AudioCodes MP-124 User Manual page 256

Parameter
Web/EMS: Enable TCP Connection
Reuse
[EnableTCPConnectionReuse]
Web/EMS: TCP Timeout
[SIPTCPTimeout]
SIP 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]
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]
SIP User's Manual
Description
Enables the reuse of the same TCP connection for all calls
to the same destination.
[0] Disable = Use a separate TCP connection for each
call.
[1] Enable = Use the same TCP connection for all calls
(default).
Defines the Timer B (INVITE transaction timeout timer) and
Timer F (non-INVITE transaction timeout timer), as defined
in RFC 3261, when the SIP Transport Type is TCP.
The valid range is 0 to 40 sec. The default value is
64*SIPT1Rtx msec.
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-ID (RPI) headers for calling and
called numbers for Tel-to-IP calls.
[0] Disable (default).
[1] Enable = RPI headers are generated in SIP INVITE
messages for both called and calling numbers.
256
MP-11x & MP-124
Document #: LTRT-65412