Configuring Linksys ATA Features. Linksys WRTP54G, WAG54GP2, SPA3102, PAP2T, WRP400, SPA8000, AG310, SPA2102, RTP300

4

Configuring Linksys ATA Features

Supported Codecs

Configuring Linksys ATA Features

This chapter contains the following topics:

•

”Supported Codecs,” on page 26

•

”Configuring a Dial Plan” section on page 27

•

”Secure Call Implementation” section on page 31

•

”Configuring a Streaming Audio Server” section on page 35

•

”Using a FAX Machine (SPA2102, SPA3102 or SPA8000)” section on page 38

•

”Managing Caller ID Service” section on page 44

•

”Silence Suppression and Comfort Noise Generation” section on page 45

Supported Codecs

The following list shows the current supported codecs for each Linksys ATA product. If you need to change the G711u codec which is configured by default, set your preferred codecs in the FXS Line tab(s); Audio Configuration. You may set your first, second, and third preferred

codec. See

”Linksys ATA Routing Field Reference,” on page 60

.

PAP2T / SPA2102 / SPA3102 / SPA8000 / AG310

• G.711u (configured by default)

• G.711a

• G.726-16

• G.726-24

• G.726-32

• G.726-40

• G.729a

• G.723

WRTP54G/RTP300 / WAG54GP2

• G.711u

(configured by default)

• G.711a

• G.726-32

• G.729a

• G.723

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Configuring Linksys ATA Features

Configuring a Dial Plan

WRP400

• G.711u

(configured by default)

• G711a

• G.726-32

• G.729a

Configuring a Dial Plan

The Linksys ATA device allows each line to be configured with a distinct dial plan. The dial plan specifies how to interpret digit sequences dialed by the user, and how to convert those sequences into an outbound dial string.

The Linksys ATA syntax for the dial plan closely resembles the corresponding syntax specified by MGCP and MEGACO. Some extensions are added that are useful in an end-point.

NOTE: When using the SPA3102 or AG310 as a PSTN gateway, gateway calls can be restricted on a per-caller basis using dial plans. Up to eight dial plans can be configured to restrict gateway calls in either direction.

Dial Plan Digit Sequences

The plans contain a series of digit sequences, separated by a vertical bar ( | ). The collection of sequences is enclosed in parentheses.

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match.

The matching sequences form a set of candidate digit sequences. As more digits are entered by the user, the set of candidates diminishes until only one or none are valid.

Any one of a set of terminating events triggers the Linksys ATA device to either accept the userdialed sequence and transmit it to initiate a call, or else to reject it as invalid. The terminating events are as follows:

• No candidate sequences remain—The number is rejected.

• Only one candidate sequence remains, and it has been matched completely—The number is accepted and transmitted after any transformations indicated by the dial plan, unless the sequence is barred by the dial plan, in which case the number is rejected.

• A timeout occurs—The digit sequence is accepted and transmitted as dialed if incomplete, or transformed as per the dial plan if complete.

• An explicit “send” (user presses the # key)—The digit sequence is accepted and transmitted as dialed if incomplete, or transformed as according to the dial plan if complete.

The time-out duration depends on the matching state. If no candidate sequences are as yet complete (as dialed), the Interdigit Long Timeout parameter applies. If a candidate sequence is

Linksys ATA Administration Guide 27

Configuring Linksys ATA Features

Configuring a Dial Plan

complete, but there exists one or more incomplete candidates, then the

Interdigit_Short_Timeout parameter applies.

The dial plan is configured in the Line tab and a default North American dial plan is provided.

The following table describes the entries to use when programming the dial plan.

Dial Plan Entry

*xx

[3469]11

0

00

[2-9]xxxxxx

1xxx[2-9]xxxxxx xxxxxxxxxx.

Function

Allows arbitrary 2-digit star code

Allows x11 sequences (for example, 311, 411, 611, 911)

Dials operator

Dials international operator

Dials US local number

Dials US 1 + 10-digit long distance number

Dials all other numbers, including international long distance

NOTE: Early production versions of the SPA2102 supported dual-line telephones on a single

FXS port. A subsequent hardware change revised the FXS ports on the SPA2102 and this function is to be reserved for future development.

Dial Plan Rules

This section describes the rules that apply to configuring and interpreting dial plans.

NOTE: White space is ignored, but may be used for readability.

Digit Sequence Syntax

Each digit sequence within the dial plan consists of a series of elements, which are individually matched to the keys pressed by the user. Elements can be one of the following:

• Individual keys 0, 1, 2 . . . 9, *, #.

• The letter x matches any one numeric digit (0 .. 9)

• A subset of keys within brackets (allows ranges): for example, [389] means 3 or 8 or 9)

– Numeric ranges (n-n) are allowed within the brackets: for example, [2-9] means any digit from 2 through 9)

– Ranges can be combined with other keys: e.g. [235-8*] means 2 or 3 or 5 or 6 or 7 or

8 or *.

Element Repetition

Any element can be repeated zero or more times by appending a period (.) to the element.

Thus, “01.” matches “0”, “01”, “011”, “0111”, … and so on.

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Configuring Linksys ATA Features

Configuring a Dial Plan

Sub-sequence Substitution

A sub-sequence of keys (possibly empty) can be automatically replaced with a different subsequence using an angle bracket notation: <dialed-subsequence : transmitted-subsequence> .

So, for example, “<8:1650>xxxxxxx” would match “85551212” and transmit “16505551212”.

Inter-sequence Tones

An “outside line” dial tone can be generated within a sequence by appending a comma (,) between digits. Thus, the sequence “9, 1xxxxxxxxxx” sounds an “outside line” dial tone after the user presses 9, until the 1 is pressed.

Number Barring

A sequence can be barred (rejected) by placing a ! character at the end of the sequence. Thus,

“1900xxxxxxx!” automatically rejects all 900 area code numbers from being dialed.

Interdigit Timer Master Override

The long and short interdigit timers can be changed in the dial plan (affecting a specific line) by preceding the entire plan with the following syntax:

• Long interdigit timer: L : delay-value,

• Short interdigit timer: S : delay-value,

Thus, “L:8,( . . . )” would set the interdigit long timeout to 8 seconds for the line associated with this dial plan. And, “L:8,S:4,( . . . )” would override both the long and the short time-out values.

Local Timer Overrides

The long and short time-out values can be changed for a particular sequence starting at a particular point in the sequence. The syntax for long timer override is: L delay-value<space>.

Note the terminating space character. The specified delay-value is measured in seconds.

Similarly, to change the short timer override, use: S delay-value<space>.

Pause

A sequence may require an explicit pause of some duration before continuing to dial digits, in order for the sequence to match. The syntax for this is similar to the timer override syntax: P delay-value <space>. The delay-value is measured in seconds.

This syntax allows for the implementation of Hot-Line and Warm-Line services. To achieve this, one sequence in the plan must start with a pause, with a 0 delay for a Hot Line, and a non-zero delay for a Warm Line.

Implicit Sequences

The Linksys ATA device implicitly appends the vertical code sequences entered in the Web

Configuration Utility Regional parameter settings to the end of the dial plan for both Line 1 and

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Configuring Linksys ATA Features

Configuring a Dial Plan

Line 2. Likewise, if the parameter Enable_IP_Dialing is enabled, then IP dialing is also accepted on the associated line.

Parameter Tab

Enable_IP_Dialing

Line

Description

Enable or disable IP dialing.

Values

The default is no.

Dial Plan Examples

The following dial plan accepts only US-style 1 + area-code + local-number, with no restrictions on the area code and number:

( 1 xxx xxxxxxx )

The following also allows 7-digit US-style dialing, and automatically inserts a 1 + 212 (local area code) in the transmitted number.

( 1 xxx xxxxxxx | <:1212> xxxxxxx )

For an office environment, the following plan requires a user to dial 8 as a prefix for local calls and 9 as a prefix for long distance. In either case, an “outside line” tone is played after the initial

8 or 9, and neither prefix is transmitted when initiating the call.

( <9,:> 1 xxx xxxxxxx | <8,:1212> xxxxxxx )

The following allows only placing international calls (011 call), with an arbitrary number of digits past a required 5 digit minimum, and also allows calling an international call operator

(00). In addition, it lengthens the default short interdigit timeout to 4 seconds.

S:4, ( 00 | 011 xxxxx x. )

The following allows only US-style 1 + area-code + local-number, but disallows area codes and local numbers starting with 0 or 1. It also allows 411, 911, and operator calls (0).

( 0 | [49]11 | 1 [2-9]xx [2-9]xxxxxx )

The following allows US-style long distance, but blocks 9xx area codes:

( 1 [2-8]xx [2-9]xxxxxx )

The following allows arbitrary long distance dialing, but explicitly blocks the 947 area code.

( 1 947 xxxxxxx ! | 1 xxx xxxxxxx )

The following implements a hot line phone, which automatically calls 1 212 5551234.

( S0 <:12125551234> )

The following provides a warm line to a local office operator (1000) after five seconds, unless a four-digit extension is dialed by the user.

( P5 <:1000> | xxxx )

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Configuring Linksys ATA Features

Secure Call Implementation

Dial Plan Timers

The dial plan functionality is regulated by the following configurable parameters:

Interdigit Long and Short Timer

Parameter Tab

Interdigit_Long_

Timeout

Regional

Interdigit_Short

_Timeout

Regional

Description

Specifies the default maximum time (in seconds) allowed between dialed digits, when no candidate digit sequence is as yet complete.

Specifies the default maximum time (in seconds) allowed between dialed digits, when at least one candidate digit sequence is complete as dialed.

Values

Range: 0–64 seconds.

The default is 10.

Range: 0–64 seconds.

The default is 3.

Dial Plans

Parameter Tab

Enable_IP_Dialing

Line

Dial Plan

Line

Description

Enable or disable IP dialing.

Dial Plan script for your Line.

Values

The default is no.

The default is: (*xx |

[3469]11 | 0 | 00 |

<:1408>[2-9]xxxxxx |

1[2-9]xx[2-9]xxxxxx |

011x. )

Secure Call Implementation

This section describes secure call implementation with a Linksys ATA device. It includes the following topics:

•

”Enabling Secure Calls” section on page 31

•

”Secure Call Details” section on page 32

•

”Using a Mini-Certificate” section on page 33

•

”Generating a Mini-Certificate” section on page 33

NOTE: This is an advanced topic meant for experience installers. See also the LVS Provisioning

Guide.

Enabling Secure Calls

A secure call is established in two stages. The first stage is no different from normal call setup.

The second stage starts after the call is established in the normal way with both sides ready to stream RTP packets.

In the second stage, the two parties exchange information to determine if the current call can switch over to the secure mode. The information is transported by base64 encoding embedded in the message body of SIP INFO requests, and responses using a proprietary format. If the

Linksys ATA Administration Guide 31

Configuring Linksys ATA Features

Secure Call Implementation

second stage is successful, the Linksys ATA device plays a special Secure Call Indication Tone for a short time to indicate to both parties that the call is secured and that RTP traffic in both directions is being encrypted.

If the user has a phone that supports call waiting caller ID (CIDCW) and that service is enabled, the CID will be updated with the information extracted from the Mini-Certificate received from the remote party. The Name field of the CID will be prepended with a ‘$’ symbol. Both parties can verify the name and number to ensure the identity of the remote party.

The signing agent is implicit and must be the same for all Linksys ATA devices that communicate securely with each other. The public key of the signing agent is pre-configured into the Linksys ATA device by the administrator and is used by the Linksys ATA device to verify the Mini-Certificate of its peer. The Mini-Certificate is valid if it has not expired, and it has a valid signature.

The Linksys ATA device can be configured so that, by default, all outbound calls are either secure or not secure. If secure by default, the user has the option to disable security when making a call by dialing *19 before dialing the target number. If not secure by default, the user can make a secure outbound call by dialing *18 before dialing the target number. However, the user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.

The Linksys ATA device will not switch to secure mode if the CID of the called party from its

Mini-Certificate does not agree with the user-id used in making the outbound call. The Linksys

ATA device performs this check after receiving the Mini-Certificate of the called party

Secure Call Details

Looking at the second stage of setting up a secure call in greater detail, this stage can be further divided into two steps.

1. The caller sends a “Caller Hello” message (base64 encoded and embedded in the message body of a SIP INFO request) to the called party with the following information:

• Message ID (4B)

• Version and flags (4B)

• SSRC of the encrypted stream (4B)

• Mini-Certificate (252B)

Upon receiving the Caller Hello, the called party responds with a Callee Hello message

(base64 encoded and embedded in the message body of a SIP response to the caller’s INFO request) with similar information, if the Caller Hello message is valid. The caller then examines the Callee Hello and proceeds to the next step if the message is valid.

2. The caller sends the “Caller Final” message to the called party with the following information:

• Message ID (4B)

• Encrypted Master Key (16B or 128b)

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Configuring Linksys ATA Features

Secure Call Implementation

• Encrypted Master Salt (16B or 128b)

The Master Key and Master Salt are encrypted with the public key from the called party mini-certificate. The Master Key and Master Salt are used by both ends for deriving session keys to encrypt subsequent RTP packets. The called party then responds with a Callee Final message (which is an empty message).

Using a Mini-Certificate

The Linksys ATA device Mini-Certificate (MC) contains the following information:

• User Name (32B)

• User ID or Phone Number (16B)

• Expiration Date (12B)

• Public Key (512b or 64B)

• Signature (1024b or 512B)

The MC has a 512-bit public key used for establishing secure calls. The administrator must provision each subscriber of the secure call service with an MC and the corresponding 512-bit private key. The MC is signed with a 1024-bit private key of the service provider, which acts as the CA of the MC. The 1024-bit public key of the CA signing the MC must also be provisioned for each subscriber.

The CA public key is used by the Linksys ATA device to verify the MC received from the other end. If the MC is invalid, the Linksys ATA device will not switch to secure mode. The MC and the

1024-bit CA public key are concatenated and base64 encoded into the single parameter Mini

Certificate. The 512-bit private key is base64 encoded into the SRTP Private Key parameter, which should be kept secret, like a password. (Mini Certificate and SRTP Private Key are configured in the Line tabs.)

Because the secure call establishment relies on exchange of information embedded in message bodies of SIP INFO requests/responses, the service provider must ensure that the network infrastructure allows the SIP INFO messages to pass through with the message body unmodified.

Generating a Mini-Certificate

Linksys provides a configuration tool called gen_mc for the generation of MC and private keys.

NOTE: For Europe, the Middle East, and Africa the gen_mc tool is available on linksys-itsp.com.

For North America and other regions, It is available on the partner section of linksys.com.

The gen_mc tool uses the following syntax: gen_mc ca-key user-name user-id expire-date

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Configuring Linksys ATA Features

Secure Call Implementation

Where:

•

ca-key

is a text file with the base64 encoded 1024-bit CA private/public key pairs for signing/verifying the MC, such as the following:

9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYx

WCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTj j13qvYs=

5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/

IqSrsf6scsmundY5j7Z5mK5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3MF+zjyY rVUFdM+pXtDBxmM+fGUfrpAuXb7/k=

•

user-name

is the name of the subscriber, such as “Joe Smith”. Maximum length is 32 characters

•

user-id

is the User ID of the subscriber, which must match exactly the user-id used in the

INVITE when making the call, such as “14083331234”. The maximum length is 16 characters.

•

expire-date

is the expiration date of the MC, such as “00:00:00 1/1/34” (34=2034).

Internally the date is encoded as a fixed 12B string: 000000010134

The tool generates the Mini Certificate and SRTP Private Key parameters that can be provisioned to the Linksys ATA device.

For example: gen_mc ca_key “Joe Smith” 14085551234 “00:00:00 1/1/34”

Produces the following Mini Certificate and SRTP Private Key:

<Mini Certificate>

Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEwMTM00O vJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/ xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhxES767G

0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/uQ/

LJQlni2kwqlm7scDvll5k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxE

OGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1 fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs=

<SRTP Private Key> b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/ e+Klm4XiiFEPaKmU8UbooxKG36SEdKusp0AQ==

Linksys ATA Administration Guide 34

Configuring Linksys ATA Features

Configuring a Streaming Audio Server

Configuring a Streaming Audio Server

This section describes how to use and configure a streaming audio server (SAS). It includes the following topics:

•

”Music On Hold” section on page 35

•

”Using a Streaming Audio Server” section on page 35

•

”Using the IVR with an SAS Line” section on page 36

•

”Example SAS with MOH” section on page 37

•

”SAS Line Not Registered with the Proxy Server” section on page 37

Music On Hold

On a connected call, the Linksys ATA device may place the remote party on hold by performing a hook-flash to initiate a three-way call or by swapping two calls during call-waiting. If the remote client indicates that it can still receive audio while the call is holding, the Linksys ATA device can be configured to contact an auto-answering streaming audio server (SAS) to stream audio to the holding party. When used this way, the SAS is referred to as an MOH Server.

Using a Streaming Audio Server

The SAS feature lets you use attach an audio source to one of the Linksys ATA device FXS ports

(Phone 1 or Phone 2 on the PAP2T) and use it as a streaming audio source device. The corresponding Line (1 or 2) can be configured as a streaming audio server (SAS). If the Linksys

ATA device has multiple FXS ports, either or both of the associated lines (Line 1 and Line 2 on the PAP2T) can be configured as an SAS server.

To connect an external music source to an FXS port, use a media signal adapter, which provides a line in from a media source and a RJ-11 port for connecting to the FXS port on the Linksys ATA device. The following is a URL for a device that has been tested with Linksys ATA devices: http://www.neogadgets.com/cart/ cart.php?target=product&product_id=17&substring=music+coupler

After installing the music source using the media signal adapter and completing the required configuration on the Linksys ATA device, when the line is called and the FXS port is off hook, the

Linksys ATA device answers the call automatically and streams audio to the caller.

If the FXS port is on-hook when the incoming call arrives, the Linksys ATA device replies with a

SIP 503 response code (Service Not Available). The SAS line will not ring for incoming calls even if the attached equipment is on-hook.

If an incoming call is auto-answered, but later the FXS port changes to on-hook, the SPA does not terminate the call but continues to stream silence packets to the caller. If an incoming call arrives when the SAS line has reached full capacity, the SPA replies with a SIP 486 response

(Busy Here).

The SAS line can be set up to refresh each streaming audio session periodically using a SIP re-

INVITE message, which detects if the connection to the caller is down. If the caller does not

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Configuring Linksys ATA Features

Configuring a Streaming Audio Server

respond to the refresh message, the SAS line terminates the call so that the streaming resource can be used for other callers.

Considerations:

Each SAS server can maintain up to five simultaneous calls. If the second line on the Linksys ATA device is disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls will receive a busy signal (SIP 486 Response).

If no calls are in session, battery is removed from tip-and-ring of the FXS port. Some audio source devices have an LED to indicate the battery status. This can be used as a visual indication as to whether audio streaming is in progress.

Set up the Proxy and Subscriber Information for the SAS Line as you normally would with a regular user account.

Call Forwarding, Call Screening, Call Blocking, DND, and Caller-ID Delivery features are not available on an SAS line.

Using the IVR with an SAS Line

The IVR can still be used on an SAS line, but the user needs to follow the following steps:

1. Power off the Linksys ATA device.

2. Connect a phone to the port and make sure the phone is on-hook.

3. Power on the Linksys ATA device.

4. Pick up handset and press * * * * to invoke IVR in the usual way.

If the Linksys ATA device boots and finds that the SAS line is on-hook, it will not remove battery from the line so that IVR may be used. But if the Linksys ATA device boots up and finds that the

SAS line is off-hook, it will remove battery from the line because no audio session is in progress.

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Configuring Linksys ATA Features

Configuring a Streaming Audio Server

Example SAS with MOH

SPA1:

IP=192.168.2.100

UserID[1]=1001, SIP Port[1]=5060

UserID[1]=1002, SIP Port[1]=5061

SPA2:

IP=192.168.2.200

UserID[1]=2001, SIP Port[1]=5060

UserID[1]=2002, SIP Port[1]=5061

IP Network

Phone 1

Phone 2

Phone 1

Phone 2

Media signal adapter

Line in

Music source

The above configuration shows MOH Application with a Linksys ATA device Line Configured as an SAS. The examples that follow are based on the configuration shown in above example.

SAS Line Registered with the Proxy Server

In this example, the SAS Line is registered with the Proxy Server as the other subscribers.

On Linksys ATA device 1:

SAS Enable[1] = no

MOH Server [1] = 1002

SAS Enable[2] = yes

On Linksys ATA 2:

SAS Enable[1] = no

MOH Server [1] = 1002

SAS Enable[2] = no

MOH Server [2] = 1002

SAS Line Not Registered with the Proxy Server

In this example, the SAS Line is not registered with the Proxy Server for the other subscribers.

On Linksys ATA device 1:

SAS Enable[1] = no

MOH Server [1] = [email protected]:5061 or [email protected]:5061

SAS Enable[2] = yes

On Linksys ATA device 2:

SAS Enable[1] = no

MOH Server [1] = [email protected]:5061

SAS Enable[2] = no

Linksys ATA Administration Guide 37

Configuring Linksys ATA Features

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

MOH Server [2] = [email protected]:5061

Configuring the Streaming Audio Server

The following provides step-by-step procedures for implementing an SAS with an external music source.

1. Connect an RJ-11 adapter between the music source and an FXS port on the Linksys ATA device (Phone 1 or Phone 2).

2. On the Web Configuration Utility, click the SIP tab and scroll down to the Streaming Audio

Server section.

3. On the SAS Enable pull-down selection list, select yes.

4. In the MOH Server field of the Call Feature Settings section, enter the User ID configured for the line attached to the audio source.

If the line is not registered with the SIP proxy, enter the IP address and SIP port number configured for the line.

5. Click Submit All Changes.

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

NOTE: T.38 Fax is only supported on the SPA2102, SPA3102, and the SPA8000. The SPA2102 and

SPA3102 support a single connection, while the SPA8000 supports one connection for each pair of ports (1/2, 3/4, 5/6, and 7/8) for a maximum of four connections.

To optimize fax completion rates, complete the following steps:

1. Upgrade the Linksys ATA firmware to the latest version

2. Ensure that you have enough bandwidth for uplink and downlink.

• For G.711 fallback, it is recommend to have approximately 100Kbps.

• For T.38, allocate at least 50 kbps.

3. To optimize G.711 fallback fax completion rates, set the following on the Line tab of your

Linksys ATA:

• Network Jitter Buffer: very high

• Jitter buffer adjustment: disable

• Call Waiting: no

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Configuring Linksys ATA Features

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

• 3 Way Calling: no

• Echo Canceller: no

• Silence suppression: no

• Preferred Codec: G.711

• Use pref. codec only: yes

4. If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax relay) and enable fax using modem passthrough.

For example: modem passthrough nse payload-type 110 codec g711ulaw fax rate disable fax protocol pass-through g711ulaw

5. Enable T.38 fax on the SPA 2102 by configuring the following parameter on the Line tab for the FXS port to which the FAX machine is connected:

FAX_Passthru_Method: ReINVITE

NOTE: If a T.38 call cannot be set-up, then the call should automatically revert to G.711 fallback.

6. If you are using a Cisco media gateway use the following settings:

Make sure the Cisco gateway is correctly configured for T.38 with the SPA dial peer. For example: fax protocol T38 fax rate voice fax-relay ecm disable fax nsf 000000 no vad

Fax Troubleshooting

If have problems sending or receiving faxes, complete the following steps:

1. Verify that your fax machine is set to a speed between 7200 and 14400.

2. Send a test fax in a controlled environment between two Linksys ATAs.

3. Determine the success rate.

4. Monitor the network and record the following statistics:

• Jitter

• Loss

• Delay

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Configuring Linksys ATA Features

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

5. If faxes fail consistently, capture a copy of the web interface settings by selecting Save As >

Web page, complete from the Web Configuration Utility page. You can send this configuration file to Technical Support.

6. Enable and capture the debug log. For instructions, refer to

”Troubleshooting and

Configuration FAQ” section on page 129

.

NOTE: You may also capture data using a sniffer trace.

7. Identify the type of Fax machine connected to the Linksys ATA.

8. Contact technical support.

If you are an end user of Linksys VoIP products, contact the reseller or Internet telephony service provider (ITSP) that supplied the equipment.

If you are an authorized Linksys Voice System partner, contact Linksys technical support.

Network Address Translation

This section describes issues that arise when using a Linksys ATA device on a network behind a network address translation (NAT) device. It includes the following topics:

•

”NAT Overview” section on page 40

•

”NAT Types” section on page 41

•

”Simple Traversal of UDP Through NAT” section on page 42

•

”SIP-NAT Interoperation” section on page 43

NAT Overview

Network Address Translation (NAT) allows multiple devices to share the same public, routable,

IP address for establishing connections over the Internet. NAT is typically performed by a router that forwards packets between the Internet and the internal, private network.

A typical application of a NAT is to allow all the devices in a subscriber home network to access the Internet through a router with a single public IP address assigned by an ISP. The IP header of the packets sent from the private network to the public network is substituted by NAT with the public IP address and a port assigned by the router. The receiver of the packets on the public network sees the packets as coming from the external address instead of the private address of the device.

The association between a private address and port and a public address and port is called a

NAT mapping. This mapping is maintained for a short period of time, that varies from a few seconds to several minutes. The expiration time is extended whenever the mapping is used to send a packet from the source device.

Linksys ATA Administration Guide 40

Configuring Linksys ATA Features

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

Private IP address

192.168.1.1

Linksys ATA NAT Device

External IP address assigned by ISP

ISP

Internet

192.168.1.100

DHCP server

ITSP

Session Border

Controller

(or outbound proxy)

The ITSP may support NAT mapping using a Session Border Controller (SBC) or an outbound proxy (see above figure). An SBC is the preferred option because it eliminates the need for managing NAT on the Linksys ATA device. If this is not available, you can set an outbound proxy.

If an outbound proxy is required, you will need to discuss with your ITSP which of the following

NAT support parameters you’ll need to set:

Parameter Tab

Outbound Proxy Line

STUN Server SIP

STUN Enable

STUN Test

Enable

SIP

SIP

Description

SIP Outbound Proxy Server where all outbound requests are sent as the first hop.

IP address or fully-qualified domain name of the

STUN server to contact for NAT mapping discovery.

Values

—

—

Enables the use of STUN to discover NAT mapping.

Select yes or no from the drop-down menu.

Yes/No

If the STUN Enable feature is enabled and a valid

STUN server is available, the Linksys ATA device can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a

Warning header in all subsequent REGISTER requests.

Yes/No

NAT Types

The different ways that NAT is implemented is sometimes divided into the following categories:

• Full cone NAT—Also known as one-to-one NAT. All requests from the same internal IP address and port are mapped to the same external IP address and port. An external host can send a packet to the internal host, by sending a packet to the mapped external address

• Restricted cone NAT—All requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host can send a packet to the internal host only if the internal host had previously sent a packet to it.

Linksys ATA Administration Guide 41

Configuring Linksys ATA Features

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

• Port restricted cone NAT/symmetric NAT—Port restricted cone NAT or symmetric NAT is like a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet to a particular port on the internal host only if the internal host had previously sent a packet from that port to the external host.

With symmetric NAT all requests from the same internal IP address and port to a specific destination IP address and port are mapped to a unique external source IP address and port. If the same internal host sends a packet with the same source address and port to a different destination, a different mapping is used. Only an external host that receives a packet can send a

UDP packet back to the internal host.

Simple Traversal of UDP Through NAT

Simple Traversal of UDP through NATs (STUN) 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: http://www.voip-info.org/wiki-Open+Source+VOIP+Software

STUN does not work with a symmetric NAT router. To determine the type of NAT your router uses, complete the following steps:

1. Enable debugging on the Linksys ATA device.

NOTE: Make sure you do not have firewall running on your PC that could block the syslog port

(by default this is 514).

2. On the Web Configuration Utility, System tab, set the Debug Server parameter to the IP address and port number of your syslog server.

NOTE: The address and port number have to be reachable from the Linksys ATA.

3. Set the Debug level parameter to 3 but you do not need to change the value of the syslog

server parameter.

4. To capture SIP signaling messages, under the Line tab, set the parameter SIP Debug Option to Full. The output is named syslog.514.log.

5. To determine the type of NAT your router is using set the parameter STUN Test Enable to yes.

6. View the syslog messages to determine if your network uses symmetric NAT or not.

Parameter

Debug Server

Tab

System

Linksys ATA Administration Guide

Description

The debug server name and port. This feature specifies the server for logging Linksys ATA device debug information. The level of detailed output depends on the debug level parameter setting.

Values

—

42

Configuring Linksys ATA Features

Using a FAX Machine (SPA2102, SPA3102 or SPA8000)

Debug Level

STUN Test

Enable

SIP Debug

Option

System

SIP

Line

The higher the debug level, the more debug information is generated. Zero (0) means no debug information is generated. To log SIP messages,

Debug Level must be set to at least 2.

The default is 0.

If the STUN Enable feature is enabled and a valid

STUN server is available, the Linksys ATA device can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a

Warning header in all subsequent REGISTER requests.

Yes/No

SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log.

For this case, set to Full.

SIP-NAT Interoperation

In the case of SIP, the addresses where messages/data should be sent to a Linksys ATA device system are embedded in the SIP messages sent by the device. If the Linksys ATA device system is sitting behind a NAT device, the private IP address assigned to it is not usable for communications with the SIP entities outside the private network.

NOTE: If the ITSP offers an outbound NAT-Aware proxy, this discovers the public IP address from the remote endpoint and eliminates the need to modify the SIP message from the UAC.

The Linksys ATA device system must substitute the private IP address information with the proper external IP address/port in the mapping chosen by the underlying NAT to communicate with a particular public peer address/port. For this, the Linksys ATA device system needs to perform the following tasks:

• Discover the NAT mappings used to communicate with the peer.

This can be done with the help of an external device, such as a STUN server. A STUN server responds to a special NAT-Mapping-Discovery request by sending back a message to the source IP address/port of the request, where the message contains the source IP address/port of the original request. The Linksys ATA device can send this request when it first attempts to communicate with a SIP entity over the Internet. It then stores the mapping discovery results returned by the server.

• Communicate the NAT mapping information to the external SIP entities.

If the entity is a SIP Registrar, the information should be carried in the Contact header that overwrites the private address/port information. If the entity is another SIP UA when establishing a call, the information should be carried in the Contact header as well as in the SDP embedded in SIP message bodies.

The VIA header in outbound SIP requests might also need to be substituted with the public address if the UAS relies on it to route back responses.

• Extend the discovered NAT mappings by sending keep-alive packets.

Because the mapping is alive only for a short period, the Linksys ATA device system continues to send periodic keep-alive packets through the mapping to extend its validity as necessary.

Linksys ATA Administration Guide 43

Configuring Linksys ATA Features

Managing Caller ID Service

Managing Caller ID Service

The choice of caller ID (CID) method is dependent on your area/region. To configure CID, use the following parameters:

Parameter Tab

Caller ID Method Regional

Caller ID FSK

Standard

Regional

Description and Value

The following choices are available:

• Bellcore (N.Amer,China)—CID, CIDCW, and VMWI. FSK sent after first ring

(same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).

• DTMF (Finland, Sweden)—CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

• DTMF (Denmark)—CID only. DTMF sentbefore first ring with no polarity reversal and no DTAS.

• ETSI DTMF—CID only. DTMF sent after DTAS (and no polarity reversal) and before first ring.

• ETSI DTMF With PR—CID only. DTMF sent after polarity reversal and DTAS and before first ring.

• ETSI DTMF After Ring—CID only. DTMF sent after first ring (no polarity reversal or DTAS).

• ETSI FSK—CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity reversal) and before first ring. Waits for ACK from CPE after DTAS for CIDCW.

• ETSI FSK With PR (UK)—CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from CPE after DTAS for

CIDCW. Polarity reversal is applied only if equipment is on hook.

• DTMF (Denmark) With PR—CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

The default is Bellcore(N.Amer, China).

The Linksys ATA device supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.23.

The default is bell 202.

This field is not found in the PAP2T.

The types of Caller ID are as follows:

• On Hook Caller ID Associated with Ringing — This type of Caller ID is used for incoming calls when the attached phone is on hook. See the following figure (a) – (c). All CID methods can be applied for this type of CID.

• On Hook Caller ID Not Associated with Ringing — This feature is used to send VMWI signal to the phone to turn the message waiting light on and off (see Figure 1 (d) and

(e)). This is available only for FSK-based CID methods: (Bellcore, ETSI FSK, and ETSI FSK

With PR).

Linksys ATA Administration Guide 44

Configuring Linksys ATA Features

Silence Suppression and Comfort Noise Generation

• Off Hook Caller ID — This is used to delivery caller-id on incoming calls when the attached phone is off hook (see the following figure). This can be call waiting caller ID

(CIDCW) or to notify the user that the far end party identity has changed or updated

(such as due to a call transfer). This is available only for FSK-based CID methods:

(Bellcore, ETSI FSK, and ETSI FSK With PR).

a) Bellcore/ETSI Onhook Post-Ring FSK

First

Ring b) ETSI Onhook Post-Ring DTMF

First

Ring c) ETSI Onhook Pre-Ring FSK/DTMF

Polarity

Reversal

CAS

(DTAS) d) Bellcore Onhook FSK w/o Ring

OSI e) ETSI Onhook FSK w/o Ring

Polarity

Reversal

CAS

(DTAS) f) Bellcore/ETSI Offhook FSK

CAS

(DTAS)

Wait For

ACK

FSK

DTMF

DTMF/

FSK

FSK

FSK

FSK

First

Ring

Silence Suppression and Comfort Noise Generation

Voice Activity Detection (VAD) with Silence Suppression is a means of increasing the number of calls supported by the network by reducing the required bandwidth for a single call. VAD uses a sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether or not speech is present. If the

VAD algorithm decides speech is not present, the silence suppression and comfort noise generation is activated. This is accomplished by removing and not transmitting the natural silence that occurs in normal two-way connection. The IP bandwidth is used only when someone is speaking. During the silent periods of a telephone call, additional bandwidth is available for other voice calls or data traffic because the silence packets are not being transmitted across the network.

Comfort Noise Generation provides artificially-generated background white noise (sounds), designed to reassure callers that their calls are still connected during silent periods. If Comfort

Noise Generation is not used, the caller may think the call has been disconnected because of the “dead silence” periods created by the VAD and Silence Suppression feature.

Silence suppression is configured in the Line and PSTN Line tabs. See

”Linksys ATA Routing Field

Reference,” on page 60

.

Linksys ATA Administration Guide 45