AudioCodes Mediant 3000 User Manual page 609

Voip media gateway
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SIP User's Manual
Parameter
Web: SIP Destination Port
EMS: Destination Port
[SIPDestinationPort]
Web: Use user=phone in SIP
URL
EMS: Is User Phone
[IsUserPhone]
Web: Use user=phone in
From Header
EMS: Is User Phone In From
[IsUserPhoneInFrom]
Web: Use Tel URI for
Asserted Identity
[UseTelURIForAssertedID]
Web: Tel to IP No Answer
Timeout
EMS: IP Alert Timeout
[IPAlertTimeout]
Web: Enable Remote Party
ID
EMS: Enable RPI Header
[EnableRPIheader]
Web: Enable History-Info
Header
EMS: Enable History Info
[EnableHistoryInfo]
Version 6.4
msec.
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 = 'user=phone' string is part of the SIP URI and SIP To
header (default).
Determines whether the 'user=phone' string is added to the From and
Contact SIP headers.
[0] No = Doesn't add 'user=phone' string (default).
[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 = 'sip:' (default)
[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. If the timer expires, the call is released.
The valid range is 0 to 3600. The default value 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
reason). If found, it is passed to ISDN according to the following
table:
SIP Reason Code
302 - Moved Temporarily
408 - Request Timeout
609
32. Viewing Syslog Messages
Description
ISDN Redirecting Reason
Call Forward Universal (CFU)
Call Forward No Answer (CFNA)
November 2011

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