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Page 116 / 186 Scroll up to view Page 111 - 115
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
Regional Tab
Feature Invocation Method
Select the method you want to use, Default or Sweden default. (Not in PAP2T)
The default is Default.
More Echo Suppression
Enable or disable more echo suppresion. The default is no.
GR909 Test
To use this test, select
yes
. Otherwise, keep the default,
no
.
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Chapter 5
Linksys ATA Field Reference
Line Tab
Line Tab
This section describes the fields for the following headings on the
Line tabs:
•
Line Enable, page 5-33
•
Streaming Audio Server (SAS), page 5-33
•
NAT Settings, page 5-34
•
Network Settings, page 5-35
•
SIP Settings, page 5-35
•
Call Feature Settings, page 5-38
•
Proxy and Registration, page 5-38
•
Subscriber Information, page 5-40
•
Supplementary Service Subscription, page 5-40
•
Audio Configuration, page 5-42
•
VoIP Fallback to PSTN (SPA3102/AG310), page 5-46
•
Gateway Accounts (SPA3102/AG310), page 5-45
•
Dial Plan, page 5-46
•
FXS Port Polarity Configuration, page 5-47
In a configuration profile, the Line parameters must be appended with the appropriate numeral (for
example, [1] or [2]) to identify the line to which the setting applies. The number of lines varies with the
model of the ATA. For example, the SPA2102 provides two Line tabs (Line 1 and Line 2), while the
SPA8000 provides eight tabs (Line1 through Line 8).
The SPA2102 provides one User tab for each Line tab (User 1 and User 2), where many of the
line-specific configuration parameters are contained. The SPA8000 does not provide User tabs, but
consolidates all the line-specific parameters on the Line tab.
Line Enable
Streaming Audio Server (SAS)
Field
Description
Line Enable
To enable this line for service, select yes. Otherwise, select no.
The default is yes.
Field
Description
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Chapter 5
Linksys ATA Field Reference
Line Tab
NAT Settings
SAS Enable
To enable the use of the line as a streaming audio source, select yes. Otherwise, select no.
If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming
calls and streams audio RTP packets to the caller.
The default is no.
SAS DLG Refresh Intvl
If this is not zero, it is the interval at which the streaming audio server sends out session
refresh (SIP re-INVITE) messages to determine whether the connection to the caller is
still active. If the caller does not respond to the refresh message, the Linksys IP phone
ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the
session refresh is disabled).
The default is 30.
SAS Inbound RTP Sink
This setting works around devices that do not play inbound RTP if the streaming audio
server line declares itself as a send-only device and tells the client not to stream out audio.
Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this is used
by the Linksys IP phone’s streaming audio server line in the SDP of its 200 response to
an inbound INVITE message from a client.
The purpose of this parameter is to work around devices that do not play inbound RTP if
the SAS line declares itself as a send-only device and tells the client not to stream out
audio. This parameter is a FQDN or IP address of a RTP sink to be used by the SPA SAS
line in the SDP of its 200 response to inbound INVITE from a client. It will appear in the
c = line and the port number and, if specified, in the m = line of the SDP.
If this value is
not specified or equal to 0, then c = 0.0.0.0 and a=sendonly will be used in the SDP to tell
the SAS client to not to send any RTP to this SAS line. If a non-zero value is specified,
then a=sendrecv and the SAS client will stream audio to the given address. Special case:
If the value is $IP, then the SAS line’s own IP address is used in the c = line and
a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line.
The default value is empty.
Field
Description
NAT Mapping Enable
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes.
Otherwise, select no.
The default is no.
NAT Keep Alive Enable
To send the configured NAT keep alive message periodically, select yes. Otherwise, select
no.
The default is no.
NAT Keep Alive Msg
Enter the keep alive message that should be sent periodically to maintain the current
NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is
$REGISTER, a REGISTER message without contact is sent.
The default is $NOTIFY.
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Chapter 5
Linksys ATA Field Reference
Line Tab
Network Settings
SIP Settings
NAT Keep Alive Dest
Destination that should receive NAT keep alive messages. If the value is $PROXY,
the messages are sent to the current or outbound proxy.
The default is $PROXY.
Field
Description
SIP ToS/DiffServ Value
TOS/DiffServ field value in UDP IP packets carrying a SIP message.
The default is 0x68.
SIP CoS Value [0-7]
CoS value for SIP messages. (Not in PAP2T)
The default is 3.
RTP ToS/DiffServ Value
ToS/DiffServ field value in UDP IP packets carrying RTP data.
The default is 0xb8.
RTP CoS Value [0-7]
CoS value for RTP data. (Not in PAP2T)
The default is 6.
Auto PSTN Failback
If enabled, the SPA will automatically route all calls to the PSTN gateway when the Line
1 proxy is down (registration failure or network link down).
The default is yes.
Network Jitter Level
Determines how jitter buffer size is adjusted by the Linksys IP phone. Jitter buffer size is
adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10
milliseconds + current RTP frame size), whichever is larger, for all jitter level settings.
However, the starting jitter buffer size value is larger for higher jitter levels. This setting
controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select
the appropriate setting: low, medium, high, very high, or extremely high.
The default is high.
Jitter Buffer Adjustment
Controls how the jitter buffer should be adjusted. Select the appropriate setting: up and
down, up only, down only, or disable.
The default is up and down.
Field
Description
SIP Port
Port number of the SIP message listening and transmission port.
The default is 5060.
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Linksys ATA Field Reference
Line Tab
SIP 100REL Enable
To enable the support of 100REL SIP extension for reliable transmission of provisional
responses (18x) and use of PRACK requests, select yes. Otherwise, select no.
The default is no.
EXT SIP Port
The external SIP port number.
Auth Resync-Reboot
If this feature is enabled, the Linksys IP phone authenticates the sender when it receives
the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes.
Otherwise, select no.
The default is yes.
SIP Proxy-Require
The SIP proxy can support a specific extension or behavior when it sees this header from
the user agent. If this field is configured and the proxy does not support it, it responds
with the message, unsupported. Enter the appropriate header in the field provided.
SIP Remote-Party-ID
To use the Remote-Party-ID header instead of the From header, select yes. Otherwise,
select no.
The default is yes.
SIP GUID
(Not in PAP2T) The Global Unique ID is generated for each line for each device. When
it is enabled, the Linksys IP phone adds a GUID header in the SIP request. The GUID is
generated the first time the unit boots up and stays with the unit through rebooting and
even factory reset. This feature was requested by Bell Canada (Nortel) to limit the
registration of SIP accounts.
The default is yes.

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