AudioCodes E-SBC - Page 1227

CHAPTER68 Configuration Parameters Reference
Mediant 1000 Gateway & E-SBC | User's Manual
Parameter Description
[DefaultReleaseCause] The default release cause is NO_ROUTE_TO_DESTINATION
(3). Other common values include NO_CIRCUIT_AVAILABLE
(34), DESTINATION_OUT_OF_ORDER (27), etc.
Note:
■ 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 more information on mapping PSTN release
causes to SIP responses, see Mapping PSTN Release
Cause to SIP Response.
■ Digital: 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.
■ For a list of SIP responses-Q.931 release cause mapping,
see Alternative Routing to Trunk upon Q.931 Call Release
Cause Code.
'Enable Microsoft
Extension'
configure voip >
sip-definition
settings >
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;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 Manipulation tables to leave only the last 3
digits (for example) for sending to a PBX.
configure voip >
sip-definition
settings > sip-uri-
for-diversion-
header
[UseSIPURIForDiversion
Header]
Defines the URI format in the SIP Diversion header.
■ [0] = 'tel:' (default)
■ [1] = 'sip:'
configure voip >
sip-definition
settings > 100-to-
18x-timeout
Defines the timeout (in msec) between receiving a 100 Trying
response and a subsequent 18x response. If a 18x response is
not received within this timeout period, the call is disconnected.
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