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Page 131 / 186 Scroll up to view Page 126 - 130
5-47
Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
Line Tab
Note
References in this section to the SPA3102 also apply to the AG310.
FXS Port Polarity Configuration
Dial Plan
Dial plan script for this line.
The default(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
The dial plan syntax is expanded in the SPA3102 to allow the designation of three
parameters to be used with a specific gateway:
•
uid – the authentication user-id
•
pwd – the authentication password
•
nat – if this parameter is present, use NAT mapping
Each parameter is separated by a semi-colon (;).
Furthermore, it recognizes gw0, gw1, …, gw4 as the locally configured gateways, where
gw0 represents the local PSTN gateway in the same SPA3102 unit.
Example 1:
*1xxxxxxxxxx<:@fwdnat.pulver.com:5082;uid=jsmith;pwd=xyz
Example 2:
*1xxxxxxxxxx<:@fwd.pulver.com;nat;uid=jsmith;pwd=xyz
Example 3:
[39]11<:@gw0>
PSTN Fallback Dial Plan
<<help here>>
Enable IP Dialing
Enable or disable IP dialing.
The default is no.
Emergency Number
Comma separated list of emergency number patterns. If outbound call matches one of the
pattern, SPA will disable hook flash event handling.
The condition is restored to normal
after the phone is on-hook. Blank signifies no emergency number. Maximum number
length is 63 characters.
The default is blank.
Field
Description
Idle Polarity
Polarity before a call is connected: Forward or Reverse.
The default is Forward.
Caller Conn Polarity
Polarity after an outbound call is connected: Forward or Reverse.
The default is Forward.
Callee Conn Polarity
Polarity after an inbound call is connected: Forward or Reverse.
The default is Forward.
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
Line Tab
Page 133 / 186
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
PSTN Line Tab
This section describes the fields for the following headings on the PSTN Line tab on the SPA3102 and
AG310:
•
Line Enable, page 5-33
•
NAT Settings, page 5-49
•
Network Settings, page 5-50
•
SIP Settings, page 5-50
•
Proxy and Registration (SPA3102/AG310), page 5-53
•
Subscriber Information (SPA3102/AG310), page 5-54
•
Audio Configuration (SPA3102/AG310), page 5-55
•
Dial Plans, page 5-57
•
VoIP-To-PSTN Gateway Setup, page 5-57
•
VoIP Users and Passwords (HTTP Authentication), page 5-59
•
FXO (PSTN) Timer Values (sec), page 5-60
•
PSTN Disconnect Detection, page 5-62
•
International Control (Settings), page 5-63
Line Enable
NAT Settings
Field
Description
Line Enable
To enable this line for service, select yes. Otherwise, select no.
The default is yes.
PSTN Contact List
Select the appropriate list:
None
,
Phone 1+2
,
Phone 1
, or
Phone 2
. The default is
Phone
1+2
.
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.
Page 134 / 186
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
Network Settings
SIP Settings
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.
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.
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.
The default is 6.
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 IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
PSTN 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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