User's Manual
54. Configuration Parameters Reference
Version 6.8
769
Mediant 500 E-SBC
Parameter
Description
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.
Web: Enable RFC 4117
Transcoding
CLI: rfc4117-trnsc-enbl
[EnableRFC4117Transcodin
g]
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.
Network Node ID
net-node-id
[NetworkNodeId]
Defines the Network Node Identifier of the device for Avaya UCID.
The valid value range is1 to 0x7FFF. The default is 0.
Notes:
To use this feature, you must set the parameter to any value other
than 0.
To enable the generation by the device of the Avaya UCID value
and adding it to the outgoing INVITE sent to the IP Group (Avaya
entity), use the IP Group table's parameter 'UUI Format'.
Web/EMS: Default Release
Cause
CLI: dflt-release-cse
[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).
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
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
Web: Enable Microsoft
Extension
CLI: microsoft-ext
[EnableMicrosoftExt]
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;[email protected] (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.,
e
622125519100
104
). Once modified, the
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