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4-9
Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
Call Scenarios
PSTN to VoIP Call with and Without Ring-Thru, page 4-9
VoIP to PSTN Call with and Without Authentication, page 4-9
Call Forwarding to PSTN Gateway, page 4-10
User Dialing 9 to Access PSTN-Gateway for Local Calls, page 4-11
Using the PSTN-Gateway for 311 and 911 Calls, page 4-12
Auto-Fallback to the PSTN-Gateway, page 4-12
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 it rings
for a delay equals 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, a regular dial plan is played to prompt the PSTN caller 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 SPA3102 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 auto-answers, it is
connected to the PSTN call. Line 1 hears a call waiting tone if it is already connected to another call.
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, page 4-9
Using HTTP Digest Authentication, page 4-10
Without Authentication, page 4-10
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 SPA3102 then
prompts the VoIP caller for a PIN. When a valid PIN is entered, the SPA3102 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 SPA3102).
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
Call Scenarios
If the PSTN Line is busy (off-hook, ringing, or PSTN line not connected) when the VoIP caller calls, the
SPA3102 replies with 503. If the PIN number is invalid or entered after the VoIP call leg is connected,
the SPA3102 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 SPA3102 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 SPA3102 PSTN gateway. When the
VoIP caller calls the PSTN Line, the SPA3102 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 SPA3102.
If the credentials are incorrect, the SPA3102 challenges the INVITE again. If the auth-ID does not exist
in the SPA3102 configuration, the SPA3102 replies 403 to the INVITE. If the target number is invalid
according to the corresponding dial plan, the SPA3102 also replies 403 to the INVITE. Again, if the
PSTN Line is busy at the time of the call, the SPA3102 replies 503.
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.
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, page 4-11
Forward-All to the PSTN gateway, page 4-11
Forward to a Particular PSTN Number, page 4-11
Forward-On-Busy to PSTN Gateway or Number, page 4-11
Forward-Selective to PSTN Gateway or Number, page 4-11
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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
Call Scenarios
Forward-On-No-Answer to the PSTN Gateway
In this scenario, Line 1 is configured to <Forward-On-No-Answer> to the PSTN Gateway. The scenario
is implemented by setting User 1 to forward to gw0 on no answer, with <No Answer 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 <Forward-All> 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 SPA3102 automatically dials the given target number on the PSTN
line right after it answers the VoIP call leg. This is a special case of one-stage 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.
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, SPA3102 plays outside dial tone. The user then dials seven digits and the SPA3102
prepends 1408 before dialing the final number on the PSTN line.
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Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
Call Scenarios
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.
Auto-Fallback to the PSTN-Gateway
To implement this scenario, enable <Auto PSTN Fallback>.
When registration fails or link is down, the
SPA3102 automatically calls “fallback@gw0” when user picks up Line 1. The SPA3102 does
not
reboot
when the link is down. However, the SPA3102 reboots when the link is back up and Line 1 and PSTN
Line are not in use.
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Linksys IP Phone Administrator Guide
Document Version 3.2
C H A P T E R
5
Linksys
ATA Field Reference
This chapter describes the fields within each section of the following administration web server pages:
Info Tab, page 5-2
System Tab, page 5-8
SIP Tab, page 5-11
Regional Tab, page 5-19
Line Tab, page 5-33
PSTN Line Tab, page 5-49
User 1/2 Tab, page 5-65
PSTN User Tab (SPA3102/AG310), page 5-70
Note
Througout this chapter, references to the SPA3102 also apply to the AG310. The AG310 provides the
same functionality as the SPA3102, as well as an ADSL modem and a four-port Ethernet switch.
For information about the tabs on the Routing page, see the documentation for any Linksys router. For
information about the Provisioning page, see the
Linksys SPA Provisioning Guide
.

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