AudioCodes Mediant 3000 User Manual page 620

Voip media gateway
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Parameter
Web: Enable RFC 4117
Transcoding
[EnableRFC4117Transcodi
ng]
Web: Enable Single DSP
Transcoding
[EnableSingleDSPTranscod
ing]
Web/EMS: Default Release
Cause
[DefaultReleaseCause]
Web: Enable Microsoft
Extension
[EnableMicrosoftExt]
SIP User's Manual
stream. The device passes packets RTP to RTP packets without
any processing.
[1] Force = Force transcoding on the outgoing IP leg. The device
interworks the media by implementing DSP transcoding. (default)
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 on page 470.
Enables the use of a single DSP for transcoding between G.711 and
LBR coders.
[0] Disable (default)
[1] Enable
Notes:
For this parameter to take effect, a device reset is required.
If the TranscodingMode and IP2IPTranscodingMode parameters
are set to 1 (i.e., Force), then this parameters is ignored.
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).
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 294.
For a list of SIP responses-Q.931 release cause mapping, see
'Release Reason Mapping' on page 261.
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
tel:622125519100;ext=104). If the parameter EnableMicrosofExt is
enabled, the device modifies the called number by adding an "e" as
the prefix, removing the "ext=" parameter, and adding the extension
number as the suffix (e.g., e622125519100104). Once modified, the
device can then manipulate the number further, using the Number
620
Description
Document #: LTRT-89712
Mediant 3000

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