AudioCodes Mediant 1000B User Manual page 950

Parameter
[SIPTCPTimeout]
SIP Destination Port
sip-dst-port
[SIPDestinationPort]
Use user=phone in SIP
URL
user=phone-in-url
[IsUserPhone]
Use user=phone in From
Header
phone-in-from-hdr
[IsUserPhoneInFrom]
Use Tel URI for Asserted
Identity
uri-for-assert-id
[UseTelURIForAssertedID
]
Tel to IP No Answer
Timeout
tel2ip-no-ans-
timeout
[IPAlertTimeout]
Enable Remote Party ID
remote-party-id
[EnableRPIheader]
Enable History-Info
Header
hist-info-hdr
[EnableHistoryInfo]
User's Manual
the SIP transport type is TCP.
The valid range is 0 to 40 sec. The default is 64 * SipT1Rtx parameter
value. For example, if SipT1Rtx is set to 500 msec, then the default of
SIPTCPTimeout is 32 sec.
Defines the SIP destination port for sending initial SIP requests.
The valid range is 1 to 65534. The default port is 5060.
Note: SIP responses are sent to the port specified in the Via header.
Determines whether the 'user=phone' string is added to the SIP URI
and SIP To header.

[0] No = 'user=phone' string is not added.

[1] Yes = (Default) 'user=phone' string is part of the SIP URI and SIP
To header.
Determines whether the 'user=phone' string is added to the From and
Contact SIP headers.

[0] No = (Default) Doesn't add 'user=phone' string.

[1] Yes = 'user=phone' string is part of the From and Contact
headers.
Determines the format of the URI in the P-Asserted-Identity and P-
Preferred-Identity headers.

[0] Disable = (Default) 'sip:'

[1] Enable = 'tel:'
Defines the time (in seconds) that the device waits for a 200 OK
response from the called party (IP side) after sending an INVITE
message, for Tel-to-IP calls. If the timer expires, the call is released.
The valid range is 0 to 3600. The default is 180.
Enables Remote-Party-Identity headers for calling and called numbers
for Tel-to-IP calls.

[0] Disable (default).

[1] Enable = Remote-Party-Identity headers are generated in SIP
INVITE messages for both called and calling numbers.
Enables usage of the History-Info header.

[0] Disable (default)

[1] Enable
User Agent Client (UAC) Behavior:

Initial request: The History-Info header is equal to the Request-URI.
If a PSTN Redirect number is received, it is added as an additional
History-Info header with an appropriate reason.

Upon receiving the final failure response, the device copies the
History-Info as is, adds the reason of the failure response to the last
entry, and concatenates a new destination to it (if an additional
request is sent). The order of the reasons is as follows:
a. Q.850 Reason
b. SIP Reason
c. SIP Response code

Upon receiving the final response (success or failure), the device
searches for a Redirect reason in the History-Info (i.e., 3xx/4xx SIP
950
Mediant 1000B Gateway and E- SBC
Description
Document #: LTRT-27044