AudioCodes Mediant 500L - Page 1087

Version 7.2 1087 Mediant 500L Gateway & E-SBC
User's Manual 69. Configuration Parameters Reference
Parameter Description
Default Release Cause
configure voip > sip-
definition settings > 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.
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.
 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 555.
 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 532.
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 setting
s >
sip-uri-for-
diversion-header
[UseSIPURIForDiversionHe
ader]
Defines the URI format in the SIP Diversion header.
 [0] = 'tel:' (default)
 [1] = 'sip:'
configure voip > sip-
definition settings >
100-to-18x-timeout
[TimeoutBetween100And18
x]
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.
The valid range is 0 to 180,000 (i.e., 3 minutes). The default is 32000
(i.e., 32 sec).