Polycom SoundPoint IP 200W Provisioning And Configuration Manual page 52

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Provisioning and Configuration Guide SoundPoint IP 200W
Name
VOIP_LOG_AUTO_UP
LOAD
SIP Server 0 - Setting regarding the Server that handles SIP VoIP.
SRV_0_SIP_UA_DATA
_DOMAIN
SRV_0_SIP_UA_DATA
_SERVER_PORT
SRV_0_SIP_UA_DATA
_PROXY_ADDR
SRV_0_SIP_UA_DATA
_PROXY_PORT
SRV_0_SIP_UA_DATA
_REREG_TIME
SRV_0_SIP_UA_DATA
_SERVER_IS_LOCAL
SRV_0_SIP_URI_DOM
AIN_CONFIG
A - 6
Type
Size
Permitted Values
uint8
1
0 = no upload
(default)
1 = upload bootlog
only,
2 = upload
continuously
uint8
32
IP address or URL,
double quoted
default is Null
uint16
1
port number
default is 5060
uint8
32
IP address or URL,
double quoted
default is Null
uint16
1
port number
default is 5060
uint16
1
positive integer,
units in seconds
default is 3600
uint8
1
0 = not local (default)
1 = local
uint8
1
0 = If application
omits <domain>, the
value from the
Domain parameter
will be added
(default)
1 = Application must
specify full URI
2 = Doesn't matter
what application
defines, the
<domain> will be
replaced by Domain
parameter
Interpretation
Configure automatic upload of GenVoip log
files.
0 = No uploading.
1 = Upload boot log only. Boot log is the first
full log buffer.
2 = Upload log every 5K.
The registration server. This parameter is
used after the @ as registration server.
If no proxy address is defined, this also
defines the server SIP REGISTERs, for
example, are sent to.
Listening port of the registration-server. Set
to 0 if you are implementing SRV for failover.
The outbound proxy server or fully qualified
domain name (FQDN). Set to domain name
or IP address. All outgoing SIP signalling is
sent to this server.
Listening port of the outbound proxy. Set to
0 if you are implementing SRV for failover.
Specified the time between re-registration at
registration server.
If set to 1, this server is regarded as being
directly available on the Internet, even
though STUN or other mechanisms have
resolved the system to be behind a
NAT-firewall.
This parameter specifies how a destination
URI of the format SIP:<user>@<domain> for
an outgoing call must be specified with
respect to the <domain> part.

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