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Page 141 / 186 Scroll up to view Page 136 - 140
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
Note
References in this section to the SPA3102 also apply to the AG310.
Dial Plans
VoIP-To-PSTN Gateway Setup
Release Unused Codec
This feature allows the release of codecs not used after codec negotiation on the first call,
so that other codecs can be used for the second line. To use this feature, select yes.
Otherwise, select no.
The default is
yes.
FAX Enable T38
To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no.
The default is
yes.
FAX Tone Detect Mode
This parameter has three possible values:
•
caller or callee - SPA will detect FAX tone whether it is callee or caller
•
caller only - SPA will detect FAX tone only if it is the caller
•
callee only - SPA will detect FAX tone only if it is the callee
The default is caller or callee.
Symmetric RTP
(SPA3102 only) Enable symmetric RTP operation. If enabled, the SPA3102 sends RTP
packets to the source address and port of the last received valid inbound RTP packet. If
disabled (or before the first RTP packet arrives) the SPA3102 sends RTP to the destination
as indicated in the inbound SDP.
The default is yes.
Field
Description
Dial Plan 1/2/3/4/5/6/7/8
Dial plan script for this line.
The default is (xx.) Dial plans in the dial plan pool to be associated with a VoIP Caller or
a PSTN Caller. Each dial plan in the pool is referenced by a index 1 to 8 corresponding to
Dial Plan 1 to 8. The dial plan syntax is the same as that used for Line 1.
Field
Description
VoIP-To-PSTN Gateway Enable
Enable or disable VoIP-To-PSTN Gateway functionality.
The default is yes.
VoIP Caller Authentication
Method
Method to be used to authenticate a VoIP Caller to access the PSTN gateway. Choose from
{none, PIN, HTTP Digest.
The default is none.
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Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
VoIP PIN Max Retry
Number of trials to allow VoIP caller to enter a PIN number (used only if authentication
method is set to PIN).
The default is 3.
One Stage Dialing
Enable one-stage dialing (applicable if authentication method is none, or HTTP Digest,
or caller is in the Access List).
The default is yes.
Line 1 VoIP Caller DP
Index of the dial plan in the dial plan pool to be used when the VoIP Caller is calling from
Line 1 of the same SPA3102 unit during normal operation (in other words, not due to
fallback to PSTN service when Line 1 VoIP service is down). Choose from {none, 1, 2, 3,
4, 5, 6, 7, 8}
Note
Authentication is skipped for Line 1 VoIP caller.
The default is 1.
Default VoIP Caller DP
Index of the dial plan in the dial plan pool to be used when the VoIP Caller is not
authenticated. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}.
The default is 1.
Field
Description
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Linksys IP Phone Administrator Guide
Document Version 3.2
Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
VoIP Users and Passwords (HTTP Authentication)
Line 1 Fallback DP
Index of the dial plan in the dial plan pool to be used when the VoIP Caller is calling from
Line 1 of the same SPA3102 unit due to fallback to PSTN service when Line 1 VoIP
service is down. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}.
The default is 1.
VoIP Caller ID Pattern
A comma separated list of caller number templates such that callers with numbers not
matching any of these templates will be rejected for PSTN gateway service regardless of
the setting of the authentication method. The comparison is applied before access list is
applied. If this parameter is blank (not specified), all callers will be considered for PSTN
gateway service.
For example: 1408*, 1512???1234.
Note: ‘?’ matches any single digit; ‘*’ matches any number of digits.
The default is blank.
VoIP Access List
A comma separated list of IP address templates, such that callers with source IP address
matching any of the templates will be accepted for PSTN gateway service without further
authentication. For example: 192.168.*.*, 66.43.12.1??.
The default is blank.
VoIP Caller 1/2/3/4/5/6/7/8 PIN
One of 8 PIN numbers that can be specified to control access to the PSTN gateway by a
VoIP Caller, when the <VoIP Caller Authentication Method> is set to PIN.
The default is blank.
VoIP Caller 1/2/3/4/5/6/7/8 DP
Index of the dial plan in the dial plan pool to be associated with the VoIP caller who enters
the PIN that matches <VoIP Caller 1/2/3/4/5/6/7/8 PIN>.
The default is 1.
Field
Description
Field
Description
VoIP User 1/2/3/4/5/6/7/8 Auth
ID
The first of 8 user-id’s that a VoIP Caller can use to authenticate itself to the SPA using
the HTTP Digest method (in other words, by embedding an Authorization header in the
SIP INVITE message sent to the SPA. If the credentials are missing or incorrect, the SPA
will challenge the caller with a 401 response). The VoIP caller whose authentication
user-id equals to this ID is referred to VoIP User 1 of this SPA.
Note: If the caller specifies an authentication user-id that does not match any of the VoIP
User Auth ID’s, the INVITE will be rejected with a 403 response.
The default is blank.
VoIP User 1/2/3/4/5/6/7/8 DP
Index of the dial plan in the dial plan pool to be used with VoIP User 1.
The default is 1.
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Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
Ring Settings
FXO (PSTN) Timer Values (sec)
VoIP User 1/2/3/4/5/6/7/8
Password
The password to be used with VoIP User 1. The user assumes the identity of VoIP User 1
must therefore compute the credentials using this password, or the INVITE will be
challenged with a 401 response
The default is blank.
Field
Description
Field
Description
VoIP User 1/2/3/4/5/6/7/8 Auth
ID
The first of 8 user-id’s that a VoIP Caller can use to authenticate itself to the SPA using
the HTTP Digest method (in other words, by embedding an Authorization header in the
SIP INVITE message sent to the SPA. If the credentials are missing or incorrect, the SPA
will challenge the caller with a 401 response). The VoIP caller whose authentication
user-id equals to this ID is referred to VoIP User 1 of this SPA.
Note: If the caller specifies an authentication user-id that does not match any of the VoIP
User Auth ID’s, the INVITE will be rejected with a 403 response.
The default is blank.
VoIP User 1/2/3/4/5/6/7/8 DP
Index of the dial plan in the dial plan pool to be used with VoIP User 1.
The default is 1.
VoIP User 1/2/3/4/5/6/7/8
Password
The password to be used with VoIP User 1. The user assumes the identity of VoIP User 1
must therefore compute the credentials using this password, or the INVITE will be
challenged with a 401 response
The default is blank.
Field
Description
Default Ring
1-8, Follow Line Cfg
Field
Description
VoIP Answer Delay
Delay in seconds before auto-answering inbound VoIP calls for the FXO account. The
range is 0-255.
The default is 3.
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Chapter 5
Linksys ATA Field Reference
PSTN Line Tab
PSTN Answer Delay
Delay in seconds before auto-answering inbound PSTN calls after the PSTN starts
ringing. The range is 0-255.
The default is 16.
VoIP PIN Digit Timeout
Timeout to wait for the 1
st
or subsequent PIN digits from a VoIP caller. The range is 0-255.
The default is 10.
PSTN PIN Digit Timeout
Timeout to wait for the 1
st
or subsequent PIN digits from a PSTN caller. The range is
0-255.
The default is 10.
VoIP DLG Refresh Intvl
Interval between (SIP) Dialog refresh messages sent by the SPA to detect if the VoIP
call-leg is still up. If value is set to 0, SPA will not send refresh messages and VoIP call-leg
status is not checked by the SPA. The refresh message is a SIP ReINVITE and the VoIP
peer must response with a 2xx response. If VoIP peer does not reply or response is not
greater than 2xx, the SPA will disconnect both PSTN and VoIP call legs automatically.
The range is 0-255.
The default is 30.
PSTN Ring Thru Delay
Delay in seconds before starting to ring thru Line 1 after the PSTN starts ringing. In order
for Line 1 to have the caller-id information, the delay should be set to larger than the delay
required to complete the PSTN caller-id delivery (such as 5s). The range is 0-255.
The default is 5.
PSTN-To-VoIP Call Max Dur
Limit on the duration of a PSTN-To-VoIP Gateway Call. Unit is in seconds. 0 means
unlimited. The range is 0-2147483647.
The default is 0.
VoIP-To-PSTN Call Max Dur
Limit on the duration of a VoIP-To-PSTN Gateway Call. Unit is in seconds. 0 means
unlimited. The range is 0-2147483647.
The default is 0.
PSTN Dialing Delay
Delay after hook before the SPA dials a PSTN number. The range is 0-255.
The default is 1.
PSTN Ring Timeout
Delay after a ring burst before the SPA decides that PSTN ring has ceased. The range is
0-255.
The default is 5.
PSTN Dial Digit Len
Determines the on/off time when transmitting digits through the FXO port. The syntax is
on-time
/
off-time
, where
on-time
and
off-time
are expressed in seconds with up to two
decimal places, within the permitted range, which is from .05 to 3.00.
The default is .1/.1. If this value is blank, the default is used.
Field
Description

Rate

4 / 5 based on 1 vote.

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