AudioCodes Mediant 1000B - Page 1071

Version 7.2 1071 Mediant 1000B Gateway & E-SBC
User's Manual 65. Configuration Parameters Reference
Parameter Description
TCP connection set up.
Note: If the destination is a Proxy server, the TCP/TLS connection is
persistent regardless of the settings of the parameter.
TCP Timeout
configure voip > sip-
definition settings > tcp-
timeout
[SIPTCPTimeout]
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 is 64 * SipT1Rtx parameter
value. For example, if SipT1Rtx is set to 500 msec, then the default of
SIPTCPTimeout is 32 sec.
SIP Destination Port
configure voip > sip-
definition settings > sip-dst-
port
[SIPDestinationPort]
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.
Use user=phone in SIP URL
configure voip > sip-
definition settings >
user=phone-in-url
[IsUserPhone]
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.
Use user=phone in From
Header
configure voip > sip-
definition settings > phone-
in-from-hdr
[IsUserPhoneInFrom]
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.
Use Tel URI for Asserted
Identity
configure voip > sip-
definition settings > uri-for-
assert-id
[UseTelURIForAssertedID]
Determines the format of the URI in the P-Asserted-Identity and P-
Preferred-Identity headers.
 [0] Disable = (Default) 'sip:'
 [1] Enable = 'tel:'
Tel to IP No Answer
Timeout
configure voip > gateway
advanced > tel2ip-no-ans-
timeout
[IPAlertTimeout]
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.
Enable Remote Party ID
configure voip > sip-
definition settings > remote-
party-id
[EnableRPIheader]
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.
Enable History-Info Header
configure voip > sip-
definition settings > hist-info-
hdr
Enables usage of the SIP History-Info header.
 [0] Disable (default)
 [1] Enable
User Agent Client (UAC) Behavior: