AudioCodes mediapack MP-500 User Manual page 330

Parameter
[EnableRport]
EMS: X Channel Header
[XChannelHeader]
Web/EMS: Progress Indicator to IP
[ProgressIndicator2IP]
[NumberOfActiveDialogs]
SIP User's Manual
Enables / disables the usage of the 'rport' parameter in the
Via header.
[0] = Enabled.
[1] = Disabled (default).
The device adds an 'rport' parameter to the Via header of
each outgoing SIP message. The first Proxy that receives
this message sets the 'rport' value of the response to the
actual port from which the request was received. This
method is used, for example, to enable the device to identify
its port mapping outside a NAT.
If the Via doesn't include 'rport' tag, the destination port of
the response is taken from the host part of the Via header.
If the Via includes 'rport' tag without a port value, the
destination port of the response is the source port of the
incoming request.
If the Via includes 'rport' tag with a port value (rport=1001),
the destination port of the response is the port indicated in
the 'rport' tag.
Determines whether the x-channel SIP header is added to
SIP messages for providing information on the physical
channel on which the call is received or placed.
[0] Disable = SIP x-channel header is not used (default).
[1] Enable = SIP 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
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 180
Ringing SIP 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
EnableEarlyMedia is set to 1, the device sends 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,
enabling the originating party to listen to network Call
Progress Tones (such as ringback tone or other network
announcements).
Defines the maximum number of active SIP dialogs that are
not call related (i.e., REGISTER and SUBSCRIBE). This
parameter is used to control the Registration / Subscription
rate.
The valid range is 1 to 5. The default value is 5.
330
MP-500 MSBG
Description
Document #: LTRT-12801