EasyManua.ls Logo

AudioCodes MediaPack MP-118 - Page 486

AudioCodes MediaPack MP-118
584 pages
To Next Page IconTo Next Page
To Next Page IconTo Next Page
To Previous Page IconTo Previous Page
To Previous Page IconTo Previous Page
Loading...
User's Manual 486 Document #: LTRT-65422
MP-11x & MP-124
Parameter Description
[XChannelHeader]
the call is received or placed.
[0] Disable = (Default) X-Channel header is not used.
[1] Enable = X-Channel header is generated by the device and sent
in INVITE messages and 180, 183, and 200 OK SIP responses. The
header includes the channel, and the device's IP address.
For example, 'x-channel: DS/DS1-1/8;IP=192.168.13.1', where:
'DS/DS-1' is a constant string
'1' is a constant string
'8' is the channel (port)
'IP=192.168.13.1' is the device's IP address
Web/EMS: Progress
Indicator to IP
[ProgressIndicator2IP]
For Analog (FXS/FXO) interfaces:
[-1] Not Configured = (Default) Default values are used. The default
for FXO interfaces is 1; The default for FXS interfaces is 0.
[0] No PI = For IP-to-Tel calls, the device sends a 180 Ringing
response to IP after placing a call to a phone (FXS) or PBX (FXO).
[1] PI = 1, [8] PI = 8: For IP-to-Tel calls, if the parameter
EnableEarlyMedia is set to 1, the device sends a 183 Session
Progress message with SDP immediately after a call is placed to a
phone/PBX. This is used to cut-through the voice path before the
remote party answers the call. This allows the originating party to
listen to network Call Progress Tones (such as ringback tone or
other network announcements).
Note: This parameter can also be configured per IP Profile (using the
IPProfile parameter) and Tel Profile (using the TelProfile parameter).
[EnableRekeyAfter181]
Enables the device to send a re-INVITE with a new (different) SRTP key
(in the SDP) if a SIP 181 response is received ("call is being
forwarded"). The re-INVITE is sent immediately upon receipt of the 200
OK (when the call is answered).
[0] = Disable (default)
[1] = Enable
Note: This parameter is applicable only if SRTP is used.
[NumberOfActiveDialogs
]
Defines the maximum number of concurrent, outgoing SIP REGISTER
dialogs. This parameter is used to control the registration rate.
The valid range is 1 to 5. The default is 5.
Notes:
Once a 200 OK is received in response to a REGISTER message,
the REGISTER message is not considered in this maximum count
limit.
This parameter applies only to outgoing REGISTER messages (i.e.,
incoming is unlimited).
Web/EMS: Default
Release Cause
[DefaultReleaseCause]
Defines the default Release Cause (sent to IP) for IP-to-Tel calls when
the device initiates a call release and an explicit matching cause for this
release is not found.
The default release cause is NO_ROUTE_TO_DESTINATION (3).
Other common values include NO_CIRCUIT_AVAILABLE (34),
DESTINATION_OUT_OF_ORDER (27), etc.
Notes:
The default release cause is described in the Q.931 notation and is
translated to corresponding SIP 40x or 50x values (e.g., 3 to SIP
404, and 34 to SIP 503).
For information on mapping PSTN release causes to SIP responses,

Table of Contents

Other manuals for AudioCodes MediaPack MP-118

Related product manuals