Linksys SPA9000 Administration Manual page 163

Voice system ip telephony system pstn voip gateway with integrated voice mail server ip phones
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SIP Reg User Agent Name
SIP Accept Language
DTMF Relay MIME Type
Hook Flash MIME Type
Remove Last Reg
Use Compact Header
Escape Display Name
RFC 2543 Call Hold
Mark All AVT Packets
SIP TCP Port Min
SIP TCP Port Max
Voice tab > SIP page
SIP Timer Values (sec) section
SIP T1
Linksys Voice System Administration Guide
User-Agent name to be used in a REGISTER request. If this value is not
specified, the <SIP User Agent Name> is also used for the REGISTER
request.
Default: blank
Accept-Language header used. There is no default (this indicates
SPA9000 does not include this header). If empty, the header is not
included.
MIME Type used in a SIP INFO message to signal a DTMF event.
Default: application/dtmf-relay
MIME Type used in a SIP INFO message to signal a hook flash event.
Default: application/hook-flash
Lets you remove the last registration before registering a new one if the
value is different. Select yes or no from the drop-down menu.
Default: no
Lets you use compact SIP headers in outbound SIP messages. Select yes
or no from the drop-down menu. If set to yes, the SPA9000 uses
compact SIP headers in outbound SIP messages. If set to no, the
SPA9000 uses normal SIP headers. If inbound SIP requests contain
compact headers, SPA9000 reuses the same compact headers when
generating the response regardless the settings of the <Use Compact
Header> parameter. If inbound SIP requests contain normal headers,
SPA9000 substitutes those headers with compact headers (if defined by
RFC 261) if <Use Compact Header> parameter is set to yes.
Default: no
Select yes if you want the SPA9000 to enclose the string (configured in
the Display Name) in a pair of double quotes for outbound SIP
messages. Any occurrences of or \ in the string is escaped with \ and \\
inside the pair of double quotes. Otherwise, select no.
Default: no
If set to yes, unit will include c=0.0.0.0 syntax in SDP when sending a SIP
re-INVITE to the peer to hold the call. If set to no, unit will not include the
c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly
syntax in the SDP in either case.
Default: yes
If set to yes, all AVT tone packets (encoded for redundancy) have the
marker bit set. If set to no, only the first packet has the marker bit set for
each DTMF event.
Default: yes
The lowest TCP port number that can be used for SIP sessions.
Default: 5060
The highest TCP port number that can be used for SIP sessions.
Default: 5080
RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds.
Default: .5
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