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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
How PSTN-To-VoIP Calls Work
How PSTN-To-VoIP Calls Work
PSTN-To-VoIP calls can be made with two-stage dialing only. The only authentication method available
is the PIN method.
The SPA3102 takes the FXO port off hook after a configurable number of rings. If PIN Authentication
is enabled, it prompts the caller to enter the PIN number followed by a # key. The Inter-PIN-digit timeout
is set at 10 seconds. Up to eight PSTN PIN numbers can be configured in the SPA3102. If the given PIN
does not match any of the PSTN PIN values, the SPA3102 plays the reorder tone to the FXO port for up
to 10 seconds, and then takes the FXO port on-hook. If the given PIN matches one of PSTN PIN values,
the SPA3102 plays dial tone to the FXO port and is ready to accept digits for the target VoIP number
from the PSTN caller. The collected digits are processed by the dial plan associated with the PIN
number.
Note
If Authentication is disabled, a default dial plan is used for all PSTN callers.
Terminating Gateway Calls
There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call leg. A gateway call
is terminated when either call leg is ended. The SPA3102 takes the FXO port on-hook when the call
terminates so the PSTN line can be used again. The SPA3102 detects that the PSTN call leg is ended
when one of the following conditions occurs during a call:
The PSTN Line voltage drops to a very low value (this occurs if the line is disconnected from the
PSTN service or if the PSTN switch provides a CPC signal)
A polarity reversal or disconnect tone is detected at the FXO port
When there is no voice activity for a configurable period of time in either direction at the FXO port
When any of the above occurs, the SPA3102 takes the FXO port on hook and sends a BYE request to
end the VoIP call leg. On the other hand, when the SPA3102 receives a SIP BYE from the VoIP during
a call, it takes the FXO port on hook to end the PSTN call leg.
In addition, the SPA3102 can also send a refresh signal periodically to the VoIP call leg to determine
whether the call leg is still up. If a refresh operation fails, the SPA3102 ends both call legs.
Table 1-29
lists parameters for terminating gateway call parameters.
Table 1-28
Two-Stage Dialing
Parameter
Group
Description
Values
VoIP Caller
1/2/3/4/5/6/7/8
PIN
PSTN
Line
The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, or 8.
31-character string
VoIP Caller
1/2/3/4/5/6/7/8
DP
PSTN
Line
Specifies which dial plan to be used for this
VoIP caller. If 0, dial plan processing is
disabled; the given target number is dialed to
the PSTN as is.
Choice of 1 to 8
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
How PSTN-To-VoIP Calls Work
VoIP Outbound Call Routing
Calls made from Line 1 are routed through the configured Line 1 service provider, by default. You can
override this behavior by IP dialing, through which the calls can be routed to any IP address entered by
the user. SPA3102 allows flexible call routing with four sets of gateway parameters and configurable
dial plans.
Table 1-30
lists VoIP outbound call routing parameters.
Table 1-29
Terminating Gateway Call Parameters
Parameter
Group
Description
Values
Detect CPC:
PSTN
Line
If yes, SPA3102 detects CPC as a disconnect signal.
Default = Yes
Yes or No
Detect Long Silence:
PSTN
Line
If yes, SPA3102 detects prolonged silence period as
a disconnect signal. Default = Yes
Yes or No
Long Silence Duration:
PSTN
Line
The minimum duration of continuous silence before
the SPA3102 disconnects the call, if <Detect Long
Silence> is enabled. Default = 30 (s)
10-255
Disconnect Tone:
PSTN
Line
Tone Script of the disconnect tone to detect. Default
= “480@-30,620@-30;4(.25/.25/1+2)”. Note:
The SPA3102 supports two frequency components.
If the tone has only one frequency, use the same
value for both frequencies.
Each cadence segment must have the same
frequency.
The level value is the threshold to detect each tone.
The total duration is the minimum duration of the
tone to be recognized as the disconnect tone
ToneScript
Detect Polarity
Reversal:
PSTN
Line
If yes, SPA3102 interprets polarity reversal as a
disconnect signal.
On an inbound PSTN call, SPA3102 disconnects on
the first polarity reversal. On an outbound PSTN
call, SPA3102 disconnects on the second polarity
reversal (because the first polarity reversal indicates
the outbound call is connected). Default = Yes
Yes or No
Detect Disconnect
Tone:
PSTN
Line
If yes, SPA3102 interprets the disconnect tone as
specified in <Disconnect Tone> as the disconnect
signal. Default = Yes
Yes or No
Silence Threshold:
PSTN
Line
This is the signal energy threshold. Below this
threshold is considered silence. Default = medium
very low,
low,
medium,
high, very
high
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
How PSTN-To-VoIP Calls Work
Gateways 1 to 4 can be specified in a dial plan with the special identifier gw1, gw2, gw3, or gw4. Also,
gw0 represents the internal PSTN gateway via the FXO port. You can specify in the dial plan to use gw
x
(
x
= 0,1,2,3,4) when making certain calls. In general, you can specify any gateway address in the dial
plan. In addition, three parameters are added that can be used with call routing:
usr—User-id used for authentication with the given gateway
pwd—Password used for authentication with the given gateway
nat—Enable or disable NAT mapping when calling the gateway
Table 1-31
lists some examples.
You can set up multiple PSTN gateways at different locations and configure Line 1 to use a different
gateway when dialing specific numbers.
Table 1-30
VoIP Outbound Call Routing Parameters
Parameters
Group
Description
Values
Gateway 1
Line 1
Fully qualified domain name (or IP address) of a
gateway. If the port number is not specified, 5060 is
assumed. Default value is [blank]
Domain name or IP
address
GW1 Nat
Mapping
Enable
Line 1
Whether to enable NAT mapping when using
Gateway 1. Default is “no”.
Yes or No
GW1 User
ID
Line 1
The authentication user name when using Gateway 1.
Default is [blank]
31-character string
GW2
Password
Line 1
The authentication password when using Gateway 1.
Default is [blank]
31-character string
Similar for
GW 2, 3,
and 4
Line 1
Similar for GW 2, 3, and 4
Table 1-31
Specifying Gateway Addresses
Example
Description
<9,:>xx.<:@gw1
Dial 9 to start outside dial tone, followed by one or more
digits, and route the call to Gateway 1.
[93]11<:@gw0>
Route 911 and 311 calls to the local PSTN gateway
<8,:1408>xxxxxxx<:@pstn.Linksys.c
om:5061;usr=joe;pwd=joe_pwd;nat>
Dial 8 to start outside dial tone, prepend 1408 followed by
seven digits, and route the call to pstn.Linksys.com:5061,
with user-id = joe, and pwd = bell_pwd, and enable NAT
mapping
<8,:1408>xxxxxxx<:@gw2:5061;usr
=”Alex
Bell”;pwd=”anything”;nat=no>
Dial 8 to start outside dial tone, prepend 1408 followed by
seven digits, and route the call to Gateway 2, but use the
given port, user-id, and password, and no
pstn.Linksys.com:5061, and with user-id = “Alex Bell” and
pwd = bell_pwd, and disable NAT mapping
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Chapter 4
Configuring the PSTN Gateway (FXO)
Configuring VoIP Failover to PSTN
Configuring VoIP Failover to PSTN
When power is disconnected from the SPA3102, the FXS port is connected to the FXO port. In this case,
the telephone attached to the FXS port is electrically connected to the PSTN service via the FXO port.
When power is applied to the SPA3102, the FXS port is disconnected from the FXO port. However, if
the PSTN line is in use when the power is applied to the SPA3102, the relay is not flipped until the PSTN
line is released. This is done so that the SPA3102 does not interrupt any call in progress on the PSTN
line.
When Line 1 VoIP service is down (because of registration failure or loss of network link), SPA3102
can be configured to automatically route all outbound calls to the internal gateway using the parameters
listed in
Table 1-32
.
Sharing One VoIP Account Between the FXS and PSTN Lines
The FXS (Line 1) and FXO (PSTN Line) can share a single VoIP account if they use different SIP ports.
If the service provider allows multiple registration contacts and simultaneous ringing, both lines can
register periodically with the service provider. In this case, both lines receive inbound calls to this VoIP
account. The PSTN Line should be configured with a sufficiently long answer delay before the call is
automatically answered to allow for the function of the PSTN gateway.
If the service provider does not allow more than one register contact, the PSTN Line should not register.
In this case, only Line 1 rings on the inbound call to this VoIP account because it is the only line
registered with the service provider.
Line 1 can have the call forwarded to the PSTN Line after a few seconds using the
Call-Forward-On-No-Answer feature with gw0 as the forward destination. Similarly, Line 1 can apply
Call-Forward-All, Call-Forward-On-Busy, and Call-Forward-Selective feature, and direct the caller to
the PSTN-Gateway.
Only PIN authentication is allowed when a VoIP caller is forwarded to the PSTN-gateway from Line 1.
If HTTP Authentication is used, the caller is not authenticated.
Another option when using the Forward-To-GW0 feature is to forward the caller to a specific PSTN
number, using the syntax <PSTN-number>@gw0 in the forward destination. When using this with
Call-Forward-Selective, you can develop some interesting applications. For example, you can forward
all callers with 408 area code to 14081234567, or all callers with 800 area code to 18005558355 (This
is the number for Tell Me). When this syntax is used, authentication is not used and the target PSTN
number is automatically dialed by the SPA3102 after the caller is forwarded to gw0.
Other Options
This section describes other options provided by the SPA3102. It includes the following topics:
PSTN Call to Ring Line 1, page 4-8
Table 1-32
Automatically Routing Outbound Calls to Internal Gateway
Parameter
Group
Description
Range
Auto PSTN Fallback
Line 1
If enabled, SPA3102 automatically routes
outbound calls to Gateway 0 when registration
fails or network link is down.
Bool
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Chapter 4
Configuring the PSTN Gateway (FXO)
Call Scenarios
Symmetric RTP, page 4-8
Call Progress Tones, page 4-8
PSTN Call to Ring Line 1
This feature allows a PSTN caller to ring Line 1. When the PSTN line rings, the PSTN Line makes a
local VoIP call to Line 1. If Line 1 is busy, it stops. After a given number of rings, the VoIP gateway
picks up the call.
Symmetric RTP
Symmetric RTP is used to send audio RTP to the source IP and port of the inbound RTP packets. This
facilitates NAT traversal.
Table 1-33
lists symmetric RTP parameters.
Call Progress Tones
Call Scenarios
This section describes some typical scenarios where the SPA3102 can be applied. Some terms are
introduced in the first few sections and reused in later sections. This section includes the following
topics:
Table 1-33
Symmetric RTP Parameters
Parameter
Group
Description
Range
Symmetric RTP
Line 1
Enable symmetric RTP operation. If enabled, SPA3102
sends RTP packets to the source address of the last
received valid inbound RTP packet. If disabled,
SPA3102 sends RTP to the destination as indicated in
the inbound SDP. Default is yes.
Yes or
No
Symmetric RTP
PSTN
Line
Same as above. Default is yes.
Yes or
No
Table 1-34
Call Progress Tones
Call Progress Tone
Description
VoIP PIN Tone
This tone is played to prompt a VoIP caller to enter a PIN number.
PSTN PIN Tone
This tone is played to prompt a PSTN caller to enter a PIN number.
Outside Dial Tone
During two-stage PSTN-gateway dialing and with a dial plan assigned, the
SPA3102 collects digits from the VoIP caller and processes the number
using the dial plan. The SPA3102 plays the <Outside Dial Tone> to prompt
the VoIP caller to enter the PSTN number. This tone should be specified to
sound different from the PSTN dial tone.

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