AudioCodes Mediant 500 E-SBC, Руководство пользователя

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User's Manual 54. Configuration Parameters Reference
Version 6.8 761 Mediant 500 E-SBC
Parameter Description
[TGRProutingPrecedence]

[0] = (Default) IP-to-Tel routing is determined by the Inbound IP
Routing table (PSTNPrefix parameter). If a matching rule is not
found in this table, the device uses the Trunk Group parameters for
routing the call.

[1] = The device first places precedence on the 'tgrp' parameter for
IP-to-Tel routing. If the received INVITE Request-URI does not
contain the 'tgrp' parameter or if the Trunk Group number is not
defined, then the Inbound IP Routing table is used for routing the
call.
Below is an example of an INVITE Request-URI with the 'tgrp'
parameter, indicating that the IP call should be routed to Trunk Group
7:
INVITE sip:200;tgrp=7;trunk-
context=example.com@10.33.2.68;user=phone SIP/2.0
Notes:

For enabling routing based on the 'tgrp' parameter, the UseSIPTgrp
parameter must be set to 2.

For IP-to-Tel routing based on the 'dtg' parameter (instead of the
'tgrp' parameter), use the parameter UseBroadsoftDTG.
CLI: use-dtg
[UseBroadsoftDTG] Determines whether the device uses the 'dtg' parameter for routing IP-
to-Tel calls to a specific Trunk Group.

[0] Disable (default)

[1] Enable
When this parameter is enabled, if the Request-URI in the received
SIP INVITE includes the 'dtg' parameter, the device routes the call to
the Trunk Group according to its value. This parameter is used instead
of the 'tgrp/trunk-context' parameters. The 'dtg' parameter appears in
the INVITE Request-URI (and in the To header).
For example, the received SIP message below routes the call to Trunk
Group ID 56:
INVITE sip:123456@192.168.1.2;dtg=56;user=phone SIP/2.0
Note: If the Trunk Group is not found based on the 'dtg' parameter, the
Inbound IP Routing table is used instead for routing the call to the
appropriate Trunk Group.
Web/EMS: Enable GRUU
CLI: enable-gruu
[EnableGRUU]
Determines whether the Globally Routable User Agent URIs (GRUU)
mechanism is used, according to RFC 5627. This is used for obtaining
a GRUU from a registrar and for communicating a GRUU to a peer
within a dialog.

[0] Disable (default)

[1] Enable
A GRUU is a SIP URI that routes to an instance-specific UA and can
be reachable from anywhere. There are a number of contexts in which
it is desirable to have an identifier that addresses a single UA (using
GRUU) rather than the group of UA’s indicated by an Address of
Record (AOR). For example, in call transfer where user A is talking to
user B, and user A wants to transfer the call to user C. User A sends a
REFER to user C:
REFER sip:C@domain.com SIP/2.0
From: sip:A@domain.com;tag=99asd
To: sip:C@domain.com
Refer-To: (URI that identifies B's UA)
The Refer-To header needs to contain a URI that user C can use to