Cisco SPA1001 Administration Guide - Page 74

How VoIP-To-PSTN Calls Work - spa 100

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How VoIP-To-PSTN Calls Work Chapter 4 Configuring the PSTN Gateway (FXO) 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, 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. 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, 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 The Request-URI of the INVITE to the PSTN Line should have the form @, where is the number dialed by the VoIP caller, and is a valid address of the SPA3102, 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 SPA3102 replies to the INVITE with a 503 response. Otherwise, it compares the with the of the PSTN Line. If they are the same, the SPA3102 interprets this as a request for two-stage dialing (see the "Two-Stage Dialing" section on page 4-3). If they are different, the SPA3102 processes the using the corresponding . If dial plan processing fails, the SPA3102 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 on the PSTN Line is blank, should be disabled for the PSTN Line. If HTTP Digest Authentication is enabled, the SPA3102 challenges the INVITE with a 401 response if it does not have a valid Authorization header. The Authorization header should include a parameter, where n refers to one of eight VoIP user accounts that can be configured on the SPA3102. The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The must match one of the VoIP accounts stored on the SPA3102. Each VoIP user account contains the information listed in Table 1-27. Linksys ATA Administrator Guide 4-2 Document Version 3.1

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4-2
Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 4
Configuring the PSTN Gateway (FXO)
How VoIP-To-PSTN Calls Work
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, 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.
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, 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
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, 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
SPA3102 replies to the INVITE with a 503 response. Otherwise, it compares the <Dialed-Number> with
the <User ID> of the PSTN Line. If they are the same, the SPA3102 interprets this as a request for
two-stage dialing (see the
“Two-Stage Dialing” section on page 4-3
). If they are different, the SPA3102
processes the <Dialed-Number> using the corresponding <Dial Plan>.
If dial plan processing fails, the SPA3102 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 <User ID> on the PSTN Line is blank, <Registration> should be disabled for the PSTN Line.
If HTTP Digest Authentication is enabled, the SPA3102 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 SPA3102.
The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The
<User ID
n
> must match one of the VoIP accounts stored on the SPA3102. Each VoIP user account
contains the information listed in
Table 1-27
.