Linksys ATA Voice Field Reference. Linksys WRTP54G, WAG54GP2, SPA3102, PAP2T, WRP400, SPA8000, AG310, SPA2102, RTP300
Below you will find brief information for PAP2T, SPA2102, SPA3102, SPA8000, AG310, RTP300, WRP400, WRTP54G, WAG54GP2. These devices can connect analog telephones to an IP network via a broadband (DSL or cable) modem or router, allowing you to make and receive calls over the internet. The devices support a variety of features, including call waiting, call forwarding, and conference calling. All models support multiple voice codecs, allowing you to make clear calls even with limited bandwidth. Certain models also include built-in routers for additional network connectivity.
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7
Linksys ATA Voice Field Reference
Info Tab
Linksys ATA Voice Field Reference
This chapter describes the Advanced Voice tab and their corresponding fields as found in the
Web Configuration Utility pages. For information about the Voice > Provisioning tab, see the
Linksys SPA Provisioning Guide
.
After you click the Voice tab, you can choose the following tabs:
•
•
”System Tab” section on page 72
•
•
”Regional Tab” section on page 81
•
”Line Tab(s)” section on page 95
•
”PSTN Line Tab (AG310 and SPA3102)” section on page 109
•
”User Tab(s)” section on page 124
•
”PSTN User Tab (AG310 and SPA3102)” section on page 128
NOTE: Not all fields listed may be applicable to your ATA device or your setup.
Info Tab
This section describes the fields for the following headings on the Info tab:
•
”Product Information” section on page 67
•
”System Status” section on page 67
•
”Line Status” section on page 68
•
”System Information (PAP2T)” section on page 69
•
”PSTN Line Status (AG310 and SPA3102)” section on page 70
NOTE: The fields on the Info tab are read-only and cannot be edited.
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Info Tab
Product Information
Field
Product Name
Serial Number
Software Version
Hardware Version
MAC Address
Client Certificate
Customization
Description
Model number/name.
Serial number.
Software version number.
Hardware version number.
MAC address.
Status of the client certificate, which can indicate if the ATA has been authorized by your ITSP.
For a Remote Configuration (RC) unit, this field indicates whether the unit has been customized or not. Pending indicates a new RC unit that is ready for provisioning. If the unit has already retrieved its customized profile, this field displays the name of the company that provisioned the unit.
System Status
Field
Current Time
Elapsed Time
RTP Packets Sent
RTP Bytes Sent
RTP Packets Recv
RTP Bytes Recv
SIP Messages Sent
SIP Bytes Sent
SIP Messages Recv
SIP Bytes Recv
External IP
Description
Current date and time of the system; for example, 10/3/2003 16:43:00.
Total time elapsed since the last reboot of the system; for example, 25 days and
18:12:36.
Total number of RTP packets sent (including redundant packets).
Total number of RTP bytes sent.
Total number of RTP packets received (including redundant packets).
Total number of RTP bytes received.
Total number of SIP messages sent (including retransmissions).
Total number of bytes of SIP messages sent (including retransmissions).
Total number of SIP messages received (including retransmissions).
Total number of bytes of SIP messages received (including retransmissions).
External IP address used for NAT mapping.
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Info Tab
Line Status
Field
(PSTN) Hook State
Registration State
Description
Hook state of the FXO port. Options are either On or Off.
Indicates if the line has registered with the SIP proxy.
Last Registration At
Next Registration In
Message Waiting
Call Back Active
Last Called Number
Last Caller Number
Mapped SIP Port
Call 1 and 2 State
Call 1 and 2 Tone
Call 1 and 2 Encoder
Call 1 and 2 Decoder
Call 1 and 2 FAX
Call 1 and 2 Type
Last date and time the line was registered.
Number of seconds before the next registration renewal.
Indicates whether you have new voicemail waiting. Options are either Yes or No. This is updated when voicemail notification is received. You can also manually modify it to clear or set the flag. Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and survives after reboot or power cycle.
Indicates whether a call back request is in progress. Options are either Yes or No.
The last number called from the FXO Line.
Number of the last caller.
Port number of the SIP port mapped by NAT.
May take one of the following values:
• Idle
• Collecting PSTN Pin
• Invalid PSTN PIN
• PSTN Caller Accepted
• Connected to PSTN
Type of tone used by the call.
Codec used for encoding.
Codec used for decoding.
Status of the fax pass-through mode.
Direction of the call. May take one of the following values:
• PSTN Gateway Call = VoIP-To-PSTN Call
• VoIP Gateway Call = PSTN-To-VoIP Call
• PSTN To Line 1 = PSTN call ring through and answered by Line 1
• Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW
• Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number
• Line 1 To PSTN Gateway
• Line 1 Fallback To PSTN Gateway
Call 1 and 2 Remote Hold Indicates whether the far end has placed the call on hold.
Call 1 and 2 Callback Indicates whether the call was triggered by a call back request.
Call 1 and 2 Peer Name
Call 1 and 2 Peer Phone
Name of the internal phone.
Phone number of the internal phone.
Call 1 and 2 Call Duration Duration of the call.
Call 1 and 2 Packets Sent Number of packets sent.
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Info Tab
Call 1 and 2 Packets Recv Number of packets received.
Call 1 and 2 Bytes Sent Number of bytes sent.
Call 1 and 2 Bytes Recv
Call 1 and 2 Decode
Latency
Call 1 and 2 Jitter
Number of bytes received.
Number of milliseconds for decoder latency.
Call 1 and 2 Round Trip
Delay
Number of milliseconds for receiver jitter.
Number of milliseconds for delay.
Call 1 and 2 Packets Lost Number of packets lost.
Call 1 and 2 Packet Error Number of invalid packets received.
The port mapped for Real Time Protocol traffic for Call 1/2. Call 1 and 2 Mapped RTP
Port
Call 1 and 2 Media
Loopback
Media loopback is used to quantitatively and qualitatively measure the voice quality experienced by the end user.
System Information (PAP2T)
Field
DHCP
Current IP
Host Name
Domain
Current Netmask
Current Gateway
Primary DNS
Secondary DNS
Description
Indicates if DHCP is enabled.
Displays the current IP address assigned to the Linksys ATA device.
Displays the current IP address assigned to the Linksys ATA device.
Displays the network domain name of the Linksys ATA device.
Displays the network mask assigned to the Linksys ATA device.
Displays the default router assigned to the Linksys ATA device.
Displays the primary DNS server assigned to the Linksys ATA device.
Displays the secondary DNS server assigned to the Linksys ATA device.
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Info Tab
PSTN Line Status (AG310 and SPA3102)
Field
(PSTN) Hook State
(PSTN) Line Voltage
Description
Hook state of the FXO port. Either On or Off.
The voltage existing on the PSTN line.
(PSTN) Loop Current
Registration State
Last Registration At
Next Registration In
The current (milliamperes) existing on the local loop.
Indicates if the line has registered with the SIP proxy.
Last date and time the line was registered.
Number of seconds before the next registration renewal.
Last Called VoIP Number The last VoIP number called from the FXO Line.
Last Called PSTN Number The PSTN number dialed by the SPA (logged only if a non-trivial dial plan is used).
Last VoIP Caller
Last PSTN Caller
Last PSTN Disconnect
Reason
The last VoIP caller to the FXO Line.
Name and number of the last PSTN caller.
PSTN Activity Timer
Reason for SPA hanging up the FXO port. Can be one of the following:
• PSTN Disconnect Tone
• PSTN Activity Timeout
• CPC Signal
• Polarity Reversal
• VoIP Call Failed
• VoIP Call Ended
• Invalid VoIP Destination
• Invalid PIN
• PIN Digit Timeout
• VoIP Dialing Timeout
• PSTN Gateway Call Timeout
• VoIP Gateway Call Timeout
Shows the time (ms) before the SPA disconnects the current gateway unless the
PSTN side has some audio activity.
Port number of the SIP port mapped by NAT.
Mapped SIP Port
Call Type May take one of the following values:
• PSTN Gateway Call = VoIP-To-PSTN Call
• VoIP Gateway Call = PSTN-To-VoIP Call
• PSTN To Line 1 = PSTN call ring through and answered by Line 1
• Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW
• Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number
• Line 1 To PSTN Gateway
• Line 1 Fallback To PSTN Gateway
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Info Tab
VoIP State
PSTN State
May take one of the following values:
• Idle
• Collecting PSTN Pin
• Invalid PSTN PIN
• PSTN Caller Accepted
• Connected to PSTN
May take one of the following values:
• Idle
• Collecting PSTN Pin
• Invalid PSTN PIN
• PSTN Caller Accepted
• Connected to PSTN
Indicates what tone is being played to the VoIP call leg.
Indicate what tone is being played to the PSTN call leg.
Name of the party at the VoIP call leg.
Name of the party at the PSTN call leg.
VoIP Tone
PSTN Tone
VoIP Peer Name
PSTN Peer Name
VoIP Peer Number
PSTN Peer Number
VoIP Call Encoder
VoIP Call Decoder
VoIP Call FAX
VoIP Call Remote Hold
VoIP Call Duration
VoIP Call Packets Sent
Phone number of the party at the VoIP call leg.
Phone number of the party at the PSTN call leg.
Audio encoder being used for the VoIP call leg.
Audio decoder being used for the VoIP call leg.
Status of the fax pass-through mode.
Indicates whether the far end has placed the call on hold.
Duration of the call.
Number of packets sent.
VoIP Call Packets Recv
VoIP Call Bytes Sent
Number of packets received.
Number of bytes sent.
VoIP Call Bytes Recv Number of bytes received.
VoIP Call Decode Latency Number of milliseconds for decoder latency.
VoIP Call Jitter Number of milliseconds for receiver jitter.
VoIP Call Round Trip Delay Number of milliseconds for delay.
VoIP Call Packets Lost
VoIP Call Packet Error
VoIP Call Mapped RTP
Port
Number of packets lost.
Number of invalid packets received.
The port mapped for Real Time Protocol traffic for Call 1/2.
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System Tab
System Tab
This section describes the fields for the following headings on the System tab:
•
”System Configuration” section on page 72
•
”Internet Connection Type (PAP2T)” section on page 72
•
”Optional Network Configuration (PAP2T)” section on page 73
•
”Miscellaneous Settings (not used with PAP2T)” section on page 73
System Configuration
Field
Restricted Access
Domains
Enable Web Server
Description
This feature is used when implementing software customization.
Web Server Port
Enable/disable web server of Linksys ATA device
This feature should only be used on firmware version 1.0.9 or later.
The default is yes.
This field is only found in the PAP2T.
Port number of the Linksys ATA device administration web server.
The default is 80.
This field is only found in the PAP2T.
Enable Web Admin Access Lets you enable or disable local access to the Web Configuration Utility. Select yes or no from the drop-down menu.
The default is yes.
Admin Password Password for the administrator. The default is no password.
User Password Password for the user. The default is no password.
Internet Connection Type (PAP2T)
Field
DHCP
Static IP
NetMask
Gateway
Description
Enable or disable DHCP.
The default is yes.
Static IP address of Linksys ATA device, which takes effect if DHCP is disabled.
The default is 0.0.0.0.
The NetMask used by Linksys ATA device when DHCP is disabled.
The default is 255.255.255.0.
The default gateway used by Linksys ATA device when DHCP is disabled.
The default is 0.0.0.0.
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System Tab
Optional Network Configuration (PAP2T)
)
Field
Host Name
Domain
Primary DNS
Secondary DNS
DNS Server Order
DNS Query Mode
Syslog Server
Debug Server
Debug Level
Primary NTP Server
Secondary NTP Server
Description
The host name of the Linksys ATA device.
The network domain of the Linksys ATA device.
DNS server used by Linksys ATA device in addition to DHCP supplied DNS servers if
DHCP is enabled; when DHCP is disabled, this is the primary DNS server.
The default is 0.0.0.0.
Sets the secondary DNS server to take over if problems are discovered with the
Primary DNS server. This is in addition to DHCP-supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the secondary DNS server.
The default is 0.0.0.0.
Specifies the method for selecting the DNS server. The options are Manual (enter the IP address of the DNS server manually; that is do not look at the DHCP-supplied
DNS table), Manual/DHCP, and DHCP/Manual.
Do parallel or sequential DNS Query. With parallel DNS query mode, the Linksys
ATA device sends the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply is accepted by the Linksys ATA device.
The default is parallel.
Specify the syslog server name and port. This feature specifies the server for logging
Linksys ATA device system information and critical events. If both Debug Server and
Syslog Server are specified, Syslog messages are also logged to the Debug Server.
The debug server name and port. This feature specifies the server for logging Linksys
ATA device debug information. The level of detailed output depends on the debug level parameter setting.
The higher the debug level, the more debug information is generated. Zero (0) means no debug information is generated. To log SIP messages, Debug Level must be set to at least 2.
The default is 0.
IP address or name of primary NTP server.
IP address or name of secondary NTP server.
Miscellaneous Settings (not used with PAP2T)
Field
Syslog Server
Debug Server
Debug Level
Description
Specifies the IP address of the syslog server.
Specifies the IP address of the debug server, which logs debug information. The level of detailed output depends on the debug level parameter setting.
Determines the level of debug information that is generated. Select 0, 1, 2, or 3 from the drop-down menu. The higher the debug level, the more debug information is generated.
The default is 0, which indicates that no debug information is generated.
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SIP Tab
SIP Tab
This section describes the fields for the following headings on the SIP tab (2102, 3102):
•
”SIP Parameters” section on page 74
•
”SIP Timer Values (sec)” section on page 75
•
”Response Status Code Handling” section on page 77
•
”RTP Parameters” section on page 77
•
”SDP Payload Types” section on page 78
•
”NAT Support Parameters” section on page 79
SIP Parameters
Field
Max Forward
Max Redirection
Max Auth
SIP User Agent Name
Description
SIP Max Forward value, which can range from 1 to 255.
The default is 70.
Number of times an invite can be redirected to avoid an infinite loop.
The default is 5.
Maximum number of times (from 0 to 255) a request may be challenged.
The default is 2.
User-Agent header used in outbound requests.
The default is $VERSION. If empty, the header is not included. Macro expansion of
$A to $D corresponding to GPP_A to GPP_D allowed.
SIP Server Name Server header used in responses to inbound responses.
The default is $VERSION.
SIP Reg User Agent Name User-Agent name to be used in a REGISTER request. If this is not specified, the SIP
User Agent Name parameter is also used for the REGISTER request.
The default is blank.
SIP Accept Language
DTMF Relay MIME Type
Accept-Language header used. There is no default (this indicates Linksys ATA device 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.
The default is application/dtmf-relay.
Hook Flash MIME Type
Remove Last Reg
MIME Type used in a SIP INFO message to signal a hook flash event.
The default is 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.
The default is no.
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SIP Tab
Use Compact Header
Escape Display Name
RFC 2543 Call Hold
Mark All AVT Packets
SIP TCP Port Min
SIP TCP Port Max
Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the Linksys ATA device uses compact SIP headers in outbound SIP messages. If set to no, the Linksys ATA device uses normal SIP headers. If inbound SIP requests contain compact headers, Linksys ATA device 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,
Linksys ATA device substitutes those headers with compact headers (if defined by
RFC 261) if Use Compact Header parameter is set to yes.
The default is no.
Lets you keep the Display Name private. Select yes if you want the Linksys ATA device 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.
The default is no.
Configures the type of call hold: a:sendonly or 0.0.0.0.
The default is no; do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax.
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.
The default is yes.
Specifies the lowest TCP port number that can be used for SIP sessions. This field is not found in the PAP2T.
Specifies the highest TCP port number that can be used for SIP sessions. This field is not found in the PAP2T.
SIP Timer Values (sec)
Field
SIP T1
SIP T2
SIP T4
SIP Timer B
SIP Timer F
SIP Timer H
SIP Timer D
Description
RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds.
The default is.5.
RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds.
The default is 4.
RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds.
The default is 5.
INVITE time-out value, which can range from 0 to 64 seconds.
The default is 32.
Non-INVITE time-out value, which can range from 0 to 64 seconds.
The default is 32.
INVITE final response, time-out value, which can range from 0 to 64 seconds.
The default is 32.
ACK hang-around time, which can range from 0 to 64 seconds.
The default is 32.
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SIP Tab
SIP Timer J
INVITE Expires
ReINVITE Expires
Reg Min Expires
Reg Max Expires
Non-INVITE response hang-around time, which can range from 0 to 64 seconds.
The default is 32.
INVITE request Expires header value. If you enter 0, the Expires header is not included in the request.
The default is 240. Range: 0–(2
31
–1).
ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request.
The default is 30. Range: 0–(2
31
–1).
Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used.
The default is 1.
Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used.
The default is 7200.
Reg Retry Intvl
Reg Retry Long Intvl
Interval to wait before the Linksys ATA device retries registration after failing during the last registration.
The default is 30.
When registration fails with a SIP response code that does not match
Retry Reg RSC, the Linksys ATA device waits for the specified length of time before retrying. If this interval is 0, the Linksys ATA device stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.
The default is 1200.
Reg Retry Random Delay Random delay range (in seconds) to add to Register Retry Intvl when retrying
REGISTER after a failure.
The default is 0, which disables this feature.
Reg Retry Long Random
Delay
Random delay range (in seconds) to add to Register Retry Long Intvl when retrying
REGISTER after a failure.
The default is 0, which disables this feature.
Reg Retry Intvl Cap The maximum value to cap the exponential back-off retry delay (which starts at
Register Retry Intvl and doubles on every REGISTER retry after a failure). In other words, the retry interval is always at Register Retry Intvl seconds after a failure. If this feature is enabled, Reg Retry Random Delay is added on top of the exponential backoff adjusted delay value.
The default value is 0, which disables the exponential back-off feature.
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SIP Tab
Response Status Code Handling
Field
SIT1 RSC
SIT2 RSC
SIT3 RSC
SIT4 RSC
Try Backup RSC
Retry Reg RSC
Description
SIP response status code for the appropriate Special Information Tone (SIT). For example, if you set the SIT1 RSC to 404, when the user makes a call and a failure code of 404 is returned, the SIT1 tone is played. Reorder or Busy tone is played by default for all unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.
SIP response status code to INVITE on which to play the SIT2 Tone.
SIP response status code to INVITE on which to play the SIT3 Tone.
SIP response status code to INVITE on which to play the SIT4 Tone.
SIP response code that retries a backup server for the current request.
Interval to wait before the Linksys ATA device retries registration after failing during the last registration.
The default is 30.
RTP Parameters
Field
RTP Port Min
RTP Port Max
RTP Packet Size
Max RTP ICMP Err
Description
Minimum port number for RTP transmission and reception. The RTP Port Min and RTP
Port Max parameters should define a range that contains at least 4 even number ports, such as 100 – 106.
The default is 16384.
Maximum port number for RTP transmission and reception.
The default is 16482.
Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds.
The default is 0.030.
Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the Linksys ATA device terminates the call. If value is set to 0, the Linksys
ATA device ignores the limit on ICMP errors.
The default is 0.
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SIP Tab
RTCP Tx Interval
No UDP Checksum
Stats In BYE
Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the Linksys ATA device can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a
SDES.(Source Description). The last RTCP packet contains an additional BYE packet.
Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to
<User ID>@<Proxy>, NAME is set to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version (such as Linksys/Linksys ATA device-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the Linksys ATA device’s local time, not the time reported by an NTP server. If the Linksys ATA device receives a RR from the peer, it attempts to compute the round trip delay and show it as the <Call Round Trip Delay> value (ms) in the Info section of Linksys ATA device web page.
The default is 0.
Select yes if you want the Linksys ATA device to calculate the UDP header checksum for SIP messages. Otherwise, select no.
The default is no.
Determines whether the Linksys ATA device includes the P-RTP-Stat header or response to a BYE message. The header contains the RTP statistics of the current call.
Select yes or no from the drop-down menu. The format of the P-RTP-Stat header is:
P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in ms>,DU=<call duration in s>,EN=<encoder>,DE=<decoder>.
The default is no.
SDP Payload Types
Field Description
NSE Dynamic Payload
AVT Dynamic Payload
INFOREQ Dynamic
Payload
G726r16 Dynamic
Payload
NSE dynamic payload type. The valid range is 96-127.
The default is 100.
AVT dynamic payload type. The valid range is 96-127.
The default is 101.
INFOREQ dynamic payload type.
There is no default.
G.726-16 dynamic payload type. The valid range is 96-127.
The default is 98.
G726r24 Dynamic
Payload
G726r40 Dynamic
Payload
G.726-24 dynamic payload type. The valid range is 96-127.
The default is 97.
G.726-40 dynamic payload type. The valid range is 96-127.
The default is 96.
G729b Dynamic Payload G.729b dynamic payload type. The valid range is 96-127.
The default is 99.
NSE Codec Name NSE codec name used in SDP.
The default is NSE.
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SIP Tab
AVT Codec Name
G711u Codec Name
G711a Codec Name
G726r16 Codec Name
G726r24 Codec Name
G726r32 Codec Name
G726r40 Codec Name
G729a Codec Name
G729b Codec Name
G723 Codec Name
EncapRTP Codec Name
AVT codec name used in SDP.
The default is telephone-event.
G.711u codec name used in SDP.
The default is PCMU.
G.711a codec name used in SDP.
The default is PCMA.
G.726-16 codec name used in SDP.
The default is G726-16.
G.726-24 codec name used in SDP.
The default is G726-24.
G.726-32 codec name used in SDP.
The default is G726-32.
G.726-40 codec name used in SDP.
The default is G726-40.
G.729a codec name used in SDP.
The default is G729a.
G.729b codec name used in SDP.
The default is G729ab.
G.723 codec name used in SDP.
The default is G723.
EncapRTP codec name used in SDP.
The default is EncapRTP.
NAT Support Parameters
Field
Handle VIA received
Handle VIA rport
Insert VIA received
Insert VIA rport
Substitute VIA Addr
Description
If you select yes, the Linksys ATA device processes the received parameter in the VIA header (this is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.
The default is no.
If you select yes, the Linksys ATA device processes the rport parameter in the VIA header (this is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.
The default is no.
Inserts the received parameter into the VIA header of SIP responses if the receivedfrom IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu.
The default is no.
Inserts the parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu.
The default is no.
Lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the drop-down menu.
The default is no.
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SIP Tab
Send Resp To Src Port
STUN Enable
STUN Test Enable
STUN Server
EXT IP
EXT RTP Port Min
NAT Keep Alive Intvl
Sends responses to the request source port instead of the VIA sent-by port. Select yes or no from the drop-down menu.
The default is no.
Enables the use of STUN to discover NAT mapping. Select yes or no from the dropdown menu.
The default is no.
If the STUN Enable feature is enabled and a valid STUN server is available, the Linksys
ATA device can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a
Warning header in all subsequent REGISTER requests. If the Linksys ATA device detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.
The default is no.
IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery.
External IP address to substitute for the actual IP address of the Linksys ATA device in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.
If this parameter is specified, the Linksys ATA device assumes this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line). However, the results of STUN and VIA received parameter processing, if available, supersede this statically configured value.
The default is 0.0.0.0.
External port mapping number of the RTP Port Min. number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range.
The default is 0.
Interval between NAT-mapping keep alive messages.
The default is 15.
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Regional Tab
This section describes the fields for the following headings on the Regional tab:
•
”Call Progress Tones” section on page 81
•
”Distinctive Ring Patterns” section on page 83
•
”Distinctive Call Waiting Tone Patterns” section on page 83
•
”Distinctive Ring/CWT Pattern Names” section on page 84
•
”Ring and Call Waiting Tone Spec” section on page 85
•
”Control Timer Values (sec)” section on page 85
•
”Vertical Service Activation Codes” section on page 86
•
”Vertical Service Announcement Codes (SPA2102)” section on page 91
•
”Outbound Call Codec Selection Codes” section on page 91
•
”Miscellaneous” section on page 92
Call Progress Tones
Field
Dial Tone
Second Dial Tone
Outside Dial Tone
Prompt Tone
Busy Tone
Reorder Tone
Off Hook Warning Tone
Ring Back Tone
Description
Prompts the user to enter a phone number. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.
The default is 350@-19,440@-19;10(*/0/1+2).
Alternative to the Dial Tone when the user dials a three-way call.
The default is 420@-19,520@-19;10(*/0/1+2).
Alternative to the Dial Tone. It prompts the user to enter an external phone number, as opposed to an internal extension. It is triggered by a, (comma) character encountered in the dial plan.
The default is 420@-19;10(*/0/1).
Prompts the user to enter a call forwarding phone number.
The default is 520@-19,620@-19;10(*/0/1+2).
Played when a 486 RSC is received for an outbound call.
The default is 480@-19,620@-19;10(.5/.5/1+2).
Played when an outbound call has failed or after the far end hangs up during an established call. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.
The default is 480@-19,620@-19;10(.25/.25/1+2).
Played when the caller has not properly placed the handset on the cradle. Off Hook
Warning Tone is played when Reorder Tone times out.
The default is 480@10,620@0;10(.125/.125/1+2).
Played during an outbound call when the far end is ringing.
The default is 440@-19,480@-19;*(2/4/1+2).
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Ring Back 2 Tone
Confirm Tone
SIT1 Tone
SIT2 Tone
SIT3 Tone
SIT4 Tone
MWI Dial Tone
Cfwd Dial Tone
Holding Tone
Conference Tone
Secure Call Indication
Tone
VoIP PIN Tone
PSTN PIN Tone
Feature Invocation Tone
Your ATA device plays this ringback tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request. The default value is the same as Ring Back Tone, except the cadence is 1s on and 1s off.
The default is 440@-19,480@-19;*(1/1/1+2).
Brief tone to notify the user that the last input value has been accepted.
The default is 600@-16; 1(.25/.25/1).
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
The default is 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/
0).
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
The default is 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/
0).
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
The default is 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/
0).
This is an alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
The default is 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/
0).
Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.
The default is 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2).
Played when all calls are forwarded.
The default is 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2).
Informs the local caller that the far end has placed the call on hold.
The default is 600@-19*(.1/.1/1,.1/.1/1,.1/9.5/1).
Played to all parties when a three-way conference call is in progress.
The default is 350@-19;20(.1/.1/1,.1/9.7/1).
Played when a call has been successfully switched to secure mode. It should be played only for a short while (less than 30 seconds) and at a reduced level (less than -
19 dBm) so it does not interfere with the conversation.
The default is 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).
Specification of the tone played to prompt a VoIP caller for a PIN number (if PIN authentication is selected and the caller requires authentication to use the PSTN gateway).
The default is 600@-10;*(0/1/1,.1/.1/1,.1/.1/1,.1/.5/1).
Specification of the tone played to prompt a PSTN caller for a PIN number (if PIN authentication is selected and the caller requires authentication to use the VoIP gateway).
The default is 600@-10;*(0/.7/1,.2/.1/1,.2/.1/1,.2/.5/1).
Played when a feature is implemented.
The default is 350@-16;*(.1/.1/1).
This field is not found in the PAP2T.
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Distinctive Ring Patterns
Field
CWT1 Cadence
CWT2 Cadence
CWT3 Cadence
CWT4 Cadence
CWT5 Cadence
CWT6 Cadence
CWT7 Cadence
Field
Ring1 Cadence
Ring2 Cadence
Ring3 Cadence
Ring4 Cadence
Ring5 Cadence
Ring6 Cadence
Description
Cadence script for distinctive ring 1.
The default is 60(2/4).
Cadence script for distinctive ring 2.
The default is 60(.3/.2, 1/.2,.3/4).
Cadence script for distinctive ring 3.
The default is 60(.8/.4,.8/4).
Cadence script for distinctive ring 4.
The default is 60(.4/.2,.3/.2,.8/4).
Cadence script for distinctive ring 5.
The default is 60(.2/.2,.2/.2,.2/.2,1/4).
Cadence script for distinctive ring 6.
The default is 60(.2/.4,.2/.4,.2/4).
Ring7 Cadence
Ring8 Cadence
Ring9 Cadence
Cadence script for distinctive ring 7.
The default is 60(.4/.2,.4/.2,.4/4)
.
Cadence script for distinctive ring 8.
The default is 60(0.25/9.75)
.
Cadence script for distinctive ring 9. This field is for the SPA2102 only.
The default is 60(.4/.2,.4/2)
.
Distinctive Call Waiting Tone Patterns
Description
Cadence script for distinctive CWT 1.
The default is 30(.3/9.7)
.
Cadence script for distinctive CWT 2.
The default is 30(.1/.1, .1/9.7)
.
Cadence script for distinctive CWT 3.
The default is 30(.1/.1, .1/.1, .1/9.3)
.
Cadence script for distinctive CWT 4.
The default is 30(.1/.1, .3/.1, .1/9.5).
Cadence script for distinctive CWT 5.
The default is 30(.3/.1,.1/.1,.3/9.1)
.
Cadence script for distinctive CWT 6.
The default is 30(.3/.1,.3/.1,.1/9.1).
Cadence script for distinctive CWT 7.
The default is 30(.1/.1, .3/.1, .1/9.3).
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CWT8 Cadence
CWT9 Cadence
Cadence script for distinctive CWT 8.
The default is 2.3(.3/2).
Cadence script for distinctive CWT 9. This field is for the SPA2102 only.
The default is 30(.3/9.7).
Distinctive Ring/CWT Pattern Names
Field
Ring1 Name
Ring2 Name
Ring3 Name
Ring4 Name
Ring5 Name
Ring6 Name
Ring7 Name
Ring8 Name
Ring9 Name
Description
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call.
The default is Bellcore-r1.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call.
The default is Bellcore-r2
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.
The default is Bellcore-r3
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call.
The default is Bellcore-r4
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call.
The default is Bellcore-r5
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call.
The default is Bellcore-r6
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call.
The default is Bellcore-r7
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call.
The default is Bellcore-r8
.
Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 9 for the inbound call. This field is for the SPA2102 only.
The default is Bellcore-r9
.
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Ring and Call Waiting Tone Spec
Ring and Call Waiting tones don’t work the same way on all phones. When setting ring tones, consider the following recommendations:
1. Begin with the default Ring Waveform, Ring Frequency, and Ring Voltage.
2. If your ring cadence doesn’t sound right, or your phone doesn’t ring, change your Ring
Waveform, Ring Frequency, and Ring Voltage to the following: a. Ring Waveform: Sinusoid b. Ring Frequency: 25 c. Ring Voltage: 80V
Field
Ring Waveform
Ring Frequency
Ring Voltage
CWT Frequency
Synchronized Ring
Description
Waveform for the ringing signal. Choices are Sinusoid or Trapezoid. The default is
Trapezoid.
Frequency of the ringing signal. Valid values are 10–100 (Hz). The default is 20.
Ringing voltage. Choices are 60–90 (V). The default is 85.
Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.
The default is 440@-10.
If this is set to Yes, when a device calls the Linksys ATA, both lines ring at the same time (similar to a regular PSTN line). After one line answers, the other stops ringing.
This field is only found in the PAP2T. No is the default.
Control Timer Values (sec)
Field
Hook Flash Timer Min
Hook Flash Timer Max
Callee On Hook Delay
Reorder Delay
Call Back Expires
Description
Minimum on-hook time before off-hook qualifies as hook-flash. Less than this the on-hook event is ignored. Range: 0.1–0.4 seconds.
The default is 0.1.
Maximum on-hook time before off-hook qualifies as hook-flash. More than this the on-hook event is treated as on-hook (no hook-flash event). Range: 0.4–1.6 seconds.
The default is 0.9.
Phone must be on-hook for at this time in sec before the Linksys ATA device will tear down the current inbound call. It does not apply to outbound calls. Range: 0–255 seconds.
The default is 0.
Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf
= never plays. Range: 0–255 seconds.
The default is 5.
Expiration time in seconds of a call back activation. Range: 0–65535 seconds.
The default is 1800.
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Call Back Retry Intvl
Call Back Delay
VMWI Refresh Intvl
Interdigit Long Timer
Interdigit Short Timer
CPC Delay
CPC Duration
Call back retry interval in seconds. Range: 0–255 seconds.
The default is 30.
Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the Linksys ATA device still considers the call as failed and keeps on retrying.
The default is 0.5.
Interval between VMWI refresh to the CPE.
The default is 0.5.
Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds.
The default is 10.
Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.
The default is 3.
Delay in seconds after caller hangs up when the Linksys ATA device starts removing the tip-and-ring voltage to the attached equipment of the called party. Range: 0–255 seconds. Linksys ATA device has had polarity reversal feature since release 1.0 which can be applied to both the caller and the callee end. This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected (remote end has answered) or disconnected (remote end has hung up). This feature should be disabled for the called party (in other words, by using the same polarity for connected and idle state) and the CPC feature should be used instead.
Without CPC enabled, reorder tone will is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored Resolution is 1 second.
The default is 2.
Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second.
The default is 0 (CPC disabled).
Vertical Service Activation Codes
Vertical Service Activation Codes are automatically appended to the dial-plan. There is
no
need to include them in dial-plan, although
no
harm is done if they are included.
Field
Call Return Code
Call Redial Code
Description
This code calls the last caller.
The default is *69.
Redials the last number called. This field is not found in the PAP2T.
The default is *07.
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Blind Transfer Code
Call Back Act Code
Call Back Deact Code
Call Back Busy Act Code
Cfwd All Act Code
Cfwd All Deact Code
CW Per Call Act Code
Begins a blind transfer of the current call to the extension specified after the activation code.
The default is *98.
Starts a callback when the last outbound call is not busy.
The default is *66.
Cancels a callback.
The default is *86.
Starts a callback when the last outbound call is busy. This field is only found in the
PAP2T.
The default is *05
Forwards all calls to the extension specified after the activation code.
The default is *72.
Cancels call forwarding of all calls.
The default is *73.
Cfwd Busy Act Code
Cfwd Busy Deact Code
Cfwd No Ans Act Code
Cfwd No Ans Deact Code Cancels call forwarding of no-answer calls.
The default is *93.
Cfwd Last Act Code
Forwards no-answer calls to the extension specified after the activation code.
The default is *92.
Cfwd Last Deact Code
Block Last Act Code
Block Last Deact Code
Accept Last Act Code
Forwards the last inbound or outbound calls to the extension specified after the activation code.
The default is *63.
Cancels call forwarding of the last inbound or outbound calls.
The default is *83.
Blocks the last inbound call.
The default is *60.
Cancels blocking of the last inbound call.
The default is *80.
Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
The default is *64.
Accept Last Deact Code
Forwards busy calls to the extension specified after the activation code.
The default is *90.
Cancels call forwarding of busy calls.
The default is *91.
CW Act Code
Cancels the code to accept the last outbound call.
The default is *84.
Enables call waiting on all calls.
The default is *56.
CW Deact Code Disables call waiting on all calls.
The default is *57.
Enables call waiting for the next call.
The default is *71.
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CW Per Call Deact Code
Block CID Act Code
Block CID Deact Code
Block CID Per Call Act
Code
Block CID Per Call Deact
Code
Block ANC Act Code
Block ANC Deact Code
DND Act Code
DND Deact Code
CID Act Code
CID Deact Code
CWCID Act Code
CWCID Deact Code
Dist Ring Act Code
Dist Ring Deact Code
Speed Dial Act Code
Disables call waiting for the next call.
The default is *70.
Blocks caller ID on all outbound calls.
The default is *67.
Removes caller ID blocking on all outbound calls.
The default is *68.
Blocks caller ID on the next outbound call.
The default is *81.
Removes caller ID blocking on the next inbound call.
The default is *82.
Blocks all anonymous calls.
The default is *77.
Removes blocking of all anonymous calls.
The default is *87.
Enables the do not disturb feature.
The default is *78.
Disables the do not disturb feature.
The default is *79.
Enables caller ID generation.
The default is *65.
Disables caller ID generation.
The default is *85.
Enables call waiting, caller ID generation.
The default is *25.
Disables call waiting, caller ID generation.
The default is *45.
Enables the distinctive ringing feature.
The default is *26
Disables the distinctive ringing feature.
The default is *46.
Assigns a speed dial number.
The default is *74.
Secure All Call Act Code
Secure No Call Act Code
Makes all outbound calls secure.
The default is *16.
Makes all outbound calls not secure.
The default is *17.
Secure One Call Act Code Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
The default is *18.
Secure One Call Deact
Code
Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
The default is *19.
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Conference Act Code
Attn-Xfer Act Code
If this code is specified, the user must enter it before dialing the third party for a conference call. Enter the code for a conference call.
If the code is specified, the user must enter it before dialing the third party for a call transfer. Enter the code for a call transfer.
Modem Line Toggle Code Toggles the line to a modem.
The default is *99. Modem pass-through mode can be triggered only by pre-dialing this code.
FAX Line Toggle Code Toggles the line to a fax machine. This field is not found in the PAP2T.
The default is #99.
Referral Services Codes These codes tell the Linksys ATA device what to do when the user places the current call on hold and is listening to the second dial tone.
One or more *code can be configured into this parameter, such as *98, or
*97|*98|*123, etc. Max total length is 79 chars. This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone.
Each *code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the Linksys ATA device to perform a blind transfer to a target number that is prepended by the service *code.
For example, after the user dials *98, the Linksys ATA device plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the Linksys ATA device sends a blind REFER to the holding party with the
Refer-To target equals to *98 target_number. This feature allows the Linksys ATA device to hand off a call to an application server to perform further processing, such as call park.
The *codes should not conflict with any of the other vertical service codes internally processed by the Linksys ATA device. You can empty the corresponding *code that you do not want to Linksys ATA device to process.
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Feature Dial Services
Codes
These codes tell the Linksys ATA device what to do when the user is listening to the first or second dial tone.
One or more *code can be configured into this parameter, such as *72, or
*72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user has a dial tone (first or second dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the Linksys
ATA device to call the target number prepended by the *code. For example, after user dials *72, the Linksys ATA device plays a special tone called a Prompt tone while awaiting the user to enter a valid target number. When a complete number is entered, the Linksys ATA device sends a INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features like call forward (*72) or BLock
Caller ID (*67).
The *codes should not conflict with any of the other vertical service codes internally processed by the Linksys ATA device. You can empty the corresponding *code that you do not want to Linksys ATA device to process.
You can add a parameter to each *code in Features Dial Services Codes to indicate what tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed tone parameters (note the use of back quotes surrounding the parameter w/o spaces)
‘c‘ = <Cfwd Dial Tone>
‘d‘ = <Dial Tone>
‘m‘ = <MWI Dial Tone>
‘o‘ = <Outside Dial Tone>
‘p‘ = <Prompt Dial Tone>
‘s‘ = <Second Dial Tone>
‘x‘ = No tones are place, x is any digit not used above
If no tone parameter is specified, the Linksys ATA device plays Prompt tone by default.
If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include it in this parameter. In that case, simple add that *code in the dial plan and the Linksys ATA device send INVITE *73@..... as usual when user dials
*73.
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Vertical Service Announcement Codes (SPA2102)
Field
Service Annc Base
Number
Service Annc Extension
Codes
Description
Base number for service announcements.
Extension codes for service announcements.
Field
Prefer G711u Code
Force G711u Code
Prefer G711a Code
Force G711a Code
Prefer G723 Code
Force G723 Code
Prefer G726r16 Code
Force G726r16 Code
Prefer G726r24 Code
Force G726r24 Code
Prefer G726r32 Code
Force G726r32 Code
Prefer G726r40 Code
Outbound Call Codec Selection Codes
These codes automatically appended to the dial-plan. So no need to include them in dial-plan
(although no harm to do so either).
Description
Makes this codec the preferred codec for the associated call.
The default is *017110.
Makes this codec the only codec that can be used for the associated call.
The default is *027110.
Makes this codec the preferred codec for the associated call.
The default is *017111
Makes this codec the only codec that can be used for the associated call.
The default is *027111.
Makes this codec the preferred codec for the associated call.
The default is *01723.
Makes this codec the only codec that can be used for the associated call.
The default is *02723.
Makes this codec the preferred codec for the associated call.
The default is *0172616.
Makes this codec the only codec that can be used for the associated call.
The default is *0272616.
Makes this codec the preferred codec for the associated call.
The default is *0172624.
Makes this codec the only codec that can be used for the associated call.
The default is *0272624.
Makes this codec the preferred codec for the associated call.
The default is *0172632.
Makes this codec the only codec that can be used for the associated call.
The default is *0272632.
Makes this codec the preferred codec for the associated call.
The default is *0172640.
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Force G726r40 Code
Prefer G729a Code
Force G729a Code
Makes this codec the only codec that can be used for the associated call.
The default is *0272640.
Makes this codec the preferred codec for the associated call.
The default is *01729.
Makes this codec the only codec that can be used for the associated call.
The default is *02729.
Miscellaneous
Field
Set Local Date (mm/dd)
Description
Sets the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits.
Set Local Time (HH/mm) Sets the local time (hh stands for hours and mm stands for minutes). Seconds are optional.
Time Zone Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00, …,
GMT+13:00.
The default is GMT-08:00.
FXS Port Impedance Sets the electrical impedance of the FXS port. Choices are 600, 900, 600+2.16uF,
900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or
200+600||100nF.
The default is 600.
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Daylight Saving Time Rule Enter the rule for calculating daylight saving time; it should include the start, end, and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below. Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by 0:0:0 of the given date.
This is the format of the rule: Start = <start-time>; end=<end-time>; save = <savetime>.
The <start-time> and <end-time> values specify the start and end dates and times of daylight saving time. Each value is in this format: <month> /<day> / <weekday>[/
HH:[mm[:ss]]]
The <save-time> value is the number of hours, minutes, and/or seconds to add to the current time during daylight saving time. The <save-time> value can be preceded by a negative (-) sign if subtraction is desired instead of addition. The
<save-time> value is in this format: [/[+|-]HH:[mm[:ss]]]
The <month> value equals any value in the range 1-12 (January-December).
The <day> value equals [+|-] any value in the range 1-31.
If <day> is 1, it means the <weekday> on or before the end of the month (in other words the last occurrence of < weekday> in that month).
The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can also equal 0. If the <weekday> value is 0, this means that the date to start or end daylight saving is exactly the date given. In that case, the <day> value must not be negative.
If the <weekday> value is not 0 and the <day> value is positive, then daylight saving starts or ends on the <weekday> value on or after the date given. If the <weekday> value is not 0 and the <day> value is negative, then daylight saving starts or ends on the <weekday> value on or before the date given.
The abbreviation HH stands for hours (0-23).
The abbreviation mm stands for minutes (0-59).
The abbreviation ss stands for seconds (0-59).
The default Daylight Saving Time Rule is start=4/1/7;end=10/-1/7;save=1.
Daylight Saving Time
Enable
FXS Port Input Gain
Daylight Saving Time can be turned on or off. This option affects the time stamp on
CallerID and affects all the lines and extensions of the phone. Default is Yes (on).
Input gain in dB, up to three decimal places. The range is 6.000 to -12.000.
The default is -3.
FXS Port Output Gain
DTMF Playback Level
DTMF Playback Length
Detect ABCD
Playback ABCD
Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call
Progress Tones and DTMF playback level are not affected by the FXS Port Output Gain parameter.
The default is -3.
Local DTMF playback level in dBm, up to one decimal place.
The default is -16.0.
Local DTMF playback duration in milliseconds.
The default is .1.
To enable local detection of DTMF ABCD, select yes. Otherwise, select no.
The default is yes. Setting has no effect if DTMF Tx Method is INFO; ABCD is always sent OOB regardless in this setting.
To enable local playback of OOB DTMF ABCD, select yes. Otherwise, select no.
The default is yes.
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Regional Tab
Caller ID Method
Caller ID FSK Standard
FXS Port Power Limit
Feature Invocation
Method
More Echo Suppression
The following choices are available:
• Bellcore (N.Amer,China)—CID, CIDCW, and VMWI. FSK sent after first ring
(same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).
• DTMF (Finland, Sweden)—CID only. DTMF sent after polarity reversal (and no
DTAS) and before first ring.
• DTMF (Denmark)—CID only. DTMF sentbefore first ring with no polarity reversal and no DTAS.
• ETSI DTMF—CID only. DTMF sent after DTAS (and no polarity reversal) and before first ring.
• ETSI DTMF With PR—CID only. DTMF sent after polarity reversal and DTAS and before first ring.
• ETSI DTMF After Ring—CID only. DTMF sent after first ring (no polarity reversal or DTAS).
• ETSI FSK—CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity reversal) and before first ring. Waits for ACK from CPE after DTAS for CIDCW.
• ETSI FSK With PR (UK)—CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from CPE after DTAS for
CIDCW. Polarity reversal is applied only if equipment is on hook.
• DTMF (Denmark) With PR—CID only. DTMF sent after polarity reversal (and no
DTAS) and before first ring.
The default is Bellcore(N.Amer, China).
The Linksys ATA device supports bell 202 and v.23 standards for caller ID generation.
Select the FSK standard you want to use, bell 202 or v.23.
The default is bell 202.
This field is not found in the PAP2T.
The choices are from 1 to 8. This field is only found in the PAP2T.
The default is 3.
Select the method you want to use, Default or Sweden default. This field is not found in the PAP2T.
The default is Default.
Enable or disable more echo suppresion. The default is no.
This field is not found in the PAP2T.
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Line Tab(s)
Line Tab(s)
This section describes the fields for the following headings on the Line tabs:
•
”Line Enable” section on page 95
•
”Streaming Audio Server (SAS)” section on page 96
•
”NAT Settings” section on page 97
•
”Network Settings” section on page 96
•
”SIP Settings” section on page 97
•
”Call Feature Settings” section on page 99
•
”Proxy and Registration” section on page 100
•
”Subscriber Information” section on page 101
•
”Supplementary Service Subscription” section on page 101
•
”Audio Configuration” section on page 104
•
”VoIP Fallback to PSTN (SPA3102/AG310)” section on page 107
•
”Gateway Accounts (SPA3102/AG310)” section on page 107
•
”Dial Plan” section on page 107
•
”FXS Port Polarity Configuration” section on page 108
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
Field
Line Enable
Description
To enable this line for service, select yes. Otherwise, select no.
The default is yes.
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Line Tab(s)
Streaming Audio Server (SAS)
Field
SAS Enable
SAS DLG Refresh Intvl
SAS Inbound RTP Sink
Description
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 autoanswers incoming calls and streams audio RTP packets to the caller.
The default is no.
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
ATA device 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.
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 ATA device’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.
Network Settings
Field
SIP ToS/DiffServ Value
SIP CoS Value [0-7]
RTP ToS/DiffServ Value
RTP CoS Value [0-7]
Description
TOS/DiffServ field value in UDP IP packets carrying a SIP message.
The default is 0x68.
CoS value for SIP messages.
The default is 3.
ToS/DiffServ field value in UDP IP packets carrying RTP data.
The default is 0xb8.
CoS value for RTP data.
The default is 6.
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Line Tab(s)
Network Jitter Level
Jitter Buffer Adjustment
Determines how jitter buffer size is adjusted by the Linksys ATA device. 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.
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.
NAT Settings
Field
NAT Mapping Enable
NAT Keep Alive Enable
NAT Keep Alive Msg
NAT Keep Alive Dest
Description
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes.
Otherwise, select no.
The default is no.
To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.
The default is no.
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.
Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server.
The default is $PROXY.
SIP Settings
Field
SIP Port
SIP Transport
SIP 100REL Enable
Description
Port number of the SIP message listening and transmission port.
The default is 5060.
The TCP choice provides “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP.
In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. Options are: UDP,
TCP, TLS. The default is UDP.
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.
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Line Tab(s)
EXT SIP Port
Auth Resync-Reboot
SIP Proxy-Require
SIP Remote-Party-ID
SIP GUID
SIP Debug Option
RTP Log Intvl
Restrict Source IP
The external SIP port number.
If this feature is enabled, the Linksys ATA device 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.
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.
To use the Remote-Party-ID header instead of the From header, select yes.
Otherwise, select no.
The default is yes.
This field is not found in the PAP2T.
The Global Unique ID is generated for each line for each device. When it is enabled, the Linksys ATA device 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.
SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows:
• none—No logging.
• 1-line—Logs the start-line only for all messages.
• 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses.
• 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses.
• 1-line excl. REG—Logs the start-line only for all messages except REGISTER requests/responses.
• 1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except
OPTIONS, NOTIFY, and REGISTER requests/responses.
• full—Logs all SIP messages in full text.
• full excl. OPT—Logs all SIP messages in full text except OPTIONS requests/ responses.
• full excl. NTFY—Logs all SIP messages in full text except NOTIFY requests/ responses.
• full excl. REG—Logs all SIP messages in full text except REGISTER requests/ responses.
• full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS,
NOTIFY, and REGISTER requests/responses.
The default is none.
The interval for the RTP log.
If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the PAP2T will drop all packets sent to its SIP Ports originated from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use
Outbound Proxy is yes).
The default is no.
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Line Tab(s)
Referor Bye Delay
Controls when the Linksys ATA device sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate period of time in seconds.
The default is 4.
Refer Target Bye Delay
For the Refer Target Bye Delay, enter the appropriate period of time in seconds.
The default is 0.
Referee Bye Delay
For the Referee Bye Delay, enter the appropriate period of time in seconds.
The default is 0.
Refer-To Target Contact
To contact the refer-to target, select yes. Otherwise, select no.
The default is no.
Sticky 183
Auth INVITE
When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.
Reply 182 On Call Waiting
When set to yes, your ATA device replies with a SIP 182 response to the caller if it is already in a call and the phone is off-hook. To use this feature, select yes. Otherwise, keep the default, no.
This field is found on the SPA2102 and SPA3102 only.
Use Anonymous with
RPID
When set to yes, use “anonymous” in the SIP message when remote party ID is requested in the SIP message. This field is found on the SPA2102 only.
Default is yes.
Use Local Addr in FROM
If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no.
The default is no.
The IP address of the local address enclosed in the FROM of the SIP message. This field is found on the SPA2102 only.
Default is no.
Call Feature Settings
Field
Blind Attn-Xfer Enable
Description
Enables the Linksys ATA device to perform an attended transfer operation by ending the current call leg and performing a blind transfer of the other call leg. If this feature is disabled, the Linksys ATA device performs an attended transfer operation by referring the other call leg to the current call leg while maintaining both call legs. To use this feature, select yes. Otherwise, select no.
MOH Server
The default is no.
User ID or URL of the auto-answering streaming audio server. When only a user ID is specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the MOH Server is not specified.
Xfer When Hangup Conf
Makes the Linksys ATA device perform a transfer when a conference call has ended.
Select yes or no from the drop-down menu.
The default is yes.
Conference Bridge URL
This feature supports external conference bridging for n-way conference calls (n > 2), instead of mixing audio locally. To use this feature, set this parameter to that of the server’s name, for example, [email protected]:12345 or conf (which uses the Proxy value as the domain). This field is found on the SPA2102 and PAP2T only.
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Line Tab(s)
Conference Bridge Ports
Select the maximum number of conference call participants. The range is 3 to 10.
The default is 3. This field is found on the SPA2102 and PAP2T only.
Proxy and Registration
Field
Proxy
Use Outbound Proxy
Outbound Proxy
Use OB Proxy In Dialog
Register
Make Call Without Reg
Register Expires
Ans Call Without Reg
Use DNS SRV
DNS SRV Auto Prefix
Proxy Fallback Intvl
Description
SIP proxy server for all outbound requests.
Enablse the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB
Proxy in Dialog parameters are ignored.
The default is no.
SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
Whether to force SIP requests to be sent to the outbound proxy within a dialog.
Ignored if the parameter Use Outbound Proxy is no, or the Outbound Proxy parameter is empty.
The default is yes.
Enable periodic registration with the Proxy parameter. This parameter is ignored if
Proxy is not specified.
The default is yes.
Allow making outbound calls without successful (dynamic) registration by the unit. If
No, dial tone will not play unless registration is successful.
The default is no.
Allow answering inbound calls without successful (dynamic) registration by the unit.
If proxy responded to REGISTER with a smaller Expires value, the PAP2T will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the PAP2T will retry with the value given in the Min-Expires header in the error response.
The default is 3600.
Expires value in sec in a REGISTER request. The PAP2T will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec
Whether to use DNS SRV lookup for Proxy and Outbound Proxy.
The default is no.
If enabled, the PAP2T will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.
The default is no.
This parameter sets the delay (sec) after which the PAP2T will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the PAP2T will not attempt to fall back after a fail over).
The default is 3600
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Line Tab(s)
Field
Proxy Redundancy
Method
Description
PAP2T will make an internal list of proxies returned in DNS SRV records. In normal mode, this list will contain proxies ranked by weight and priority.
if Based on SRV port is configured the PAP2T does normal first, and also inspect the port number based on 1st proxy’s port on the list.
The default is Normal.
Voice Mail Server
Enter the URL or IP address of the server.
Mailbox Subscribe Expires
Expiry time to the voicemail server. The time to send another subscribe message to the voicemail server.
Subscriber Information
Field
Display Name
Description
Display name for caller ID.
Extension number for this line.
User ID
Password
Use Auth ID
Password for this line.
To use the authentication ID and password for SIP authentication, select yes.
Otherwise, select no to use the user ID and password.
The default is no.
Auth ID
Directory Number
Call Capacity
Authentication ID for SIP authentication.
Enter the number for this line.
Maximum number of calls allowed on this line interface. Choices:
{unlimited,1,2,3,…25 }. Default is 16. Note that the Linksys ATA device does not distinguish between incoming and outgoing calls when talking about call capacity.
Note: unlimited = 16
Mini Certificate
SRTP Private Key
Base64 encoded of Mini-Certificate concatenated with the 1024-bit public key of the
CA signing the MC of all subscribers in the group.
The default is empty.
Base64 encoded of the 512-bit private key per subscriber for establishment of a secure call.
The default is empty.
Supplementary Service Subscription
The Linksys ATA device provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service. A supplementary service should be disabled if a) the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the Linksys ATA device.
Field
Call Waiting Serv
Description
Enable Call Waiting Service.
The default is yes.
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Line Tab(s)
Field
Block CID Serv
Block ANC Serv
Dist Ring Serv
Cfwd All Serv
Cfwd Busy Serv
Cfwd No Ans Serv
Cfwd Sel Serv
Cfwd Last Serv
Block Last Serv
Accept Last Serv
DND Serv
CID_Serv
CWCID Serv
Call Return Serv
Call Redial Serv
Call Back Serv
Three Way Call Serv
Three Way Conf Serv
Attn Transfer Serv
Description
Enable Block Caller ID Service.
The default is yes.
Enable Block Anonymous Calls Service
The default is yes.
Enable Distinctive Ringing Service
The default is yes.
Enable Call Forward All Service
The default is yes.
Enable Call Forward Busy Service
The default is yes.
Enable Call Forward No Answer Service
The default is yes.
Enable Call Forward Selective Service
The default is yes.
Enable Forward Last Call Service
The default is yes.
Enable Block Last Call Service
The default is yes.
Enable Accept Last Call Service
The default is yes.
Enable Do Not Disturb Service
The default is yes.
Enable Caller ID Service
The default is yes.
Enable Call Waiting Caller ID Service
The default is yes.
Enable Call Return Service
The default is yes.
Enable Call Redial Service. This field is not found in the PAP2T.
Enable Call Back Service.
Enable Three Way Calling Service. Three Way Calling is required for Three Way
Conference and Attended Transfer.
The default is yes.
Enable Three Way Conference Service. Three Way Conference is required for
Attended Transfer.
The default is yes.
Enable Attended Call Transfer Service. Three Way Conference is required for
Attended Transfer.
The default is yes.
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Line Tab(s)
Field
Unattn Transfer Serv
MWI Serv
VMWI Serv
Speed Dial Serv
Secure Call Serv
Referral Serv
Feature Dial Serv
Service Announcement
Serv
Description
Enable Unattended (Blind) Call Transfer Service.
The default is yes.
Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment.
The default is yes.
Enable VMWI Service (FSK).
The default is yes.
Enable Speed Dial Service.
The default is yes.
Enable Secure Call Service.
The default is yes.
Enable Referral Service. See the Referral Services Codes parameter for more details.
The default is yes.
Enable Feature Dial Service. See the Feature Dial Services Codes parameter for more details.
The default is yes.
Enable Service Announcement Service.
The default is yes.
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Line Tab(s)
Audio Configuration
A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the
G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G.729a resource is already allocated and since only one G.729a resource is allowed per device, no other low-bit-rate codec may be allocated for subsequent calls; the only choices are G711a and G711u. On the other hand, two
G.723.1/G.726 resources are available per device.
Therefore it is important to disable the use of G.729a in order to guarantee the support of two simultaneous G.723/G.726 codec.
Field Description
Preferred Codec
Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u,
G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723.
The default is G711u.
Second Preferred Codec
Second preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following:
Unspecified, G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or
G723.
The default is Unspecified.
Third Preferred Codec
Third preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following:
Unspecified, G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or
G723.
The default is Unspecified.
Use Pref Codec Only
To use only the preferred codec for all calls, select yes. (The call fails if the far end does not support this codec.) Otherwise, select no.
The default is no.
Silence Supp Enable
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no.
The default is no.
Silence Threshold
Select the appropriate setting for the threshold: high, medium, or low.
The default is medium.
G729a Enable
To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no.
The default is yes.
Echo Canc Enable
To enable the use of the echo canceller, select yes. Otherwise, select no.
The default is yes.
G723 Enable
To enable the use of the G.723a codec at 6.3 kbps, select yes. Otherwise, select no.
The default is yes.
Echo Canc Adapt Enable
To enable the echo canceller to adapt, select yes. Otherwise, select no.
The default is yes.
G726-16 Enable
To enable the use of the G.726 codec at 16 kbps, select yes. Otherwise, select no.
The default is yes.
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Echo Supp Enable
G726-24 Enable
FAX CED Detect Enable
G726-32 Enable
FAX CNG Detect Enable
G726-40 Enable
FAX Passthru Codec
DTMF Process INFO
FAX Codec Symmetric
DTMF Process AVT
FAX Passthru Method
DTMF Tx Method
DTMF Tx Mode
To enable the use of the echo suppressor, select yes. Otherwise, select no.
The default is yes.
To enable the use of the G.726 codec at 24 kbps, select yes. Otherwise, select no.
The default is yes.
To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no.
The default is yes.
To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no.
The default is yes.
To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no.
The default is yes.
To enable the use of the G.726 codec at 40 kbps, select yes. Otherwise, select no.
The default is yes.
Select the codec for fax passthrough, G711u or G711a.
The default is G711u.
To use the DTMF process info feature, select yes. Otherwise, select no.
The default is yes.
To force the Linksys ATA device to use a symmetric codec during fax passthrough, select yes. Otherwise, select no.
The default is yes.
To use the DTMF process AVT feature, select yes. Otherwise, select no. This field is not available for the PAP2T.
The default is yes.
Select the fax passthrough method: None, NSE, or ReINVITE.
The default is NSE.
Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO,
Auto, InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends DTMF as eypents. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation.
The default is Auto.
DTMF Detection Tx Mode is available for SIP information and AVT . Options are: Strict or Normal. The default is Strict for which the following are true:
• A DTMF digit requires an extra hold time after detection.
• The DTMF level threshold is raised to -20 dBm.
• The minimum and maximum duration thresholds are:
• strict mode for AVT: 70 ms
• normal mode for AVT: 40 ms
• strict mode for SIP info: 90 ms
• normal mode for SIP info: 50 ms
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Line Tab(s)
DTMF Tx Strict Hold Off
Time:
FAX Process NSE
Hook Flash Tx Method
FAX Disable ECAN
Release Unused Codec
FAX Enable T38
FAX T38 Redundancy
Fax Tone Detect Mode
FAX Tone Detect Mode
Symmetric RTP
This parameter is in effect only when "DTMF Tx Mode" is set to "strict," and when"DTMF Tx Method" is set to out-of-band; i.e. either AVT or SIP-INFO. If a user inadvertently sets the value to less than the default value, the system checks and reverts to the default value. There is no max limit on what the user can set of this parameter. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of dtmf detection, which is needed for interactive voice response systems (IVR).
Default is 90 ms.
To use the fax process NSE feature, select yes. Otherwise, select no.
The default is yes.
Select the method for signaling hook flash events: None, AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16). INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting.
The default is None.
If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no.
The default is no.
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.
To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select
no.
T.38 is supported by the SPA2102 and the SPA8000. The SPA2102 supports a single
T.38 connection. The SPA8000 supports one T.38 connection for each of its four modules (Line 1-2, 3-4, 5-6, and 7-8). The SPA8000 supports a maximum of four connections, but it does not support two fax devices connected to the same module.
The default is yes.
Select the appropriate number.
The default is 1.
If you want the Gateway to detect the fax tone whether the Gateway is a caller or callee, then select caller or callee. If you want the Gateway to detect the fax tone only if the Gateway is the caller, then select caller only. If you want the Gateway to detect the fax tone only if the Gateway is the callee, then select callee only.
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.
(SPA3102 and AG310 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.
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Line Tab(s)
Gateway Accounts (SPA3102/AG310)
Field
Gateway1/2/3/4
GW1/2/3/4 NAT Mapping
Enable
GW1/2/3/4 Auth ID
GW1/2/3/4 Password
Description
The first of 4 gateways that can be specified to be used in the <Dial Plan> to facilitate call routing specification (that overrides the given proxy information). This gateway is represented by gw1 in the <Dial Plan>. For example, the rule
1408xxxxxxx<:@gw1> can be added to the dial plan such that when the user dials
1408+7digits, the call will be routed to Gateway 1. Without the <:@gw1> syntax, all calls are routed to the given proxy by default (except IP dialing).
The default is blank.
If enabled, the ATA uses NAT mapping when contacting Gateway 1.
The default is no.
This is the authentication user-id to be used by the SPA to authenticate itself to
Gateway 1.
The default is blank.
This is the password to be used by the SPA to authenticate itself to Gateway 1.
The default is blank.
VoIP Fallback to PSTN (SPA3102/AG310)
Field
Auto PSTN Fallback
Description
If enabled, the ATA automatically routes all calls to the PSTN gateway when the Line
1 proxy is down (registration failure or network link down).
The default is yes.
Dial Plan
The default dial plan script for each line is as follows: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-
9]xxxxxx|xxxxxxxxxxxx.). The syntax for a dial plan expression is as follows:
Dial Plan Entry
*xx
[3469]11
0
00
[2-9]xxxxxx
1xxx[2-9]xxxxxx xxxxxxxxxxxx.
Functionality
Allow arbitrary 2 digit star code
Allow x11 sequences
Operator
Int’l Operator
US local number
US 1 + 10-digit long distance number
Everything else (Int’l long distance, FWD, ...)
If IP dialing is enabled, one can dial [user-id@]a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and
255, and port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled.
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Line Tab(s)
Field
Dial Plan
Enable IP Dialing
Emergency Number
Description
Dial plan script for this line.
The default is (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-
9]xxxxxxS0|xxxxxxxxxxxx.)
The dial plan syntax is expanded in the SPA3102 and AG310 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 or AG310 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>
Enable or disable IP dialing.
The default is no.
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.
FXS Port Polarity Configuration
Field
Idle Polarity
Caller Conn Polarity
Callee Conn Polarity
Description
Polarity before a call is connected: Forward or Reverse.
The default is Forward.
Polarity after an outbound call is connected: Forward or Reverse.
The default is Forward.
Polarity after an inbound call is connected: Forward or Reverse.
The default is Forward.
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PSTN Line Tab (AG310 and SPA3102)
PSTN Line Tab (AG310 and SPA3102)
This section describes the fields for the following headings on the PSTN Line tab on the
SPA3102 and AG310:
•
”Line Enable” section on page 95
•
”NAT Settings” section on page 109
no in 3102
•
”Network Settings” section on page 110
•
”SIP Settings” section on page 110
•
”Proxy and Registration” section on page 112
•
”Subscriber Information” section on page 113
•
”Audio Configuration” section on page 114
•
”Dial Plans” section on page 116
the rest are not in 3102
•
”VoIP-To-PSTN Gateway Setup” section on page 116
•
”VoIP Users and Passwords (HTTP Authentication)” section on page 117
•
”FXO (PSTN) Timer Values (sec)” section on page 118
•
”PSTN Disconnect Detection” section on page 120
•
”International Control (Settings)” section on page 122
Line Enable
Field
Line Enable
PSTN Contact List
Description
To enable this line for service, select yes. Otherwise, select no.
The default is yes.
Select the appropriate list: None, Phone 1+2, Phone 1, or Phone 2. The default is
Phone1+2.
NAT Settings
Field
NAT Mapping Enable
NAT Keep Alive Enable
Description
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes.
Otherwise, select no.
The default is no.
To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.
The default is no.
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PSTN Line Tab (AG310 and SPA3102)
NAT Keep Alive Msg
NAT Keep Alive Dest
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. Escape sequence of %xx is also accepted. For example, %0d%0a is unescaped into \r\n (CRLF).
The default is $NOTIFY.
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.
Network Settings
Field
SIP ToS/DiffServ Value
SIP CoS Value [0-7]
RTP ToS/DiffServ Value
RTP CoS Value [0-7]
Network Jitter Level
Jitter Buffer Adjustment
Description
TOS/DiffServ field value in UDP IP packets carrying a SIP message.
The default is 0x68.
CoS value for SIP messages.
The default is 3.
ToS/DiffServ field value in UDP IP packets carrying RTP data.
The default is 0xb8.
CoS value for RTP data.
The default is 6.
Determines how jitter buffer size is adjusted by the Linksys ATA device. 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.
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.
SIP Settings
Field
SIP Port
SIP 100REL Enable
EXT SIP Port
Description
Port number of the SIP message listening and transmission port.
The default is 5060.
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.
The external SIP port number.
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PSTN Line Tab (AG310 and SPA3102)
Auth Resync-Reboot
SIP Proxy-Require
SIP Remote-Party-ID
SIP GUID
SIP Debug Option
RTP Log Intvl
Restrict Source IP
If this feature is enabled, the Linksys ATA device 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.
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.
To use the Remote-Party-ID header instead of the From header, select yes.
Otherwise, select no.
The default is yes.
This field is not available with the PAP2T. The Global Unique ID is generated for each line for each device. When it is enabled, the Linksys ATA device 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.
SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows:
• none—No logging.
• 1-line—Logs the start-line only for all messages.
• 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses.
• 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses.
• 1-line excl. REG—Logs the start-line only for all messages except REGISTER requests/responses.
• 1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except
OPTIONS, NOTIFY, and REGISTER requests/responses.
• full—Logs all SIP messages in full text.
• full excl. OPT—Logs all SIP messages in full text except OPTIONS requests/ responses.
• full excl. NTFY—Logs all SIP messages in full text except NOTIFY requests/ responses.
• full excl. REG—Logs all SIP messages in full text except REGISTER requests/ responses.
• full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS,
NOTIFY, and REGISTER requests/responses.
The default is none.
The interval for the RTP log.
If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the PAP2T will drop all packets sent to its SIP Ports originated from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use
Outbound Proxy is yes).
The default is no.
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PSTN Line Tab (AG310 and SPA3102)
Referor Bye Delay
Refer Target Bye Delay
Referee Bye Delay
Refer-To Target Contact
Sticky 183
Controls when the Linksys ATA device sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate period of time in seconds.
The default is 4.
For the Refer Target Bye Delay, enter the appropriate period of time in seconds.
The default is 0.
For the Referee Bye Delay, enter the appropriate period of time in seconds.
The default is 0.
To contact the refer-to target, select yes. Otherwise, select no.
The default is no.
If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no.
The default is no.
Proxy and Registration
Field
Proxy
Use Outbound Proxy
Outbound Proxy
Use OB Proxy In Dialog
Register
Make Call Without Reg
Register Expires
Ans Call Without Reg
Use DNS SRV
Description
SIP proxy server for all outbound requests.
Enable the use of Outbound Proxy. If set to no, the Outbound Proxy parameter and Use
OB Proxy in Dialog is ignored.
The default is no.
SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
Whether to force SIP requests to be sent to the outbound proxy within a dialog.
Ignored if the Use Outbound Proxy parameter is no, or if the Outbound Proxy parameter is empty.
The default is yes.
Enable periodic registration with the Proxy. This parameter is ignored if the Proxy parameter is not specified.
The default is yes.
Allow making outbound calls without successful (dynamic) registration by the unit. If
No, dial tone will not play unless registration is successful.
The default is no.
Allow answering inbound calls without successful (dynamic) registration by the unit.
If proxy responded to REGISTER with a smaller Expires value, the PAP2T will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the PAP2T will retry with the value given in the Min-Expires header in the error response.
The default is 3600.
Expires value in sec in a REGISTER request. PAP2T will periodically renew registration shortly before the current registration expired. This parameter is ignored if the
Register parameter is no. Range: 0 – (231 – 1) sec
Whether to use DNS SRV lookup for Proxy and Outbound Proxy.
The default is no.
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PSTN Line Tab (AG310 and SPA3102)
Field
DNS SRV Auto Prefix
Proxy Fallback Intvl
Proxy Redundancy
Method
Description
If enabled, the PAP2T will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.
The default is no.
This parameter sets the delay (sec) after which the PAP2T will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the PAP2T will not attempt to fall back after a fail over).
The default is 3600
The PAP2T makes an internal list of proxies returned in DNS SRV records. In normal mode this list will contain proxies ranked by weight and priority.
If the parameter Based on SRV port is configured, the PAP2T creates a list in normal mode first, and then inspects the port numbers based on the 1 st proxy’s port on the list.
The default is Normal.
Subscriber Information
Field
Display Name
User ID
Password
Use Auth ID
Auth ID
Call Capacity
Mini Certificate
SRTP Private Key
Description
Display name for caller ID.
Extension number for this line.
Password for this line.
To use the authentication ID and password for SIP authentication, select yes.
Otherwise, select no to use the user ID and password.
The default is no.
Authentication ID for SIP authentication.
Maximum number of calls allowed on this line interface. Choices:
{unlimited,1,2,3,…25 }. Default is 16. Note that the Linksys ATA device does not distinguish between incoming and outgoing calls when talking about call capacity.
Note: unlimited = 16
Base64 encoded of Mini-Certificate concatenated with the 1024-bit public key of the
CA signing the MC of all subscribers in the group.
The default is empty.
Base64 encoded of the 512-bit private key per subscriber for establishment of a secure call.
The default is empty.
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PSTN Line Tab (AG310 and SPA3102)
Audio Configuration
A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the
G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G729a resource is already allocated and since only one G.729a resource is allowed per device, no other low-bit-rate codec may be allocated for subsequent calls; the only choices are G711a and G711u. On the other hand, two
G.723.1/G.726 resources are available per device.
Therefore it is important to disable the use of G.729a in order to guarantee the support of two simultaneous G.723/G.726 codec.
Field
Preferred Codec
Silence Supp Enable
Description
Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u,
G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723.
The default is G711u.
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no.
The default is no.
Use Pref Codec Only
Silence Threshold
G726-32 Enable
To use only the preferred codec for all calls, select yes. (The call fails if the far end does not support this codec.) Otherwise, select no.
The default is no.
Select the appropriate setting for the threshold: high, medium, or low.
The default is medium.
G729a Enable
Echo Canc Enable
G723 Enable To enable the use of the G723a codec at 6.3 kbps, select yes. Otherwise, select no.
The default is yes.
Echo Canc Adapt Enable To enable the echo canceller to adapt, select yes. Otherwise, select no.
The default is yes.
G726-16 Enable To enable the use of the G726 codec at 16 kbps, select yes. Otherwise, select no.
The default is yes.
Echo Supp Enable
To enable the use of the G729a codec at 8 kbps, select yes. Otherwise, select no.
The default is yes.
To enable the use of the echo canceller, select yes. Otherwise, select no.
The default is yes.
G726-24 Enable
To enable the use of the echo suppressor, select yes. Otherwise, select no.
The default is yes.
To enable the use of the G726 codec at 24 kbps, select yes. Otherwise, select no.
The default is yes.
FAX CED Detect Enable To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no.
The default is yes.
To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no.
The default is yes.
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PSTN Line Tab (AG310 and SPA3102)
FAX CNG Detect Enable
G726-40 Enable
FAX Passthru Codec
DTMF Process INFO
FAX Codec Symmetric
DTMF Process AVT
FAX Passthru Method
DTMF Tx Method
FAX Process NSE
Hook Flash Tx Method
FAX Disable ECAN
Release Unused Codec
FAX Enable T38
FAX Tone Detect Mode
To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no.
The default is yes.
To enable the use of the G726 codec at 40 kbps, select yes. Otherwise, select no.
The default is yes.
Select the codec for fax passthrough, G711u or G711a.
The default is G711u.
This field is not available for the PAP2T. To use the DTMF process info feature, select yes. Otherwise, select no.
The default is yes.
To force the Linksys ATA device to use a symmetric codec during fax passthrough, select yes. Otherwise, select no.
The default is yes.
This field is not available for the PAP2T. To use the DTMF process AVT feature, select yes. Otherwise, select no.
The default is yes.
Select the fax passthrough method: None, NSE, or ReINVITE.
The default is NSE.
Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto,
InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends
DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation.
The default is Auto.
To use the fax process NSE feature, select yes. Otherwise, select no.
The default is yes.
Select the method for signaling hook flash events: None, AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16). INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting.
The default is None.
If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no.
The default is no.
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.
To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no.
The default is yes.
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.
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PSTN Line Tab (AG310 and SPA3102)
Symmetric RTP (SPA3102 and AG310 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.
Dial Plans
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.
VoIP-To-PSTN Gateway Setup
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.
VoIP PIN Max Retry
One Stage Dialing
Line 1 VoIP Caller DP
Default VoIP Caller DP
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.
Enable one-stage dialing (applicable if authentication method is none, or HTTP
Digest, or caller is in the Access List).
The default is yes.
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 or AG310 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}
Authentication is skipped for Line 1 VoIP caller.
The default is 1.
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.
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PSTN Line Tab (AG310 and SPA3102)
Field
Line 1 Fallback DP
VoIP Caller ID Pattern
VoIP Access List
VoIP Caller 1/2/3/4/5/6/7/
8 PIN
VoIP Caller 1/2/3/4/5/6/7/
8 DP
Description
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 or AG310 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.
A comma-separated list of caller number templates such that callers with numbers not matching any of these templates are rejected for PSTN gateway service, regardless of the setting of the authentication method. The comparison is applied before the access list is applied. If this parameter is blank (not specified), all callers are considered for PSTN gateway service.
For example: 1408*, 1512???1234.
Note: ‘?’ matches any single digit; ‘*’ matches any number of digits.
The default is blank.
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.
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 parameter is set to PIN.
The default is blank.
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.
VoIP Users and Passwords (HTTP Authentication)
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.
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PSTN Line Tab (AG310 and SPA3102)
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.
Ring Settings
Field
Default Ring
Description
1-8, Follow Line Cfg
FXO (PSTN) Timer Values (sec)
Field
VoIP Answer Delay
PSTN Answer Delay
VoIP PIN Digit Timeout
PSTN PIN Digit Timeout
Description
Delay in seconds before auto-answering inbound VoIP calls for the FXO account. The range is 0-255.
The default is 3.
Delay in seconds before auto-answering inbound PSTN calls after the PSTN starts ringing. The range is 0-255.
The default is 16.
Timeout to wait for the 1 st or subsequent PIN digits from a VoIP caller. The range is 0-
255.
The default is 10.
Timeout to wait for the 1 st or subsequent PIN digits from a PSTN caller. The range is
0-255.
The default is 10.
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PSTN Line Tab (AG310 and SPA3102)
Field
VoIP DLG Refresh Intvl
PSTN Ring Thru Delay
PSTN-To-VoIP Call Max
Dur
VoIP-To-PSTN Call Max
Dur
PSTN Dialing Delay
PSTN Ring Timeout
PSTN Dial Digit Len
PSTN Hook Flash Len
PSTN Ring Thru CWT
Delay
PSTN Ring Timeout
PSTN Dialing Delay
PSTN Dial Digit Len
PSTN Hook Flash Len
Description
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 callleg 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.
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.
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.
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.
Delay after hook before the SPA dials a PSTN number. The range is 0-255.
The default is 1.
Delay after a ring burst before the SPA decides that PSTN ring has ceased. The range is 0-255.
The default is 5.
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.
The length of the hook flash in seconds. During a PSTN-to-VoIP gateway call, the
Linksys ATA processes the out-of-band hook flash signal sent from the VoIP peer through a hook-flash (momentary on-hook signal) on the FXO port. This allows the
VoIP peer to initiate a three-way conference call and subsequent call transfer. The duration of the on-hook signal can be configured using this parameter.
The default is 0.25. The permitted range is limited to 0.02 to 1.6 seconds.
Specify the delay before incoming PSTN calls will ring Line 1 using a Call Waiting
Tone. The default is 3.
Specify the delay after a ring burst before the Gateway decides that the PSTN ring has ended. The default is 5.
Specify the delay after the PSTN phone line is on-hook before the Gateway dials a
PSTN number. The default is 1.
Specify the on/off time when the Gateway transmits digits through the Line (FXO) port. The syntax is on-time/off-time, expressed in seconds with up to two decimal places. The permitted range is 0.05 to 3.00. The default is .1/.1.
Default is .25.
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PSTN Line Tab (AG310 and SPA3102)
PSTN Disconnect Detection
Field
Detect CPC
Detect Polarity Reversal
Detect (PSTN) Long
Silence
Min CPC Duration
Detect Disconnect Tone
Description
CPC is a brief removal of tip-and-ring voltage. If enabled, the SPA will disconnect both call legs when this signal is detected during a gateway call.
The default is yes.
If enabled, SPA will disconnect both call legs when this signal is detected during a gateway call. If it is a PSTN gateway call, the 1st polarity reversal is ignored and the
2 nd one triggers the disconnection. For VoIP gateway call, the 1 st polarity reversal triggers the disconnection.
The default is yes.
If enabled, SPA will disconnect both call legs when the PSTN side has no voice activity for a duration longer than the length specified in the Long Silence Duration parameter during a gateway call
The default is yes.
Specify the minimum duration of a low tip-and-ring voltage (below 1V) for the
Gateway to recognize it as a CPC signal or PSTN line removal. The default is 0.2.
If enabled, SPA will disconnect both call legs when it detects the disconnect tone from the PSTN side during a gateway call. Disconnect tone is specified in the
Disconnect Tone parameter, which depends on the region of the PSTN service.
The default is yes.
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PSTN Line Tab (AG310 and SPA3102)
Field
(PSTN) Long Silence
Duration
Silence Threshold
Disconnect Tone
Description
This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if Detect Long Silence is yes.
The default is 30.
This parameter adjusts the sensitivity of PSTN silence detection. Choose from {very low, low, medium, high, very high}. The higher the setting, the easier to detect silence and hence easier to trigger a disconnection.
The default is medium.
This is the tone script which describes to the SPA the tone to detect as a disconnect tone. The syntax follows a standard Tone Script with some restrictions. Default value is standard US reorder (fast busy) tone, for 4 seconds.
Restrictions:
• 2 frequency components must be given. If single frequency is desired, the same frequency is used for both
• The tone level value is not used. –30 (dBm) should be used for now.
• Only 1 segment set is allowed
• Total duration of the segment set is interpreted as the minimum duration of the tone to trigger detection
• 6 segments of on/off time (seconds) can be specified. A 10% margin is used to validated cadence characteristics of the tone.
The Disconnect Tone Script and Impedance value for various countries follow:
US—480@-30,620@-30;4(.25/.25/1+2) —Impedance: 600
UK—400@-30,400@-30; 2(3/0/1+2) —Impedance: 370+620
France—440@-30,440@-30; 2(0.5/0.5/1+2) —Impedance: 270+750||150nF
Germany—425@-10; 10(0.48/0.48/1) —Impedance:220+820||120nF
Netherlands—425@-30,425@-30; 2(0.5/0.5/1+2) —Impedance: 600
Sweden—425@-10; 10(0.25/0.25/1) —Impedance:600
Norway—425@-10; 10(0.5/0.5/1) —Impedance: 600
Italy—425@-30,425@-30; 2(0.2/0.2/1+2)— Impedance: 220+820||120nF
Spain—425@-10; 10(0.2/0.2/1,0.2/0.2/1,0.2/0.6/1) —Impedance: 220+820||120nF
Portugal—425@-10; 10(0.5/0.5/1)— Impedance:220+820||120nF
Poland—425@-10; 10(0.5/0.5/1)— Impedance: n/a
Denmark—425@-10; 10(0.25/0.25/1)— Impedance: 600
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PSTN Line Tab (AG310 and SPA3102)
International Control (Settings)
Field
FXO Port Impedance
Description
Desired impedance of the FXO Port. Choose from {600, 900, 370+620,
270+750||150nF, 220+820||120nF, 370 + 620 || 310nf, 320 + 1050 || 230nf, 370 +
820 || 110 nf, 275 + 780 || 115nf, 120 + 820 || 110nf, 350 + 1000 || 210nf, 0 + 900 ||
130nf }
The default is 600.
The Disconnect Tone Script and Impedance values for various countries follos:
US—480@-30,620@-30;4(.25/.25/1+2) —Impedance: 600
UK—400@-30,400@-30; 2(3/0/1+2) —Impedance: 370+620
France—440@-30,440@-30; 2(0.5/0.5/1+2) —Impedance: 270+750||150nF
Germany—425@-10; 10(0.48/0.48/1) —Impedance:220+820||120nF
Netherlands—425@-30,425@-30; 2(0.5/0.5/1+2) —Impedance: 600
Sweden—425@-10; 10(0.25/0.25/1) —Impedance:600
Norway—425@-10; 10(0.5/0.5/1) —Impedance: 600
Italy—425@-30,425@-30; 2(0.2/0.2/1+2)— Impedance: 220+820||120nF
Spain—425@-10; 10(0.2/0.2/1,0.2/0.2/1,0.2/0.6/1) —Impedance: 220+820||120nF
Portugal—425@-10; 10(0.5/0.5/1)— Impedance:220+820||120nF
Poland—425@-10; 10(0.5/0.5/1)— Impedance: n/a
Denmark—425@-10; 10(0.25/0.25/1)— Impedance: 600
Ring Frequency Min
SPA To PSTN Gain
Ring Frequency Max
PSTN To SPA Gain
Specify the lower limit of the ring frequency used to detect the ring signal. The default is 10.
dB of digital gain (or attenuation if negative) to be applied to the signal sent from the
SPA to the PSTN side. The range is -15 to 12.
The default is 0.
Specify the higher limit of the ring frequency used to detect the ring signal. The default is 100.
dB of digital gain (or attenuation if negative) to be applied to the signal sent from the
PSTN side to the SPA. The range is -15 to 12.
The default is 0.
Ring Validation Time
Tip/Ring Voltage Adjust
Operational Loop Current
Min
Choices for mA are: {10, 12, 14, 16).
The default is 10.
On-Hook Speed
Current Limiting Enable
Choose from {Less than 0.5ms, 3ms (ETSI), 26ms (Australia)}.
The default is Less than 0.5ms.
Enable or disable current limiting.
The default is no.
Ring Frequency Min
Ring Frequency Max
Specify the minimum signal duration required by the Gateway for recognition as a ring signal. The default is 256 ms.
Choices are {3.1, 3.2, 3.35, 3.5}.
The default is 3.5.
Minimum ring frequency to detect. The range is 5-100.
The default is 10.
Maximum ring frequency to detect. The range is 5-100.
The default is 100.
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PSTN Line Tab (AG310 and SPA3102)
Field
Ring Validation Time
Ring Indication Delay
Ring Timeout
Ring Threshold
Ringer Impedance
Line-In-Use Voltage
Description
Choose from {100, 150, 200, 256, 384, 512, 640, 1024} (ms).
The default is 256ms.
Choose from {0, 512, 768, 1024, 1280, 1536, 1792} (ms).
The default is 512ms.
Choose from {0, 128, 256, 384, 512, 640, 768, 896, 1024, 1152, 1280, 1408, 1536, 1664,
1792, 1920} (ms).
The default is 640 ms.
Choose from {13.5–16.5, 19.35–2.65, 40.5–49.5} (Vrms).
The default is 13.5-16.5 Vrms.
Choose from {High, Synthesized(Poland, S.Africa, Slovenia)}.
The default is high.
Determines the voltage threshold at which the SPA-3000 assumes the PSTN is in use by another handset sharing the same line (and will declare PSTN gateway service not available to incoming VoIP callers).
The default value is 40v.
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User Tab(s)
User Tab(s)
This section describes the fields for the following headings on the User 1 and User 2 tabs:
•
”Call Forward Settings” section on page 124
•
”Selective Call Forward Settings” section on page 125
•
”Speed Dial Settings” section on page 125
•
”Supplementary Service Settings” section on page 125
•
”Distinctive Ring Settings” section on page 126
•
”Ring Settings” section on page 127
NOTE: For the SPA8000, the settings on this page occur on each Line tab ([1] to [8]).
When a call is made from Line 1 or Line 2, Linksys ATA device shall use the user and line settings for that Line; there is no user login support in Linksys ATA device v1.0. Per user parameter tags must be appended with [1] or [2] (corresponding to line 1 or 2) in the configuration profile. It is omitted below for readability.
Call Forward Settings
Field
Cfwd All Dest
Cfwd Busy Dest
Cfwd No Ans Dest
Cfwd No Ans Delay
Description
Forward number for Call Forward All Service
In addition to normal call forward destination as used in the other ATAs, on the
SPA3102 or AG310, you can specify the following additional parameters: gw0 – forward the caller to use the PSTN gateway
<pstn-number>@gw0 – forward to caller to the PSTN number (dialed automatically by the SPlocalA through the PSTN gateway)
The default is blank.
Forward number for Call Forward Busy Service. Same as Cfwd All Dest.
The default is blank.
Forward number for Call Forward No Answer Service. Same as Cfwd All Dest.
In addition to normal call forward destination as used in the other ATAs, on the
SPA3102 or AG310, you can specify the following additional parameters: gw0 – forward the caller to use the PSTN gateway
<pstn-number>@gw0 – forward to caller to the PSTN number (dialed automatically by the SPA through the PSTN gateway)
The default is blank.
Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest.
The default is 20.
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User Tab(s)
Selective Call Forward Settings
Field
Cfwd Sel1- 8 Caller
Cfwd Sel1 - 8 Dest
Block Last Caller
Accept Last Caller
Cfwd Last Caller
Cfwd Last Dest
Description
Caller number pattern to trigger Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8.
The default is blank.
Forward number for Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8.
Same as Cfwd All Dest.
The default is blank.
ID of caller blocked via the Block Last Caller service.
The default is blank.
ID of caller accepted via the Accept Last Caller service.
The default is blank.
The Caller number that is actively forwarded to Cfwd Last Dest by using the Call
Forward Last activation code
The default is blank.
Forward number for the Cfwd Last Caller parameter.
Same as Cfwd All Dest.
The default is blank.
Speed Dial Settings
This section does not apply to the WIP310 wireless phone.
Field
Speed Dial 2-9
Description
Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9.
The default is blank.
Supplementary Service Settings
The Linksys ATA device provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service. A supplementary service should be disabled if a) the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the Linksys ATA device.
Field
CW Setting
Block CID Setting
Block ANC Setting
Description
Call Waiting on/off for all calls.
The default is yes.
Block Caller ID on/off for all calls.
The default is no.
Block Anonymous Calls on or off.
The default is no.
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User Tab(s)
Field
DND Setting
CID Setting
CWCID Setting
Dist Ring Setting
Secure Call Setting
Message Waiting
Accept Media Loopback
Request
Media Loopback Mode
Media Loopback Type
Description
DND on or off.
The default is no.
Caller ID Generation on or off.
The default is yes.
Call Waiting Caller ID Generation on or off.
The default is yes.
Distinctive Ring on or off.
The default is yes.
If yes, all outbound calls are secure calls by default.
The default is no.
This is updated when there is voicemail notification received by the Linksys ATA device. The user can also manually modify it to clear or set the flag. Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle.
The default is no.
Controls how to handle incoming requests for loopback operation. Choices are:
Never, Automatic, and Manual, where:
• never—never accepts loopback calls; reply 486 to the caller
• automatic—automatically accepts the call without ringing
• manual —rings the phone first, and the call must be picked up manually before loopback starts.
The default is Automatic.
The loopback mode to assume locally when making call to request media loopback.
Choices are: Source and Mirror. Default is Source.
Note that if the Linksys ATA device answers the call, the mode is determined by the caller.
The loopback type to use when making call to request media loopback operation.
Choices are Media and Packet. Default is Media.
Note that if the Linksys ATA device answers the call, then the loopback type is determined by the caller (the Linksys ATA device always picks the first loopback type in the offer if it contains multiple types.)
Distinctive Ring Settings
Caller number patterns are matched from Ring 1 to Ring 8. The first match (not the closest match) will be used for alerting the subscriber.
Field
Ring1 - 9 Caller
Description
Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, 8, or 9.
The default is blank.
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User Tab(s)
Ring Settings
Field
Default Ring
Default CWT
Hold Reminder Ring
Call Back Ring
Cfwd Ring Splash Len
Cblk Ring Splash Len
VMWI Ring Splash Len
VMWI Ring Policy
Ring On No New VM
Description
Default ringing pattern, 1 – 8, for all callers.
The default is 1.
Default CWT pattern, 1 – 8, for all callers.
The default is 2.
Ring pattern for reminder of a holding call when the phone is on-hook.
The default is None.
Ring pattern for call back notification.
The default is None.
Duration of ring splash when a call is forwarded
(0 – 10.0s).
The default is 0.
Duration of ring splash when a call is blocked (0 – 10.0s).
The default is 0.
Duration of ring splash when new messages arrive before the VMWI signal is applied
(0 – 10.0s).
The default is .5.
The parameter controls when a ring splash is played when a the VM server sends a
SIP NOTIFY message to the Linksys ATA device indicating the status of the subscriber’s mail box. 3 settings are available:
• New VM Available—ring as long as there is 1 or more unread voicemail
• New VM Becomes Available—ring when the number of unread voicemail changes from 0 to non-zero
• New VM Arrives—ring when the number of unread voicemail increases.
The default is New VM Available.
If enabled, the Linksys ATA device will play a ring splash when the VM server sends
SIP NOTIFY message to the Linksys ATA device indicating that there are no more unread voicemails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp.
The default is no.
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PSTN User Tab (AG310 and SPA3102)
PSTN User Tab (AG310 and SPA3102)
This section describes the fields for the following headings on the PSTN User tab on the
SPA3102 and AG310:
•
”PSTN-To-VoIP Selective Call Forward Settings” section on page 128
•
”PSTN-To-VoIP Speed Dial Settings” section on page 128
•
”PSTN Ring Thru Line 1 Distinctive Ring Settings” section on page 128
•
”PSTN Ring Thru Line 1 Ring Settings” section on page 128
PSTN-To-VoIP Selective Call Forward Settings
Field
Cfwd Sel1-8 Caller
Cfwd Sel1-8 Dest
Description
Eight PSTN Caller Number Patterns to be blocked for VoIP gateway services or forwarded to a certain VoIP number. If the caller is blocked, the SPA will not autoanswers the call.
Eight VoIP destinations to forward a PSTN caller matching the Cfwd Sel x Caller
parameter. If this entry is blank, the PSTN caller is blocked for VoIP service.
PSTN-To-VoIP Speed Dial Settings
Field
Speed Dial 1-9
Description
The VoIP number to call when the PSTN caller dials a single digit ‘2’
PSTN Ring Thru Line 1 Distinctive Ring Settings
Field
Ring1-8 Caller
Description
Eight PSTN Caller Number Patterns such that the corresponding ring will be used to ring through Line 1 if the PSTN caller matches this pattern.
PSTN Ring Thru Line 1 Ring Settings
Field
Default Ring
Description
The default ring to be used to ring through Line 1. Choose from
{1,2,3,4,5,6,7,8,Follow Line 1}. If Follow Line 1 is selected, the ring to be used is determined by Line 1’s distinctive ring settings.
The default is 1.
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Key Features
- Analog phone connectivity
- IP network access
- Support for multiple voice codecs
- Call waiting, call forwarding, and conference calling
- Built-in router (for some models)
- QoS support (for some models)
- PSTN connectivity (for some models)
