize the SAS feature, the VoIP CPEs such as IP phones or

residential gateways need to be defined so that their Proxy and Registrar destination

addresses and UDP port equal the SAS feature's IP address and SAS local SIP UDP port.

Notes:

• For a detailed explanation on SAS and for configuring various SAS

setups, refer to ''Stand-Alone Survivability (SAS) Feature'' on page 315).

• For additional SAS parameters (configurable only using the ini file), refer

to ''SIP Configuration Parameters'' on page 260.

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