SIP User's Manual 396 Document #: LTRT-65415
MediaPack Series
Parameter Description
Web/EMS: Progress
Indicator to IP
[ProgressIndicator2IP]
For Analog (FXS/FXO) interfaces:
[-1] Not Configured (default) = Default values are used. The default
for FXO interfaces is 1; The default for FXS interfaces is 0.
[0] No PI = For IP-to-Tel calls, the device sends a 180 Ringing
response to IP after placing a call to a phone (FXS) or PBX (FXO).
[1] PI = 1, [8] PI = 8: For IP-to-Tel calls, if the parameter
EnableEarlyMedia is set to 1, the device sends a 183 Session
Progress message with SDP immediately after a call is placed to a
phone/PBX. This is used to cut-through the voice path before the
remote party answers the call. This allows the originating party to
listen to network Call Progress Tones (such as ringback tone or
other network announcements).
Note: This parameter can also be configured per IP Profile (using the
IPProfile parameter) and Tel Profile (using the TelProfile parameter).
[EnableRekeyAfter181]
Enables the device to send a Re-INVITE with a new (different) SRTP
key (in the SDP) upon receipt of a SIP 181 response ("call is being
forwarded").
[0] = Disable (default)
[1] = Enable
Note: This parameter is applicable only if SRTP is used.
[NumberOfActiveDialogs
]
Defines the maximum number of active SIP dialogs that are not call
related (i.e., REGISTER and SUBSCRIBE). This parameter is used to
control the Registration/Subscription rate.
The valid range is 1 to 5. The default value is 5.
Web/EMS: Default
Release Cause
[DefaultReleaseCause]
Defines the default Release Cause (sent to IP) for IP-to-Tel calls when
the device initiates a call release and an explicit matching cause for this
release is not found.
The default release cause is NO_ROUTE_TO_DESTINATION (3).
Other common values include NO_CIRCUIT_AVAILABLE (34),
DESTINATION_OUT_OF_ORDER (27), etc.
Notes:
The default release cause is described in the Q.931 notation and is
translated to corresponding SIP 40x or 50x values (e.g., 3 to SIP
404, and 34 to SIP 503).
For an explanation on mapping PSTN release causes to SIP
responses, see Mapping PSTN Release Cause to SIP Response on
page 278.
For a list of SIP responses-Q.931 release cause mapping, see
Release Reason Mapping.
Web: Enable Microsoft
Extension
[EnableMicrosofExt]
Modifies the called number for numbers received with Microsoft's
proprietary "ext=xxx" parameter in the SIP INVITE URI user part.
Microsoft Office Communications Server sometimes uses this
proprietary parameter to indicate the extension number of the called
party.
[0] Disable (default).
[1] Enable.
For example, if a calling party makes a call to telephone number
622125519100 Ext. 104, the device receives the SIP INVITE (from
Microsoft's application) with the URI user part as INVITE
sip:622125519100;ext=104@10.1.1.10 (or INVITE
tel:622125519100;ext=104). If the parameter EnableMicrosofExt is
enabled, the device modifies the called number by adding an "e" as the
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