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Linksys ATA
Administrator Guide
Document Version 3.1
Chapter 3
Configuring Linksys ATAs
Troubleshooting and Configuration FAQ
B. Press
CTRL + F5
. This is a hard refresh, which forces Windows Explorer to load new webpages, not
cached ones.
C. Click
Tools
. Click
Internet Options
. Click the
Security
tab. Click the
Default level
button. Make
sure the security level is Medium or lower. Then click the
OK
button.
3.
How do I save my current SPA configuration?
Currently, the only way is to do HTTPGET from an HTTP client, from which you get the entire
HTML page. Alternatively, from your browser you can select
File
>
Save as
>
HTML
from any of
the administration web server pages. Do this in Admin, Advanced mode.
This saves all the tabs into one HTML file. This HTML file is helpful to provide to our support team
when you have a problem or technical question.
4.
How do I debug my SPA? Is there a syslog?
SPA sends out debug information via syslog to a syslog server. The ports can be configured (by default
the port is 514).
A.
Make sure you do not have firewall running on your PC that could block port 514.
B.
On the administration web server System tab, set <Debug Server> as the IP address and port
number of your syslog server. Note that this address has to be reachable from the Linksys ATA).
C.
Also, set <Debug level> to
3.
±
You do not need to change the value of the <syslog server> parameter.
D.
To capture SIP signaling messages, under the Line tab, set <SIP Debug Option> to
Full
.
±
The file output is syslog.<portnum>.log (for the default port setting, syslog.514.log)
5.
How do I access the
Linksys ATA
if I forget my password?
By default, the User and Admin accounts have no password. If the ITSP set the password for either
account and you do not know what it is, you need to contact the ITSP. If the password for the user
account was configured after you received the Linksys ATA, you can reset the device to the user
factory default, which preserves any provisioning completed by the ITSP. If the Admin account
needs to be reset, you have to perform a full factory reset, which also erases any provisioning.
To reset the Linksys ATA to the factory defaults, perform the following steps:
A.
Connect an analog phone to the Linksys ATA and access the IVR by pressing ****.
B.
Press the appropriate code to reset the unit:
±
Press 877778# to reset the unit to the defaults as it shipped from the ITSP. This will reset the
User account password to the default of blank.
Press
73738
# to perform a full reset of unit to the defaults as it shipped from Linksys. This will
reset the Admin account password to the default of blank.
C.
Press 1 to confirm the operation.
±
Press * to cancel the operation.
D.
Login to the unit using the User or Admin account without a password and reconfigure the unit as
necessary.
6.
My
Linksys ATA
is behind a NAT device or firewall and I’m unable to make a call or I’m only
receiving a one-way connection. What should I do?
A.
Configure your router to port forward “TCP port 80" to the ip address currently being used by
SPA. If you do this often, we suggest that you use static IP address for the SPA, instead of
DHCP. (For help with port forwarding, consult your router documentation)
B.
On the Line tab of the administration web server, change the value of <Nat Mapping Enable> to
yes
. On the SIP tab; change <Substitute VIA Addr> to
yes
, and <EXT IP> to the IP address of
your router.
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Linksys ATA
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Chapter 3
Configuring Linksys ATAs
Troubleshooting and Configuration FAQ
C.
Make
sure you are not blocking the UDP PORT 5060,5061 and port for UDP packets in the
range of 16384-16482. Also, disable “SPI” if this feature is provided by your firewall. Identify
the SIP server to which the Linksys ATA is registering, if it supports NAT, using the <Outbound
Proxy> parameter.
D.
Add a STUN server to allow traversal of UDP packets through the NAT device. On the SIP tab of the
administration web server, set <STUN Enable> to
yes
, and enter the IP address of the STUN server
in <STUN Server>.
±
±
STUN (Simple Traversal of UDP through NATs) is a protocol defined by RFC 3489, that allows a
client behind a NAT device to find out its public address, the type of NAT it is behind, and the port
associated on the Internet connection with a particular local port. This information is used to set up
UDP communication between two hosts that are both behind NAT routers. Open source STUN
software can be obtained at the following website:
Note
STUN does not work with a symmetric NAT router. Enable debug through syslog (see FAQ#10),
and set <STUN Test Enable> to
yes
. The messages indicate whether you have symmetric NAT
or not.
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Administrator Guide
Document Version 3.1
C H A P T E R
4
Configuring the PSTN Gateway (FXO)
This chapter describes how to configure the PSTN gateway provided by Analog Telephone Adapters
(ATAs) with one or more FXO ports, which includes the AG310 and SPA3102. It includes the following
sections:
Overview, page 4-1
How VoIP-To-PSTN Calls Work, page 4-2
How PSTN-To-VoIP Calls Work, page 4-4
Configuring VoIP Failover to PSTN, page 4-7
Sharing One VoIP Account Between the FXS and PSTN Lines, page 4-7
Other Options, page 4-7
Call Scenarios, page 4-8
Note
Througout this chapter, when references are made to the software configuration of the SPA3102, the
information also applies to the AG310. The AG310 provides the same functionality as the SPA3102, as
well as an ADSL modem and a four-port Ethernet switch.
Overview
The SPA3102 and AG310 have the following ports for connection to telephony devices:
FXS port (Phone)—Connected to a standard analog telephone or fax machine, configured using the
Line tab
FXO port (Line)—Connected 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 PSTN-To-VoIP calling function as a
VoIP gateway
. Note
the following definitions:
VoIP caller—One who calls the AG310 via VoIP to obtain PSTN service
VoIP user—VoIP caller that has a user account (user-id and password) on the AG310
PSTN caller—One who calls the AG310 from the PSTN to obtain VoIP service
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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
.
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Chapter 4
Configuring the PSTN Gateway (FXO)
How VoIP-To-PSTN Calls Work
Note
If Authentication is disabled, a default dial plan is used for all unknown VoIP users.
Two-Stage Dialing
In two-stage dialing, the SPA3102 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 SPA3102 (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 SPA3102
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 SPA3102. A dial plan can
be selected for each PIN number. If the caller enters a wrong PIN or the SPA3102 times out waiting for
more PIN digits, the SPA3102 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.
Table 1-28
lists the parameters used in two-stage dialing.
Table 1-27
VoIP User Account Information
Parameter
Group
Description
Values
User ID
1/2/3/4/5/6/7/
8
PSTN
Line
The username value.
31-character string
Password
1/2/3/4/5/6/7/
8
PSTN
Line
The password value.
31-character string
User
1/2/3/4/5/6/7/
8 DP
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

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