AudioCodes Mediant 1000B User Manual page 791

User's Manual
Parameter
[EnableIP2IPApplication]
[IP2IPTranscodingMode]
Web: Enable RFC 4117
Transcoding
CLI: rfc4117-trnsc-enbl
[EnableRFC4117Transcodin
g]
Web/EMS: Default Release
Cause
CLI: dflt-release-cse
[DefaultReleaseCause]
Web: Enable Microsoft
Extension
CLI: microsoft-ext
[EnableMicrosoftExt]
Version 6.8

[1] Enable
Note: For this parameter to take effect, a device reset is required.
Note: This parameter is no longer valid and must not be used.
Defines the voice transcoding mode (media negotiation) between two
user agents for the IP-to-IP application. This parameter must always
be set to 1 when using the IP-to-IP application.

[0] Only if Required = Do not force transcoding. Many of the media
settings (such as gain control) are not implemented on the voice
stream. The device passes packets RTP to RTP packets without
any processing.

[1] Force = (Default) Force transcoding on the outgoing IP leg. The
device interworks the media by implementing DSP transcoding.
Enables transcoding of calls according to RFC 4117.

[0] Disable (default)

[1] Enable
Notes:

For this parameter to take effect, a device reset is required.

For more information on transcoding, see Transcoding using Third-
Party Call Control.
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).

Analog: For information on mapping PSTN release causes to SIP
responses, see Mapping PSTN Release Cause to SIP Response
on page 381.

When the Trunk is disconnected or is not synchronized, the internal
cause is 27. This cause is mapped, by default, to SIP 502.

For mapping SIP-to-Q.931 and Q.931-to-SIP release causes, see
Configuring Release Cause Mapping on page 375.

For a list of SIP responses-Q.931 release cause mapping, see
''Alternative Routing to Trunk upon Q.931 Call Release Cause
Code'' on page 403.
Enables the modification of the called and calling 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 or calling 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
791
51. Configuration Parameters Reference
Description
Mediant 1000B Gateway & SBC