5
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Connecting to PSTN and VoIP Services
Configuring the PSTN (FXO) Gateway
(AG310 and SPA3102)
This chapter describes how to configure the PSTN gateway provided by Linksys ATAs devices with one or more FXO ports, which includes the AG310 and SPA3102 devices. It includes the following sections:
•
”Connecting to PSTN and VoIP Services” section on page 46
•
”How VoIP-To-PSTN Calls Work” section on page 47
•
”How PSTN-To-VoIP Calls Work” section on page 49
•
”Configuring VoIP Failover to PSTN” section on page 51
•
”Sharing One VoIP Account Between the FXS and PSTN Lines” section on page 52
•
”Other Options” section on page 53
•
”Call Scenarios” section on page 54
Connecting to PSTN and VoIP Services
The SPA3102 and AG310 devices have the following ports for connection to telephony devices:
• FXS port (Phone)—Connect to a standard analog telephone or fax machine, configured using the Line tab.
• FXO port (Line)—Connect to a standard telephone wall jack for connectivity to the
PSTN, configured using the PSTN Line tab.
Line 1 does not provide a gateway because it provides only VoIP service. The VoIP-To-PSTN calling function is referred to as a PSTN gateway, and the PSTN-To-VoIP calling function is referred to as a VoIP gateway. Note the following definitions:
• VoIP caller—One who calls the Linksys ATA device via VoIP to obtain PSTN service
• VoIP user—VoIP caller that has a user account (user-id and password) on the Linksys ATA device
• PSTN caller—One who calls the Linksys ATA device from the PSTN to obtain VoIP service
Line 1 can be configured with a regular VoIP account and can be used in the same way as the
Line 1 of any Linksys ATA.
With the SPA3102 and AG310 devices, a second VoIP account can be configured to support
PSTN gateway calls exclusively. A different SIP port should be assigned to Line 1 and the PSTN
Line. The same VoIP account may be used for both Line 1 and the PSTN Line if a different SIP port is assigned to each.
Linksys ATA Administration Guide 46
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
How VoIP-To-PSTN Calls Work
VoIP callers can be authenticated by one of the following methods:
• No Authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
• HTTP digest—SIP INVITE must contain a valid authorization header.
PSTN callers can be authenticated by one of the following methods:
• No authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
How VoIP-To-PSTN Calls Work
To obtain PSTN services through the SPA3102 or AG310 devices, the VoIP caller establishes a connection with the PSTN Line by way of a standard SIP INVITE request addressed to the PSTN
Line. The PSTN Line can be configured to support one-stage and two-stage dialing as described in the following sections.
One-Stage Dialing
One-stage dialing allows a call to be started over VoIP and then immediately get a dial tone on the PSTN.
To use one-stage dialing, the Request-URI of the INVITE to the PSTN Line should have the form
<Dialed-Number>@<SPA-Address>, where <Dialed-Number> is the number dialed by the VoIP caller, and <SPA-Address> is a valid address of the SPA3102 or AG310 device, such as
10.0.0.100:5061.
If the FXO port is currently in use (off-hook) or the PSTN line is being used by another extension, the Linksys ATA device replies to the INVITE with a 503 response. Otherwise, it compares the
<Dialed-Number> with the User ID parameter of the PSTN Line. If they are the same, the Linksys
ATA device interprets this as a request for two-stage dialing (see the
”Two-Stage Dialing” section on page 48
). If they are different, the Linksys ATA device processes the <Dialed-Number> using the corresponding <Dial Plan>.
If dial plan processing fails, the Linksys ATA device replies with a 403 response. Otherwise, it replies with a 200 and at the same time takes the FXO port off hook and dials the target number returned after processing the dial plan.
NOTE: If the User ID parameter on the PSTN Line is blank, the Register parameter should be disabled for the PSTN Line.
If HTTP Digest Authentication is enabled, the Linksys ATA device challenges the INVITE with a
401 response if it does not have a valid Authorization header. The Authorization header should include a <User ID
n
> parameter, where n refers to one of eight VoIP user accounts that can be configured on the Linksys ATA device. The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The <User ID
n
> parameter must
Linksys ATA Administration Guide 47
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
How VoIP-To-PSTN Calls Work
match one of the VoIP accounts stored on the Linksys ATA device. Each VoIP user account contains the information listed below.
Parameter
User ID 1/2/3/4/
5/6/7/8
Password 1/2/3/
4/5/6/7/8
User 1/2/3/4/5/
6/7/8 DP
Tab Description
PSTN Line The username value.
PSTN Line The password value.
Values
31-character string
31-character string
PSTN Line Specifies the dial plan to be used for this VoIP user. If
0, dial plan processing is disabled; the given target number is dialed to the PSTN as is.
Choice of 0-8
NOTE: If Authentication is disabled, a default dial plan is used for all unknown VoIP users.
Two-Stage Dialing
In two-stage dialing, the Linksys ATA device takes the FXO port off-hook but does not automatically dial any digits after accepting the call. To invoke two-stage dialing, the VoIP caller should INVITE the PSTN Line without the user-id in the Request-URI or with a user-id that matches exactly the <User ID
n
> of the PSTN Line. A different user-id in the Request-URI is treated as a request for one-stage dialing if one-stage dialing is enabled, or dropped by the
Linksys ATA device (as if no user-id is given) if one-stage dialing is disabled.
NOTE: If Authentication is disabled, a default dial plan is assigned to all VoIP callers.
HTTP Digest Authentication can be also used for two-stage dialing, as in one-stage dialing. If using HTTP Digest Authentication or Authentication is disabled, the VoIP caller should hear the
PSTN dial tone right after the call is answered (by a SIP 200 response).
If PIN Authentication is enabled, the VoIP caller is prompted to enter a PIN number after the
Linksys ATA device answers the call. The PIN number must end with a # key. The inter-PIN-digit timeout is 10 seconds (not configurable). Up to eight VoIP caller PIN numbers can be configured on the Linksys ATA device. A dial plan can be selected for each PIN number. If the caller enters a wrong PIN or the Linksys ATA device times out waiting for more PIN digits, the
Linksys ATA device tears down the call immediately with a BYE request.
NOTE: When the source address of the INVITE is 127.0.0.1, authentication is automatically disabled because this is a call by the local user. This applies to both one-stage and two-stage dialing.
The following table lists the parameters used in two-stage dialing.
Parameter
VoIP Caller 1/2/3/4/
5/6/7/8 PIN
VoIP Caller 1/2/3/4/
5/6/7/8 DP
Tab
PSTN
Line
PSTN
Line
Description
The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, or 8.
Values
31-character string
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
Linksys ATA Administration Guide 48
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
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 Linksys ATA device 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 Linksys ATA device. If the given PIN does not match any of the PSTN PIN values, the Linksys ATA device 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 Linksys
ATA device 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. When the call terminates, the FXO port goes onhook so the PSTN line can be used again. The Linksys ATA device 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.
• There is no voice activity for a configurable period of time in either direction at the FXO port.
When any of the above conditions occur, the Linksys ATA device takes the FXO port on hook and sends a BYE request to end the VoIP call leg. On the other hand, when the Linksys ATA device 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 Linksys ATA device 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 Linksys ATA device ends both call legs.
The following table lists parameters for terminating gateway calls.
Parameter
Detect CPC
Detect Long Silence
Tab
PSTN
Line
PSTN
Line
Description
If yes, the Linksys ATA device detects CPC as a disconnect signal.
If yes, the Linksys ATA device detects prolonged silence period as a disconnect signal.
Values
Yes or No
The default is
Yes.
Yes or No
Linksys ATA Administration Guide 49
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
How PSTN-To-VoIP Calls Work
Long Silence Duration:
Disconnect Tone:
Detect Polarity Reversal:
PSTN
Line
PSTN
Line
PSTN
Line
Detect Disconnect Tone: PSTN
Silence Threshold:
Line
PSTN
Line
The minimum duration of continuous silence before the
Linksys ATA device disconnects the call, if the Detect (PSTN)
Long Silence parameter is enabled.
10-255
The default is
30(s).
Tone Script of the disconnect tone to detect. The Linksys
ATA device 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
The default is
480@-
30,620@-
30;4(.25/.25/
1+2)”
If yes, the Linksys ATA device interprets polarity reversal as a disconnect signal.
On an inbound PSTN call, Linksys ATA device disconnects on the first polarity reversal. On an outbound PSTN call,
Linksys ATA device disconnects on the second polarity reversal (because the first polarity reversal indicates the outbound call is connected).
If yes, the Linksys ATA device interprets the disconnect tone as specified in the Disconnect Tone parameter as the disconnect signal.
Yes or No
The default is
Yes.
This is the signal energy threshold. Below this threshold is considered silence.
Yes or No
The default is
Yes.
very low, low, medium, high, very high
The default is
Medium.
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. The Linksys ATA device allows flexible call routing with four sets of gateway parameters and configurable dial plans. The following table lists VoIP outbound call routing parameters.
Parameter
Gateway 1
GW1 Nat
Mapping Enable
GW1 User ID
GW2 Password
Tab
Line 1
Line 1
Line 1
Line 1
Description
Fully qualified domain name (or IP address) of a gateway. If the port number is not specified, 5060 is assumed.
Values
Domain name or IP address.
The default is blank.
Whether to enable NAT mapping when using
Gateway 1.
The authentication user name when using Gateway
1.
Yes or No
The default is no.
31-character string
The default is blank.
The authentication password when using Gateway 1. 31-character string.
The default is blank.
Linksys ATA Administration Guide 50
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Configuring VoIP Failover to PSTN
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 gwx (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
The following table lists some examples.
Example
<9,:>xx.<:@gw1
[93]11<:@gw0>
<8,:1408>xxxxxxx<:@pstn.Linksys.com:506
1;usr=joe;pwd=joe_pwd;nat>
<8,:1408>xxxxxxx<:@gw2:5061;usr=”Alex
Bell”;pwd=”anything”;nat=no>
Description
Dial 9 to start outside dial tone, followed by one or more digits, and route the call to Gateway 1.
Route 911 and 311 calls to the local PSTN gateway
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
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
You can set up multiple PSTN gateways at different locations and configure Line 1 to use a different gateway when dialing specific numbers.
Configuring VoIP Failover to PSTN
When power is disconnected from the SPA3102 or AG310 device, 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 Linksys ATA device, the FXS port is disconnected from the FXO port. However, if the PSTN line is in use when the power is applied to the Linksys ATA device, the relay is not flipped until the PSTN line is released. This is done so that the Linksys ATA device 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), the
Linksys ATA device can be configured to automatically route all outbound calls to the internal gateway using the parameter listed below.
Parameter
Auto PSTN Fallback
Tab
Line 1
Description
If enabled, the Linksys ATA device automatically routes outbound calls to Gateway 0 when registration fails or network link is down.
Value
The default is
yes.
Linksys ATA Administration Guide 51
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Sharing One VoIP Account Between the FXS and PSTN Lines
Sharing One VoIP Account Between the FXS and PSTN
Lines
Both the FXS (Line 1) and FXO (PSTN Line) can to receive incoming calls for a single VoIP account if they are different ports. Consider the following:
• 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.
• When using the Forward-To-GW0 feature, you can 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 Linksys ATA device after the caller is forwarded to gw0.
Linksys ATA Administration Guide 52
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Other Options
Other Options
This section describes other options provided by the Linksys ATA device. It includes the following topics:
•
”PSTN Call to Ring Line 1” section on page 53
•
”Symmetric RTP” section on page 53
•
”Call Progress Tones” section on page 53
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
The Symmetric RTP parameter is used to send audio RTP to the source IP and port of the inbound RTP packets. This facilitates NAT traversal.
The following table lists symmetric RTP parameters.
Parameter
Symmetric RTP
Symmetric RTP
Tab Description Values
Line 1 Enable symmetric RTP operation. If enabled, the Linksys ATA device sends RTP packets to the source address of the last received valid inbound RTP packet. If disabled, the Linksys ATA device sends RTP to the destination as indicated in the inbound
SDP.
Yes or
No
The default is yes.
PSTN Line Same as above for the PSTN line. Yes or
No
The default is yes.
Call Progress Tones
The Linksys ATA has configurable call progress tones. Call progress tones are generated locally on the ATA, so an end user is advised of status (such as ringback). Parameters for each type of tone (for instance a dial tone played back to an end user) may include:
• number of frequency components
• frequency and amplitude of each component
• cadence information.
When one VoIP account is shared between the FXS and PSTN Lines, the following parameters
are recommended to be set. See the Regional tab in the
”Linksys ATA Voice Field Reference,” on page 66
for these and other call progress tone parameters.
Linksys ATA Administration Guide 53
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Call Scenarios
Call Progress Tone
VoIP PIN Tone
PSTN PIN Tone
Outside Dial Tone
Description
This tone is played to prompt a VoIP caller to enter a PIN number.
This tone is played to prompt a PSTN caller to enter a PIN number.
During two-stage PSTN-gateway dialing and with a dial plan assigned, the Linksys
ATA device collects digits from the VoIP caller and processes the number using the dial plan. The Linksys ATA device 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.
Call Scenarios
This section describes some typical scenarios where the Linksys ATA device can be applied.
Some terms are introduced in the first few sections and reused in later sections. This section includes the following topics:
•
”PSTN to VoIP Call with and Without Ring-Thru” section on page 54
•
”VoIP to PSTN Call With and Without Authentication” section on page 55
•
”Call Forwarding to PSTN Gateway” section on page 57
•
”User Dialing 9 to Access PSTN-Gateway for Local Calls” section on page 58
•
”Using the PSTN-Gateway for 311 and 911 Calls” section on page 58
•
”Auto-Fallback to the PSTN-Gateway” section on page 59
PSTN to VoIP Call with and Without Ring-Thru
The PSTN caller calls the PSTN line connected to the FXO port. Ring-Thru is disabled. After the call rings for a delay equal to the value in PSTN Answer Delay, the VoIP gateway answers the call and prompts the PSTN caller to enter a PIN number (assuming PIN authentication is enabled).
After a valid PIN is entered, the caller is prompted to dial the VoIP number. A dial plan is selected according to the PIN number entered by the caller. If authentication is disabled, the default PSTN dial plan is used. Note than the dial plan choice cannot be 0 for a PSTN caller.
NOTE: A PSTN Access List in terms of Caller ID (ANI) patterns can be configured into the Linksys
ATA device to automatically grant access to the PSTN caller without entering the PIN. In this case, the default PSTN dial plan is also used.
The same scenario can be implemented using Ring-Thru. When the PSTN line rings, Line 1 rings also. This feature is called Ring-Thru. If Line 1 is picked up before the VoIP gateway autoanswers, it is connected to the PSTN call. Line 1 hears a call waiting tone if it is already connected to another call.
Linksys ATA Administration Guide 54
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Call Scenarios
VoIP to PSTN Call With and Without Authentication
This section describes three scenarios with and without authentication and includes the following topics:
•
”Using PIN Authentication” section on page 55
•
”Using HTTP Digest Authentication” section on page 55
•
”Without Authentication” section on page 56
Using PIN Authentication
This scenario assumes that the PSTN Line has a different VoIP account than the Line 1 account.
The VoIP caller calls the FXO number, which auto-answers after VoIP Answer Delay. The Linksys
ATA device then prompts the VoIP caller for a PIN. When a valid PIN is entered, the SPA3102 or
AG310 device plays the Outside Dial Tone and prompts the caller to dial the PSTN number.
The number dialed is processed by the dial plan corresponding to the VoIP caller. If the dial plan choice is 0, no dial plan is needed and the user hears the PSTN dial tone right after the PIN is entered. If the dial plan choice is not 0, the final number returned from the dial plan after the complete number is dialed by the caller is dialed to the PSTN. The caller does not hear the PSTN dial tone (except for a little leakage before the first digit of the final number is auto-dialed by the Linksys ATA device).
If the PSTN Line is busy (off-hook, ringing, or PSTN line not connected) when the VoIP caller calls, the Linksys ATA device replies with 503. If the PIN number is invalid or entered after the
VoIP call leg is connected, the Linksys ATA device plays the reorder tone to the VoIP caller and eventually ends the call when the reorder tone times out.
NOTE: If VoIP Caller ID Pattern is specified and the VoIP caller ID does not match any of the given patterns, the Linksys ATA device rejects the call with a 403. This rule applies regardless of the authentication method, even when the source IP address of the INVITE request is in the VoIP
Access List .
Using HTTP Digest Authentication
The same scenario can be implemented with HTTP digest authentication when the calling device supports the configuration of a auth-ID and password to access the Linksys ATA device
PSTN gateway. When the VoIP caller calls the PSTN Line, the Linksys ATA device challenges the
INVITE request with a 401 response. The calling device should then provide the correct credentials in a subsequent retry of the INVITE, computed with the auth-ID and password using
MD5.
If the credentials are correct, the target number specified in the user-id field of the INVITE
Request-URI is processed by the dial plan corresponding to the VoIP user (assuming the dial plan choice is not 0). The final number is then auto-dialed by the Linksys ATA device.
If the credentials are incorrect, the Linksys ATA device challenges the INVITE again. If the auth-
ID does not exist in the Linksys ATA device configuration, the Linksys ATA device replies 403 to the INVITE. If the target number is invalid according to the corresponding dial plan, the Linksys
ATA device also replies 403 to the INVITE. Again, if the PSTN Line is busy at the time of the call, the Linksys ATA device replies 503.
Linksys ATA Administration Guide 55
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Call Scenarios
NOTE: HTTP Digest Authentication is one way to perform one-stage dialing of a VoIP-To-PSTN call. The other way is with no authentication require. However, if the target number is not specified in the Request-URI or the number matches the account user-id of the PSTN Line, the call reverts to two-stage dialing.
Without Authentication
This scenario can also be implemented without authentication, using one-stage or two-stage dialing, as in the HTTP Authentication case. The default VoIP caller dial plan is used in this scenario. Authentication is performed when the method is none or when the source IP address of the inbound INVITE matches one of the VoIP Access List patterns.
The following table lists the parameters used in VoIP to PSTN Call With and Without
Authentication.
Parameter
VoIP Answer Delay
Outside Dial Tone
VoIP Caller ID Pattern
VoIP Access List
Tab Description Values
PSTN Line A comma-separated list of IP address templates, such that callers with a 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.
Regional 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 encountered in the dial plan.
The default is
420@-
19;10(*/0/1).
PSTN Line 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.
PSTN Line A comma-separated list of IP address templates, such that callers with a source IP address matching any of the templates are accepted for PSTN gateway service without further authentication.
For example:
1408*,
1512???1234.
Note: ‘?’ matches any single digit; ‘*’ matches any number of digits.
The default is blank.
For example:
192.168.*.*,
66.43.12.1??.
The default is blank.
Linksys ATA Administration Guide 56
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Call Scenarios
Call Forwarding to PSTN Gateway
This section describes a number of scenarios that forward calls to the PSTN gateway. It includes the following topics:
•
”Forward-On-No-Answer to the PSTN Gateway” section on page 57
•
”Forward-All to the PSTN gateway” section on page 57
•
”Forward to a Particular PSTN Number” section on page 57
•
”Forward-On-Busy to PSTN Gateway or Number” section on page 57
•
”Forward-Selective to PSTN Gateway or Number” section on page 58
Forward-On-No-Answer to the PSTN Gateway
In this scenario, Line 1 is configured to Cfwd No Ans Dest to the PSTN Gateway. The scenario is implemented by setting User 1 to forward to gw0 on no answer, with Cfwd No Ans Delay set to six seconds.
The caller calls Line 1 and if Line 1 is not picked up after six seconds, the PSTN Line picks up the call and the call reverts to a PSTN-Gateway call, as described above. In this case, HTTP authentication is not allowed because Line 1 does not authenticate inbound INVITE requests. If you need to authenticate the VoIP caller in this case, you must select the PIN authentication method, or else the caller is not authenticated.
NOTE: If the PSTN Line is busy at the moment of the forward, it does not answer the VoIP call.
The call forward rule is ignored and Line 1 continues to ring.
Forward-All to the PSTN gateway
In this scenario, Line 1 is configured with Cfwd All Dest parameter to the PSTN gateway.This scenario is the same the previous case, except the FXO picks up the Line 1 call immediately.
If the PSTN Line is busy at the moment of the call, the PSTN Line does not pick up the call, the call forward rule is ignored, and Line 1 continues to ring.
Forward to a Particular PSTN Number
In this scenario, the forward destination is set to <target-number>@gw0>. This is the same as in the previous examples, except that the Linksys ATA device automatically dials the given target number on the PSTN line right after it answers the VoIP call leg. This is a special case of onestage dialing where the target number is specified in the configuration. The caller is not authenticated in this case regardless of the authentication method. However, the caller is still limited by the VoIP Caller ID Pattern parameter
Forward-On-Busy to PSTN Gateway or Number
This scenario is similar to the previous cases of call forwarding to gw0, but this applies when
Line 1 is active.
Linksys ATA Administration Guide 57
Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Call Scenarios
Forward-Selective to PSTN Gateway or Number
This scenario is similar to the previous cases of call forwarding to gw0, but this applies when the caller matches the specific caller-id pattern.
User Dialing 9 to Access PSTN-Gateway for Local Calls
To implement this scenario, add the rule “<9,:1408>xxxxxxx<:@gw0>” to the Line 1 dial plan.
When user dials 9, Linksys ATA device plays outside dial tone. The user then dials seven digits and the Linksys ATA device prepends 1408 before dialing the final number on the PSTN line.
Using the PSTN-Gateway for 311 and 911 Calls
To implement this scenario, add the rule “[39]11<:@gw0>” to Line 1. When the user dials 311 or
911, the call is routed to the PSTN gateway.
NOTE: If the PSTN Line is busy after the user dials 311 or 911, the call still fails. For true life-line supports, therefore, the PSTN Line cannot be shared.
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Configuring the PSTN (FXO) Gateway (AG310 and SPA3102)
Call Scenarios
Auto-Fallback to the PSTN-Gateway
To implement this scenario, enable the Auto PSTN Fallback parameter. When registration fails or link is down, the Linksys ATA device automatically calls “fallback@gw0” when user picks up Line
1. The Linksys ATA device does not reboot when the link is down. However, the Linksys ATA device reboots when the link is back up and Line 1 and PSTN Line are not in use.
The following table lists the parameters used in Call Forwarding to PSTN Gateway.
Parameter
Cfwd No Ans Dest
Cfwd No Ans Delay
Cfwd All Dest
VoIP Caller ID Pattern
Auto PSTN Fallback
Tab
User
User
Description
Forward number for Call Forward All Service
Delay in sec before Call Forward No Answer triggers.
Same as Cfwd All Dest.
Values
—
20
User Forward number for Call Forward All Service
PSTN Line 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.
PSTN Line 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).
—
For example:
1408*,
1512???1234.
The default is blank.
The default is
yes.
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