Administration Guide

Section 51

51

Ext SIP Port

External port to substitute for the actual SIP port of

the unit in all outgoing SIP messages. If “0” is

specified, no SIP port substitution is performed.

Port

0

Ext RTP Port Min

External port mapping of <RTP Port Min>. If this

value is non-zero, the RTP port number in all

outgoing SIP messages is substituted by the

corresponding port value in the external RTP port

range.

Port

0

NAT Mapping Enable

Enable the use of externally mapped of IP address

and SIP/RTP ports in SIP messages. The mapping

may be discovered by any of the supported

methods.

Bool

No

NAT Keep Alive Enable

If set to “yes”, the configured <NAT Keep Alive

Msg> is sent periodically every <NAT Keep Alive

Intvl> seconds.

Bool

No

NAT Keep Alive Msg

Contents of the keep-alive message to be sent to a

given destination periodically to maintain the

current NAT-mapping. It could be an empty string. If

value is $NOTIFY, a NOTIFY message is sent as

keep alive. If value is $REGISTER, a REGISTER

message w/o Contact is sent.

Str31

$NOTIFY

NAT Keep Alive Dest

Destination to send NAT keep alive messages to. If

value is $PROXY, it will be sent to the current proxy

or outbound proxy

FQDN

$PROXY

Use Outbound Proxy

Enable the use of <Outbound Proxy>. If set to “no”,

<Outbound Proxy> and <Use OB Proxy in Dialog)

is ignored.

Bool

No

Outbound Proxy

SIP Outbound Proxy Server where all outbound

requests are sent as the first hop.

FQDN

No

Use OB Proxy In Dialog Whether to force SIP requests to be sent to the

outbound proxy within a dialog. Ignored if <Use

Outbound Proxy> is “no” or <Outbound Proxy> is

empty

Bool

Yes

NAT Keep Alive Intvl

Interval between sending NAT-mapping keep alive

message in sec

Uns16

15

4.6.

Media and SDP (Session Description Protocol) Configuration

4.6.1.

DTMF and Hookflash

By default, the PHONE ADAPTER sends DTMF to the far end using RFC2833-style "AVT tones".

This method of conveying DTMF tones sends a representation of a tone (someone pressed the "7"

key) to the RTP peer as a separate RTP audio codec, but with timing information synchronized with

the speech audio codec. This method of DTMF conveyance works in most topologies, however in

some environments, the service provider may have an application server which is not in the media

path, or may be responsible for protocol conversion to a protocol or device which does not support

AVT tones.

Likewise, hookflash events by default are handled internally by the PHONE ADAPTER and used to

trigger supplementary services which are implemented on the PHONE ADAPTER. If a provider needs

to convey a hookflash event to an application server to initiate a network-oriented feature, the

PHONE ADAPTER is configurable to send these events.

52

The administrator can select a method for conveying DTMF and hookflash on a per-line basis. In

addition, the administrator can also configure the MIME type (Content-Type header) used when

conveying DTMF or hookflash in SIP INFO messages. The MIME type is set once for both lines.

DTMF Tx Method

Method to transmit DTMF signals to the far end:

Inband = Send DTMF using the audio path; INFO =

Use the SIP INFO method, AVT = Send DTMF as

AVT events; Auto = Use Inband or AVT based on

outcome of codec negotiation

Choice:

{InBand,

AVT,

INFO

Auto}

Auto

Hook Flash Tx Method

Select the method to signal Hook Flash events:

• None: do not signal hook flash events

• AVT: use RFC2833 AVT (event=16)

• INFO: use SIP INFO method with the single line

“signal = hf” in the message body. The MIME type for

this message body is taken from the <Hook Flash

MIME Type> paramter

Choice:

{None,

AVT,

INFO}

None

DTMF Relay MIME

Type

This is the MIME Type to be used in a SIP

INFO message used to signal DTMF event.

Str31

application/dtmf-relay

Hook Flash MIME

Type

This is the MIME Type to be used in a SIP

INFO message used to signal hook flash

event.

Str31

application/hook-flash

4.6.2.

Codec and Audio Settings

The following parameters are used to enable or disable access to specific codecs, echo cancellation,

and FAX support.

Parameter Name

Description

Type

Default

Preferred Codec

Select a preferred codec for all calls. However, the

actual codec used in a call still depends on the

outcome of the codec negotiation protocol.G711u,

G711a, G726-16, G726-24, G726-32, G726-40,

G729a, G723

Choice

G711u

Use Pref Codec Only

Only use the preferred codec for all calls. The call will

fail if the far end does not support this codec.

Bool

No

Silence Supp Enable

Enable silence suppression so that silent audio

frames are not transmitted

Bool

No

Echo Canc Enable

Enable the use of echo canceller

Bool

Yes

Echo Canc Adapt

Enable

Enable echo canceller to adapt

Bool

Yes

Echo Supp Enable

Enable the use of echo suppressor. If <Echo Canc

Enable> is “no”, this parameter is ignored

Bool

Yes

G729a Enable

1

Enable the use of G729a codec at 8 kbps.

Bool

Yes

G723 Enable

1

Enable the use of G723 codec at 6.3 kbps

Bool

Yes

G726-16 Enable

1

Enable the use of G726 codec at 16 kbps

Bool

Yes

G726-24 Enable

1

Enable the use of G726 codec at 24 kbps

Bool

Yes

G726-32 Enable

1

Enable the use of G726 codec at 32 kbps

Bool

Yes

G726-40 Enable

1

Enable the use of G726 codec at 40 kbps

Bool

Yes

FAX Passthru Enable

*** This parameter has been removed. ***

Bool

Yes

53

FAX CED Detect Enable Enable detection of FAX tone.

Bool

Yes

FAX CNG Detect

Enable

Bool

Yes

FAX Passthru Codec

Codec to use for fax passthru

{G711u,

G711a}

G711u

FAX Codec Symmetric

Force unit to use symmetric codec during FAX

passthru

Bool

Yes

FAX Passthru Method

Choices: None / NSE / ReINVITE

Choice

NSE

FAX Process NSE

Bool

Yes

Release Unused Codec

Bool

Yes

Notes:

1. A codec resource is considered as allocated if it has been included in the SDP codec list of an

active call, even though it eventually may not be the one chosen for the connection. So, if the G.729a

codec is enabled and included in the codec list, that resource is tied up until the end of the call

whether or not the call actually uses G.729a. If the G729a resource is already allocated and since

only one G.729a resource is allowed per PHONE ADAPTER, no other low-bit-rate codec may be

allocated for subsequent calls; the only choices are G711a and G711u. On the other hand, two

G.723.1/G.726 resources are available per PHONE ADAPTER. Therefore it is important to disable

the use of G.729a in order to guarantee the support of 2 simultaneous G.723/G.726 codec.

4.6.3.

Dynamic Payload Types and SDP Codec Names

Note: You should only need to change the payload type mappings if you are interworking with a non-

standard implementation.

Parameter Name

Description

Type

Default

NSE Dynamic Payload

1,2

NSE dynamic payload type

Uns8

100

AVT Dynamic Payload

1,2

AVT dynamic payload type

Uns8

101

G726r16 Dynamic Payload

1,2

G726-16 dynamic payload type

Uns8

98

G726r24 Dynamic Payload

1,2

G726-24 dynamic payload type

Uns8

97

G726r40 Dynamic Payload

1,2

G726-40 dynamic payload type

Uns8

96

G729b Dynamic Payload

1,2

G729b dynamic payload type

Uns8

99

Notes:

1. Valid range is 96 – 127

2. The configured dynamic payloads are used for outbound calls only where the PHONE ADAPTER

presents the SDP offer. For inbound calls with a SDP offer, PHONE ADAPTER will follow the caller’s

dynamic payload type assignments

Parameter Name

Description

Type

Default

NSE Codec Name

NSE Codec name used in SDP

Str31

NSE

AVT Codec Name

AVT Codec name used in SDP

Str31

telephone-event

G711a Codec Name

G711a Codec name used in SDP

Str31

PCMA

G711u Codec Name

G711u Codec name used in SDP

Str31

PCMU

G726r16 Codec Name

G726-16 Codec name used in SDP

Str31

G726-16

G726r24 Codec Name

G726-24 Codec name used in SDP

Str31

G726-24

G726r32 Codec Name

G726-32 Codec name used in SDP

Str31

G726-32

54

G726r40 Codec Name

G726-40 Codec name used in SDP

Str31

G726-40

G729a Codec Name

G729a Codec name used in SDP

Str31

G729a

G729b Codec Name

G729b Codec name used in SDP

Str31

G729ab

G723 Codec Name

G723 Codec name used in SDP

Str31

G723

Notes:

1. PHONE ADAPTER uses the configured codec names in its outbound SDP

2. PHONE ADAPTER ignores the codec names in incoming SDP for standard payload types (0 –

95).

3. For dynamic payload types, PHONE ADAPTER identifies the codec by the configured codec

names. Comparison is case-insensitive.

4.6.4.

Secure Media Implementation:

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

Right after the call is established in the normal way with both sides ready to stream RTP packets, the

second stage starts where 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 and embedding

in the message body of SIP INFO requests and responses with a proprietary format. If the second

stage is successful, the PHONE ADAPTER will play a special “Secure Call Indication Tone” for short

while to indicate to both parties that the call is secured and that RTP traffic in both directions are

encrypted. If the user has a CIDCW capable phone and CIDCW service is enabled, then the CID will

be updated with the information extracted from the Mini-Certificate received from the other end. The

Name field of this CID will be prepended with a ‘$’ symbol.

The second stage in setting up a secure all can be further divided into two steps. Step 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 callee 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 step 2 if the message is valid. In step 2 the caller sends the “Caller Final” message to the

callee with the following information:

-

Message ID (4B)

-

Encrypted Master Key (16B or 128b)

-

Encrypted Master Salt (16B or 128b)

With the master key and master salt encrypted with the public key from the callee’s mini-certificate.

The master key and master salt are used by both ends for the derivation of session keys for

encrypting subsequent RTP packets. The callee then responds with a Callee Final message (which is

an empty message).

A Mini-Certificate contains the following information:

-

User Name (32B)

-

User ID or Phone Number (16B)

55

- Expiration Date (12B)

- Public Key (512b or 64B)

- Signature (1024b or 512B)

The signing agent is implicit and must be the same for all PHONE ADAPTER’s that intended to

communicate securely with each other. The public key of the signing agent is pre-configured into the

PHONE ADAPTER’s by the administrator and will be used by the PHONE ADAPTER to verify the

Mini-Certificate of its peer. The Mini-Certificate is valid if a) it has not expired, and b) its signature

checks out.

User Interface

The PHONE ADAPTER can be set up such that all outbound calls are secure calls by default, or not

secure by default. If outbound calls are secure by default, user has the option to disable security

when making the next call by dialing *19 before dialing the target number. If outbound calls are not

secure by default, user has the option to make the next outbound call secure by dialing *18 before

dialing the target number. On the other hand, user cannot force inbound calls to be secure or not

secure; it is at the mercy of the caller whether he/she enables security or not for that call.

If the call successfully switches to the secure mode, both parties will hear the “Secure Call Indication

Tone” for a short while and the CID will be updated with the Name and Number extracted from the

Mini-Certificate sent by the other party, provided CIDCW service and equipment are available: the

CID Name in this case will have a ‘$’ sign inserted at the beginning. The callee should check the

name and number again to ensure the identity of the caller. The caller should also double check the

name and number of the callee to make sure this is what he/she expects. Note that the PHONE

ADAPTER will not switch to secure mode if the callee’s CID Number from its Mini-Certificate does not

agree with the user-id used in making the outbound call: the caller’s PHONE ADAPTER will perform

this check after receiving the callee’s Mini-Certificate.

Service Provider Requirements

The PHONE ADAPTER Mini-Certificate (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 who acts as the CA of the MC. The 1024-bit public key of the CA signing the MC must also

be provisioned to each subscriber. The CA public key is used by the PHONE ADAPTER to verify the

MC received from the other end. If the MC is invalid, the PHONE ADAPTER 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 hidden from the PHONE ADAPTER’s web interface like a

password.

Since the secure call establishment relies on exchange of information embedded in message bodies

of SIP INFO requests/responses, the service provider must maker sure that their infrastructure will

allow the SIP INFO messages to pass through with the message body unmodified.

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

the following syntax:

gen_mc <ca-key> <user-name> <user-id> <expire-date>

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

9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qq

56

e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZ

YTccnZ75TuTjj13qvYs=

5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/IqSrsf6scsmundY5j7Z5m

K5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3MF+zjyYrVUFdM+pXtDBxmM+f

GUfrpAuXb7/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 and must be exactly the same as the user-id used in the

INVITE when making the call, such as “14083331234”. 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 PHONE ADAPTER.

For Example:

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

Produces:

<Mini Certificate>

Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwM

DAwMDEwMTM00OvJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/xMGFfoPmDK

Gx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhxES767G0JIS+N8l

QBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/uQ/LJQlni2kwqlm7scDvll5

k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3

VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZ

OGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs=

<SRTP Private Key>

b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/e+Klm4XiiFEPaKmU8UbooxKG36SEd

Kusp0AQ==

Mini Certificate

Base64 encoded of Mini-Certificate concatenated

with the 1024-bit public key of the CA signing the

MC of all subscribers in the group.

Str508

Empty

SRTP Private Key

Base64 encoded of the 512-bit private key per

subscriber for establishment of a secure call.

Str88

Empty

4.6.5.

Outbound Call Codec Selection Codes:

The User can use additional feature codes on the PHONE ADAPTER to force or prefer specific

codecs. These codes are automatically appended to the dial-plan. There is no need to include them

explicitly in dial-plan

Parameter Name

Description

Type

Default

Prefer G711u Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*017110

Force G711u Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*027110

Prefer G711a Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*017111

Force G711a Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*027111

57

Prefer G723 Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*01723

Force G723 Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*02723

Prefer G726r16 Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*0172616

Force G726r16 Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*0272616

Prefer G726r24 Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*0172624

Force G726r24 Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*0272624

Prefer G726r32 Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*0172632

Force G726r32 Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*0272632

Prefer G726r40 Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*0172640

Force G726r40 Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*0272640

Prefer G729a Code

Dialing code will make this codec the preferred

codec for the associated call.

ActCode

*01729

Force G729a Code

Dialing code will make this codec the only

codec that can be used for the associated call.

ActCode

*02729

4.7.

Supplementary Services

Each line of the PHONE ADAPTER has settings which enable or disable each of the supplementary

services implemented directly in the PHONE ADAPTER. The expected behavior when a specific

service is enabled is described in Section 5.

The PHONE ADAPTER provides native support of a large set of enhanced or supplementary

services. All of these services are optional. The parameters listed in the following table are used to

enable or disable a specific supplementary service. A supplementary service should be disabled if a)

the user has not subscribed for it, or b) the Service Provider intends to support similar service using

other means than relying on the PHONE ADAPTER.

Parameter Name

Description

Type

Default

Call Waiting Serv

Enable Call Waiting Service

Bool

Yes

Block CID Serv

Enable Block Caller ID Service

Bool

Yes

Block ANC Serv

Enable Block Anonymous Calls Service

Bool

Yes

Dist Ring Serv

Enable Distinctive Ringing Service

Bool

Yes

Cfwd All Serv

Enable Call Forward All Service

Bool

Yes

Cfwd Busy Serv

Enable Call Forward Busy Service

Bool

Yes

Cfwd No Ans Serv

Enable Call Forward No Answer Service

Bool

Yes

Cfwd Sel Serv

Enable Call Forward Selective Service

Bool

Yes

Cfwd Last Serv

Enable Forward Last Call Service

Bool

Yes

Block Last Serv

Enable Block Last Call Service

Bool

Yes

Accept Last Serv

Enable Accept Last Call Service

Bool

Yes

DND Serv

Enable Do Not Disturb Service

Bool

Yes

CID_Serv

Enable Caller ID Service

Bool

Yes

58

CWCID Serv

Enable Call Waiting Caller ID Service

Bool

Yes

Call Return Serv

Enable Call Return Service

Bool

Yes

Call Back Serv

Enable Call Back Service

Bool

Yes

Three Way Call Serv

1

Enable Three Way Calling Service

Bool

Yes

Three Way Conf

Serv

1,2

Enable Three Way Conference Service

Bool

Yes

Attn Transfer Serv

1,2

Enable Attended Call Transfer Service

Bool

Yes

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer

Service

Bool

Yes

MWI Serv

3

Enable MWI Service

Bool

Yes

VMWI Serv

Enable VMWI Service (FSK)

Bool

Yes

Speed Dial Serv

Enable Speed Dial Service

Bool

Yes

Secure Call Serv

Enable Secure Call Service

Bool

Yes

Referral Serv

Enable Referral Service. See <Referral

Services Codes> for more details

Bool

Yes

Feature Dial Serv

Enable Feature Dial Service. See <Feature

Dial Services Codes> for more details

Bool

Yes

Notes:

1. Three Way Calling is required for Three Way Conference and Attended Transfer.

2. Three Way Conference is required for Attended Transfer.

3. MWI is available only if a Voice Mail Service is set-up in the deployment.

4.7.1.

Supplementary Services activated internally

Once Supplementary Services on the PHONE ADAPTER are Enabled, the services can be activated

or deactivated dynamically by dialing specific (configurable) dial strings. For example, the default dial

string to activate or deactivate most features is a "*" character followed by a two digit code. The

following table lists the parameters which set these dial strings used internally by the PHONE

ADAPTER. If a provider wishes to offer a service which is activated or deactivated in an application

server in their network instead of internally in the PHONE ADAPTER, the dial pattern for that service

should NOT be present in these configuration parameters.

Parameter Name

Description

Type

Default

Call Return Code

Call the last caller.

ActCode

*69

Blind Transfer Code

Blind transfer current call to the target

specified after the activation code

ActCode

*98

Cfwd All Act Code

Forward all calls to the target specified

after the activation code

ActCode

*72

Cfwd All Deact Code

Cancel call forward all

ActCode

*73

Cfwd Busy Act Code

Forward busy calls to the target specified

after the activation code

ActCode

*90

Cfwd Busy Deact Code

Cancel call forward busy

ActCode

*91

Cfwd No Ans Act Code

Forward no-answer calls to the target

specified after the activation code

ActCode

*92

Cfwd No Ans Deact Code

Cancel call forward no-answer

ActCode

*93

Cfwd Last Act Code

Forward the last inbound or outbound calls

to the target specified after the activation

code

ActCode

*63

59

Cfwd Last Deact Code

Cancel call forward last

ActCode

*83

Block Last Act Code

Block the last inbound call

ActCode

*60

Block Last Deact Code

Cancel blocking of the last inbound call

ActCode

*80

Accept Last Act Code

Accept the last outbound call. Let it ring

through when DND or Call Forward All is in

effect

ActCode

*64

Accept Last Deact Code

Cancel Accept Last

ActCode

*84

Call Back Act Code

Callback when the last outbound call is not

busy

ActCode

*66

Call Back Deact Code

Cancel callback

ActCode

*86

CW_Act_Code

Enable Call Waiting on all calls

ActCode

*56

CW_Deact_Code

Disable Call Waiting on all calls

ActCode

*57

CW_Per_Call_Act_Code

Enable Call Waiting for the next call

ActCode

*71

CW_Per_Call_Deact_Code

Disable Call Waiting for the next call

ActCode

*70

Block_CID_Act_Code

Block CID on all outbound calls

ActCode

*67

Block_CID_Deact_Code

Unblock CID on all outbound calls

ActCode

*66

Block_CID_Per_Call_Act_Code

Block CID on the next outbound call

ActCode

*81

Blcok_CID_Per_Call_Deact_Code

Unblock CID on the next inbound call

ActCode

*82

Block_ANC_Act_Code

Block all anonymous calls

ActCode

*77

Block_ANC_Deact_Code

Unblock all anonymous calls

ActCode

*87

DND_Act_Code

Enable Do Not Disturb

ActCode

*78

DND_Deact_Code

Disable Do Not Disturb

ActCode

*79

CID_Act_Code

Enable Caller-ID Generation

ActCode

*65

CID_Deact_Code

Disable Call-ID Generation

ActCode

*85

CWCID_Act_Code

Enable Call Waiting Caller-ID generation

ActCode

*25

CWCID_Deact_Code

Disable Call Waiting Caller-ID generation

ActCode

*45

Dist_Ring_Act_Code

Enable Distinctive Ringing

ActCode

*61

Dist_Ring_Deact_Code

Disable Distinctive Ringing

ActCode

*81

Speed Dial Act Code

Assign a speed dial number

ActCode

*74

Secure All Call Act Code

Make all outbound calls secure

ActCode

*16

Secure No Call Act Code

Make all outbound calls not secure

ActCode

*17

Secure One Call Act Code

Make the next outbound call secure. This

operation is redundant if all outbound calls

are secure by default.

ActCode

*18

Secure One Call Deact Code

Make the next outbound call not secure.

This operation is redundant if all outbound

calls are not secure by default.

ActCode

*19

In addition to the dynamic activation and deactivation codes, the following parameters control the

default activation or deactivation of internal parameters.

Parameter Name

Description

Type

Default

CW Setting

Call Waiting on/off by default for all calls

Bool

Yes

Block CID Setting

Block Caller ID on/off by default for all calls

Bool

No

Block ANC Setting

Block Anonymous Calls on or off

Bool

No

DND Setting

Do Not Disturb on or off

Bool

No

CID Setting

Caller ID Generation on or off

Bool

Yes

CWCID Setting

Call Waiting Caller ID Generation on or off

Bool

Yes

Dist Ring Setting

Distinctive Ring on or off

Bool

Yes

60

Secure Call Setting

If yes, all outbound calls are secure calls by default

Bool

No

4.7.2.

Call Forwarding Implemented internally

The PHONE ADAPTER supports local call forwarding services (Call Forward All, Call Forward Busy,

Call Forward No Answer, and Selective Call Forwarding for up to 8 numbers).

Parameter Name

Description

Type

Default

Cfwd All Dest

Forward number for Call Forward All Service

Phone

Cfwd Busy Dest

Forward number for Call Forward Busy Service

Phone

Cfwd No Ans Dest

Forward number for Call Forward No Answer Service

Phone

Cfwd No Ans Delay Delay in sec before Call Forward No Answer triggers

Uns8

20

Cfwd Sel1 Caller

Caller number pattern to trigger Call Forward Selective 1

PhTmplt

Cfwd Sel2 Caller

Caller number pattern to trigger Call Forward Selective 2

PhTmplt

Cfwd Sel3 Caller

Caller number pattern to trigger Call Forward Selective 3

PhTmplt

Cfwd Sel4 Caller

Caller number pattern to trigger Call Forward Selective 4

PhTmplt

Cfwd Sel5 Caller

Caller number pattern to trigger Call Forward Selective 5

PhTmplt

Cfwd Sel6 Caller

Caller number pattern to trigger Call Forward Selective 6

PhTmplt

Cfwd Sel7 Caller

Caller number pattern to trigger Call Forward Selective 7

PhTmplt

Cfwd Sel8 Caller

Caller number pattern to trigger Call Forward Selective 8

PhTmplt

Cfwd Sel1 Dest

Forward number for Call Forward Selective 1

Phone

Cfwd Sel2 Dest

Forward number for Call Forward Selective 2

Phone

Cfwd Sel3 Dest

Forward number for Call Forward Selective 3

Phone

Cfwd Sel4 Dest

Forward number for Call Forward Selective 4

Phone

Cfwd Sel5 Dest

Forward number for Call Forward Selective 5

Phone

Cfwd Sel6 Dest

Forward number for Call Forward Selective 6

Phone

Cfwd Sel7 Dest

Forward number for Call Forward Selective 7

Phone

Cfwd Sel8 Dest

Forward number for Call Forward Selective 8

Phone

Block Last Caller

ID of caller blocked via the “Block Last Caller” service

Phone

Accept Last Caller

ID of caller accepted via the “Accept Last Caller” service

Phone

Cfwd Last Caller

The Caller number that is actively forwarded to <Cfwd

Last Dest> by using the Call Forward Last activation

code

Phone

Cfwd Last Dest

Forward number for the <Cfwd Last Caller>

Phone

4.7.3.

Supplementary Services implemented in the service provider network

For services which are activated or deactivated in the service provider network (for example in an

application server), instead of internally in the PHONE ADAPTER, The

Feature_Dial_Services_Codes and Referral_Services_Codes parameters contain a list of dial strings

that correspond to feature codes in the network after which the PHONE ADAPTER needs to collect a

target number. These codes are automatically appended to the dial plan, so there is no need to

explicitly include them in the dial plan. For example, if call forwarding is implemented in the network,

the code to activate call forwarding and collect the target number should be included in the

Feature_Dial_Services_Codes parameter, but the code to deactivate call forwarding should not (since

it does not require collection of a target phone number).

Feature Dial Services Codes

61

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. Max

total length is 79 chars. This parameter applies when the user has a dial tone (1st or 2nd dial tone).

Enter *code (and the following target number according to current dial plan) entered at the dial tone

triggers the PHONE ADAPTER to call the target number prepended by the *code. For example, after

user dials *72, the PHONE ADAPTER plays a prompt tone awaiting the user to enter a valid target

number. When a complete number is entered, the PHONE ADAPTER sends a INVITE to

*72<target_number> as in a normal call. This feature allows the proxy to process features like call

forward (*72) or BLock Caller ID (*67).

Notes:

- The *codes should not conflict with any of the other vertical service codes internally processed by

the PHONE ADAPTER. You can empty the corresponding *code that you do not want to PHONE

ADAPTER to process.

- You can add a parameter to each *code in "Features Dial Services Codes" to indicate what tone to

play after the *code is entered, such as *72`c`|*67`p`. Below are a list of allowed tone parameters

(note the use of back quotes surrounding the parmeter w/o spaces)

`c` = <Cfwd Dial Tone>

`d` = <Dial Tone>

`m` = <MWI Dial Tone>

`o` = <Outside Dial Tone>

`p` = <Prompt Dial Tone>

`s` = <Second Dial Tone>

`x` = No tones are place, x is any digit not used above

If no tone parameter is specified, the PHONE ADAPTER plays Prompt tone by default.

- If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not

include it in this parameter. In that case, simply add that *code in the dial plan and the PHONE

ADAPTER will send INVITE *73@..... as usual when user dials *73.

Referral Services Codes

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, etc. Max total

length is 79 chars. This parameter applies when the user places the current call on hold (by Hook

Flash) and is listening to 2nd dial tone. Each *code (and the following valid target number according

to current dial plan) entered on the 2nd dial-tone triggers the PHONE ADAPTER to perform a blind

transfer to a target number that is prepended by the service *code. For example, after the user dials

*98, the PHONE ADAPTER plays a special dial tone called the "Prompt Tone" while waiting for the

user the enter a target number (which is checked according to dial plan as in normal dialing). When a

complete number is entered, the PHONE ADAPTER sends a blind REFER to the holding party with

the Refer-To target equals to *98<target_number>. This feature allows the PHONE ADAPTER to

"hand off" a call to an application server to perform further processing, such as call park.

Notes:

- The *codes should not conflict with any of the other vertical service codes internally processed by

the PHONE ADAPTER. You can empty the corresponding *code that you do not want to PHONE

ADAPTER to process.

4.8.

Dial Plan Configuration

The PHONE ADAPTER 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 PHONE ADAPTER 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.

62

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

•

Interdigit_Long_Timer

•

Interdigit_Short_Timer

•

Dial_Plan ([1] and [2])

•

Enable_IP_Dialing

Other timers are configurable via parameters, but do not directly pertain to the dial plan itself. They

are discussed elsewhere in this document.

Interdigit Long Timer:

ParName:

Interdigit_Long_Timer

Default:

10

The Interdigit_Long_Timer specifies the default maximum time (in seconds) allowed between dialed

digits, when no candidate digit sequence is as yet complete (see discussion of Dial_Plan parameter

for an explanation of candidate digit sequences).

Interdigit Short Timer:

ParName:

Interdigit_Short_Timer

Default:

3

The Interdigit_Short_Timer specifies the default maximum time (in seconds) allowed between dialed

digits, when at least one candidate digit sequence is complete as dialed (see discussion of Dial_Plan

parameter for an explanation of candidate digit sequences).

Dial Plan[1] and Dial Plan[2]:

ParName:

Dial_Plan[1] and Dial_Plan[2]

Default:

( *xx | [3469]11 | 0 | 00 | <:1408>[2-9]xxxxxx |

1[2-9]xx[2-9]xxxxxx | 011x. )

The Dial_Plan parameters contain the actual dial plan scripts for each of lines 1 and 2.

Dial Plan Digit Sequences:

The plans contain a series of digit sequences, separated by the ‘|’ character. The collection of

sequences is enclosed in parentheses, ‘(‘ and ‘)’.

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.

63

Any one of a set of terminating events triggers the PHONE ADAPTER to either accept the user-dialed

sequence, and transmit it to initiate a call, or else reject it as invalid. The terminating events are:

•

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 (barring is discussed later), 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 per the dial plan if complete.

The timeout duration depends on the matching state. If no candidate sequences are as yet complete

(as dialed), the Interdigit_Long_Timeout applies. If a candidate sequence is complete, but there

exists one or more incomplete candidates, then the Interdigit_Short_Timeout applies.

White space is ignored, and 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): ‘[‘ set ‘]’ (e.g. [389] means ‘3’ or ‘8’ or ‘9’)

o Numeric ranges are allowed within the brackets: digit ‘-‘ digit (e.g. [2-9] means ‘2’ or ‘3’ or

… or ‘9’)

o 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 (‘.’ character) to the element.

Hence, “01.” matches “0”, “01”, “011”, “0111”, … etc.

Subsequence Substitution:

A subsequence 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”.

Intersequence Tones:

An “outside line” dial tone can be generated within a sequence by appending a ‘,’ character 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.

64

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 timeout values.

Local Timer Overrides:

The long and short timeout 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 ‘ ‘. 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>.

These overrides are especially useful to terminate dialing in countries with predictable but variable

length numbering plans, or to provide an exception when a rule with fewer digits is known to override

a rule waiting for more digits. For example, assuming a generic international calling sequence of

011xxxxxxxxx. in North America, the PHONE ADAPTER can be configured to complete dialing to

France after the country code and exactly 10 digits using 01133xxxxxxxxxxS:0 as a dial plan digit

sequence. When this sequence matches, it overrides the short interdigit timer, causing an immediate

call. If the S:0 had been absent, the PHONE ADAPTER would wait for the short interdigit timer to

expire before placing the call.

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 PHONE ADAPTER implicitly appends the vertical code sequences entered in the Regional

parameter settings to the end of the dial plan for both line 1 and line 2. Likewise, if Enable_IP_Dialing

is enabled, then ip dialing is also accepted on the associated line.

Maximum Length

Each dial plan cannot exceed 2047 bytes, after all configured vertical codes have been added to the

Dial_Plan parameter.

Examples:

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

the area code and number.

65

( 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 5 seconds, unless a 4 digit

extension is dialed by the user.

66

( P5 <:1000> | xxxx )

Explanation of Default Dial Plan

The Default Dial Plan script for each line is:

“(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.)”

Dial Plan Entry

Functionality

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

Int’l Operator

[2-9]xxxxxx

US "local" number

1xxx[2-9]xxxxxx US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else (Int’l long distance, FWD, ...)

IP Dialing

If IP dialing is enabled, one can dial [user-id@]a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by

entering “*”, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and

255, and port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are

optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular

phone number according to the dial plan. The INVITE message, however, is still sent to the outbound

proxy if it is enabled.

4.8.1.

Speed Dialing Settings

If assigned, Speed Dials enable a user to dial a single digit from 2 through 9 and then the "#"

character, to dial the number configured in the PHONE ADAPTER. Speed dials are specified per line.

Parameter Name

Description

Type

Default

Speed Dial 2

Target phone number (or URL) assigned to speed dial “2”

Phone

Speed Dial 3

Target phone number (or URL) assigned to speed dial “3”

Phone

Speed Dial 4

Target phone number (or URL) assigned to speed dial “4”

Phone

Speed Dial 5

Target phone number (or URL) assigned to speed dial “5”

Phone

Speed Dial 6

Target phone number (or URL) assigned to speed dial “6”

Phone

Speed Dial 7

Target phone number (or URL) assigned to speed dial “7”

Phone

Speed Dial 8

Target phone number (or URL) assigned to speed dial “8”

Phone

Speed Dial 9

Target phone number (or URL) assigned to speed dial “9”

Phone

67

4.9.

Progress Tone and Ring Configuration

The progress tones and ring tones on the PHONE ADAPTER are extremely configurable. There are

18 configurable call progress tones, 8 configurable ringing cadences, and 8 configurable call waiting

cadences. Progress tones and Ring cadences are configured using FreqScipts and CadScripts

respectively (described in Section 4.1).

4.9.1.

Distinctive Ring and Other Ring Settings

Distinctive Ringing and Distinctive Call Waiting Tones can be associated with specific callers

configured directly into the PHONE ADPATER, by setting the appropriate callers in the Ring_n_Caller

parameters. The Ring_1_Caller parameter specifies which callers will trigger ring cadence 1, and so

forth. If a provider wishes to offer a distinctive ringing service by providing hints from the network, the

provider can insert an Alert-Info SIP header into incoming calls. If the value in the Alert-Info header

matches one of the strings in the Ring_n_Name set of parameters, the corresponding ring cadence

will be used.

In addition to ordinary and distinctive rings, there are number of other situations where the PHONE

ADAPTER can provide a short burst of ringing. These ring settings are described below.

Parameter Name

Description

Type

Default

Ring 1 Caller

Caller number pattern to play Distinctive Ring/CWT 1

PhTmplt

Ring 2 Caller

Caller number pattern to play Distinctive Ring/CWT 2

PhTmplt

Ring 3 Caller

Caller number pattern to play Distinctive Ring/CWT 3

PhTmplt

Ring 4 Caller

Caller number pattern to play Distinctive Ring/CWT 4

PhTmplt

Ring 5 Caller

Caller number pattern to play Distinctive Ring/CWT 5

PhTmplt

Ring 6 Caller

Caller number pattern to play Distinctive Ring/CWT 6

PhTmplt

Ring 7 Caller

Caller number pattern to play Distinctive Ring/CWT 7

PhTmplt

Ring 8 Caller

Caller number pattern to play Distinctive Ring/CWT 8

PhTmplt

Default Ring

Default ringing pattern, 1 – 8, for all callers

{1,2,3,4,

5,6,7,8}

1

Default CWT

Default CWT pattern, 1 – 8, for all callers

{1,2,3,4,

5,6,7,8}

1

Hold Reminder Ring

Ring pattern for reminder of a holding call when the

phone is on-hook

{1,2,3,4,

5,6,7,8,

None}

None

Call Back Ring

Ring pattern for call back notification

{1,2,3,4,

5,6,7,8}

None

Ring1 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 1 for the inbound call

Str31

Bellcore-r1

Ring2 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 2 for the inbound call

Str31

Bellcore-r2

Ring3 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 3 for the inbound call

Str31

Bellcore-r3

Ring4 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 4 for the inbound call

Str31

Bellcore-r4

Ring5 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 5 for the inbound call

Str31

Bellcore-r5

Ring6 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 6 for the inbound call

Str31

Bellcore-r6

Ring7 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 7 for the inbound call

Str31

Bellcore-r7

68

Ring8 Name

Name in an INVITE’s Alert-Info Header to pick

distinctive ring/CWT 8 for the inbound call

Str31

Bellcore-r8

Cfwd Ring Splash

Len

2

Duration of ring splash when a call is forwarded

(0 – 10.0s)

Time3

0

Cblk Ring Splash

Len

2

Duration of ring splash when a call is blocked (0 –

10.0s)

Time3

0

VMWI Ring Splash

Len

Duration of ring splash when new messages arrive

before the VMWI signal is applied (0 – 10.0s)

Time3

.5

VMWI Ring Policy

The parameter controls when a ring splash is played

when a the VM server sends a SIP NOTIFY message

to the PHONE ADAPTER indicating the status of the

subscriber’s mail box. 3 settings are available:

New VM Available – ring as long as there is 1 or more

unread voice mail

New VM Becomes Available – ring when the number

of unread voice mail changes from 0 to non-zero

New VM Arrives – ring when the number of unread

voice mail increases

Choice

New VM

Available

Ring On No New VM If enabled, the PHONE ADAPTER will play a ring

splash when the VM server sends SIP NOTIFY

message to the PHONE ADAPTER indicating that

there are no more unread voice mails. Some

equipment requires a short ring to precede the FSK

signal to turn off VMWI lamp

Bool

No

Notes:

1. Caller number patterns are matched from Ring 1 to Ring 8. The first match (not the closest

match) will be used for alerting the subscriber.

Parameter Name

Description

Type

Default

Ring1 Cadence

Cadence script for distinctive ring 1

CadScript

60(2/4)"

Ring2 Cadence

Cadence script for distinctive ring 2

CadScript

60(.3/.2,

1/.2,.3/4)"

Ring3 Cadence

Cadence script for distinctive ring 3

CadScript

60(.8/.4,.8/4)

Ring4 Cadence

Cadence script for distinctive ring 4

CadScript

60(.4/.2,.3/.2,.8/4)

Ring5 Cadence

Cadence script for distinctive ring 5

CadScript

60(.4/.2,.3/.2,.8/4)

Ring6 Cadence

Cadence script for distinctive ring 6

CadScript

60(.4/.2,.3/.2,.8/4)

Ring7 Cadence

Cadence script for distinctive ring 7

CadScript

60(.4/.2,.3/.2,.8/4)

Ring8 Cadence

Cadence script for distinctive ring 8

CadScript

60(.4/.2,.3/.2,.8/4)

CWT 1 Cadence

Cadence script for distinctive CWT

(Call Waiting Tone) 1

CadScript

30(.3/9.7)

CWT2 Cadence

Cadence script for distinctive CWT 2

CadScript

30(.1/.1, .1/9.7)"

CWT3 Cadence

Cadence script for distinctive CWT 3

CadScript

30(.1/.1, .1/.1,

.1/9.5)

CWT4 Cadence

Cadence script for distinctive CWT 4

CadScript

30(.1/.1, .3/.1,

.1/9.3)

CWT5 Cadence

Cadence script for distinctive CWT 5

CadScript

30(.3/.1,.1/.1,.3/9.

1)

CWT6 Cadence

Cadence script for distinctive CWT 6

CadScript

30(.1/.1, .3/.1,

.1/9.3)

CWT7 Cadence

Cadence script for distinctive CWT 7

CadScript

30(.1/.1, .3/.1,

.1/9.3)

69

CWT8 Cadence

Cadence script for distinctive CWT 8

CadScript

2.3(..3/2)

Ring Waveform

Waveform for the ringing signal

{Sinusoid,

Trapezoid}

Sinusoid

Ring Frequency

Frequency of the ringing signal. Valid values

are 10 – 100 (Hz)

Uns8

25

Ring Voltage

Ringing voltage. 60-90 (V)

Uns8

70

CWT Frequency

Frequency script of the call waiting tone. All

distinctive CWT is based on this tone.

FreqScript

440@-10

4.9.2.

Progress Tones

Most of the 18 progress tones in the PHONE ADAPTER are played automatically in response to fixed

stimuli. However, the administrator can select which SIP response codes correspond to the 4 SIT

tones.

Response Status Code Handling

SIT1 RSC

1

SIP response status code to INVITE on which

to play the SIT1 Tone

RscTmplt

SIT2 RSC

1

SIP response status code to INVITE on which

to play the SIT2 Tone

RscTmplt

SIT3 RSC

1

SIP response status code to INVITE on which

to play the SIT3 Tone

RscTmplt

SIT4 RSC

1

SIP response status code to INVITE on which

to play the SIT4 Tone

RscTmplt

The Frequencies of the actual progress tones are configurable to accommodate local and regional

conventions.

Parameter Name

Description

Type

Default

Dial Tone

1

Played when prompting the user to enter a

phone number

ToneScript

350@-19,440@-

19;10(*/0/1+2)

Second Dial Tone

An alternative to <Dial Tone> when user

tries to dial a 3-way call

ToneScript

420@-19,520@-

19;10(*/0/1+2)

Outside Dial Tone

1

An alternative to <Dial Tone> usually used

to prompt the user to enter an external

phone number (versus an internal

extension). This is triggered by a “,”

character encountered in the dial plan.

ToneScript

420@-16;10(*/0/1)

Prompt Tone

1

Played when prompting the user to enter a

call forward phone number

ToneScript

520@-19,620@-

19;10(*/0/1+2)

Busy Tone

Played when a 486 RSC is received for an

outbound call

ToneScript

480@-19,620@-

19;10(.5/.5/1+2)

Reorder Tone

1,2

Played when an outbound call has failed

or after the far end hangs up during an

established call

ToneScript

480@-19,620@-

19;10(.25/.25/1+2)

Off Hook Warning

Tone

2

Played when the subscriber does not

place the handset on the cradle properly

ToneScript

480@-

10,620@0;10(.125/

.125/1+2)

Ring Back Tone

Played for an outbound call when the far

end is ringing

ToneScript

440@-19,480@-

19;*(2/4/1+2)

70

Confirm Tone

This should be a brief tone to notify the

user that the last input value has been

accepted.

ToneScript

600@-

16;1(.25/.25/1)"

SIT1 Tone

An alternative to <Reorder Tone> played

when an error occurs while making an

outbound call. The RSC to trigger this tone

is configurable (see Section ???)

ToneScript

985@-16,1428@-

16,1777@-

16;20(.380/0/1,.380

/0/2,.380/0/3,0/4/0)

SIT2 Tone

See <SIT1 Tone>

ToneScript

914@-16,1371@-

16,1777@-

16;20(.274/0/1,.274

/0/2,.380/0/3,0/4/0)

SIT3 Tone

See <SIT1 Tone>

ToneScript

914@-16,1371@-

16,1777@-

16;20(.380/0/1,.380

/0/2,.380/0/3,0/4/0)

SIT4 Tone

See <SIT 1 Tone>

ToneScript

985@-16,1371@-

16,1777@-

16;20(.380/0/1,.274

/0/2,.380/0/3,0/4/0)

MWI Dial Tone

1

This tone is played instead of <Dial Tone>

when there are unheard messages in the

subscriber’s mail box

ToneScript

350@-19,440@-

19;2(.1/.1/1+2);10(*

/0/1+2)

Cfwd Dial Tone

Special dial tone played when call forward

all is activated

ToneScript

350@-19,440@-

19;2(.2/.2/1+2);10(*

/0/1+2)

Holding Tone

Indicate to the local user that the far end

has placed the call on hold

ToneScript

600@-

16;*(.1/.1/1,.1/.1/1,.

1/9.5/1)

Conference Tone

Plays to all parties when a 3-way

conference is in progress

ToneScript

350@-

16;30(.1/.1/1,.1/9.7/

1)

Secure Call

Indication Tone

This tone is played when a call is

successfully switched to secure mode. It

should be played only for a short while (<

30s) and at a reduced level (< -19 dBm) so

that it will not interfere with the

conversation.

ToneScript

397@-19,507@-

19;15(0/2/0,.2/.1/1,.

1/2.1/2)

Notes:

1. Reorder Tone is played automatically when <Dial Tone> or any of its alternatives times out

2. Off Hook Warning Tone (also called Howler Tone) is played when Reorder Tone times out

4.10. Less Frequently Used Paramters

4.10.1.

Advanced Protocol Parameters

Parameter Name

Description

Type

Default

SIP Parameters

Max Forward

SIP Max-Forward value. Range: 1 – 255

Uns8

70

71

Max Redirection

Number of times to allow an INVITE to be

redirected by a 3xx response to avoid an

infinite loop.

Note: This parameter currently has no effect: there is

no limit on number of redirection.

Uns8

5

Max Auth

Maximum number of times a request may be

challenged (0-255)

Uns8

2

SIP User Agent

Name

User-Agent Header to be used by the unit in

outbound requests. If empty, the header is not

included.

Str63

Linksys/

$version

SIP Server Name

Server Header to used by the unit in

responses to inbound responses. If empty,

the header is not included.

Str63

Linksys/

$version

SIP Accept

Language

Accept-Language Header to be used by the

unit.

If empty, the header is not included.

Str31

Remove Last Reg

Remove last registration before registering a

new one if value is different one.

Bool

no

Use Compact

Header

If set to yes, the PHONE ADAPTER will use

compact SIP headers in outbound SIP

messages. If set to no the PHONE ADAPTER

will use normal SIP headers.

Bool

no

SIP Timer Values (sec)

SIP T1

RFC 3261 T1 value (RTT Estimate). Range: 0

– 64 sec

Time3

.5

SIP T2

RFC 3261 T2 value (Maximum retransmit

interval for non-INVITE requests and INVITE

responses). Range: 0 – 64 sec

Time3

4

SIP T4

RFC 3261 T4 value (Maximum duration a

message will remain in the network). Range:

0 – 64 sec

Time3

5

SIP Timer B

INVITE time out value. Range: 0 – 64 sec

Time3

32

SIP Timer F

Non-INVITE time out value. Range: 0 – 64

sec

Time3

32

SIP Timer H

INVITE final response time out value. Range:

0 – 64 sec

Time3

32

SIP Timer D

ACK hang around time. Range: 0 – 64 sec

Time3

32

SIP Timer J

Non-INVITE response hang around time.

Range: 0 – 64 sec

Time3

32

INVITE Expires

INVITE request Expires header value in sec.

0 = do not include Expires header in INVITE.

Range: 0 – (2

31

– 1)

Time0

180

ReINVITE Expires

ReINVITE request Expires header value in

sec. 0 = do not include Expires header in the

request. Range: 0 – (2

31

– 1)

Time0

30

Reg Min Expires

Minimum registration expiration time allowed

from the proxy in the Expires header or as a

Contact header parameter. If proxy returns

something less this value, then the minimum

value is used.

Time0

1

Reg Max Expires

Maximum registration expiration time allowed

from the proxy in the Min-Expires header. If

Time0

7200

72

value is larger than this, then the maximum

value is used

Reg Retry Intvl

Interval to wait before the PHONE ADAPTER

retries registration again after encountering a

failure condition during last registration

Time0

30

Reg Retry Long

Interval

When Registration fails with a SIP response

code that does no match <Retry Reg RSC>,

the PHONE ADAPTER will wait for the delay

specified in this parameter before retrying. If

this parameter is 0, the PHONE ADAPTER

will stop retrying. This value should be much

larger than <Reg Retry Intvl> which should

not be 0.

Time0

1200

Response Status Code Handling

SIT1 RSC

1

SIP response status code to INVITE on which

to play the SIT1 Tone

RscTmplt

SIT2 RSC

1

SIP response status code to INVITE on which

to play the SIT2 Tone

RscTmplt

SIT3 RSC

1

SIP response status code to INVITE on which

to play the SIT3 Tone

RscTmplt

SIT4 RSC

1

SIP response status code to INVITE on which

to play the SIT4 Tone

RscTmplt

Try Backup RSC

SIP response status code on which to retry a

backup server for the current request

RscTmplt

Retry Reg RSC

Interval to wait before the PHONE ADAPTER

retries registration again after encountering a

failure condition during last registration

Time0

30

RTP Parameters

RTP Port Min

2

Minimum port number for RTP transmission

and reception

Port

16384

RTP Port Max

2

Maximum port number for RTP transmission

and reception

Port

16482

RTP Packet Size

Packet size in sec. Valid values must be

multiple of 0.01s. Range: 0.01 – 0.16

Time3

0.02

RTCP Tx Interval

4

Controls the interval (sec) to send out RTCP

sender report on an active connection.

Range: 0 – 255 (s)

Time0

0

Notes:

1. Reorder or Busy Tone will be played by default for all unsuccessful response status code

2. <RTP Port Min> and <RTP Port Max> should define a range that contains at least 4 even number

ports, such as 100 – 106

3. If inbound SIP requests contain compact headers, PHONE ADAPTER will reuse the same

compact headers when generating the response regardless the settings of the <Use Compact

Header> parameter. If inbound SIP requests contain normal headers, PHONE ADAPTER will

substitute those headers with compact headers (if defined by RFC 261) if <Use Compact

Header> parameter is set to “yes.”

4. During an active connection, the PHONE ADAPTER can be programmed to send out compound

RTCP packet on the connection. Each compound RTP packet except the last one contains a SR

(Sender Report) and a SDES.(Source Description). The last RTCP packet contains an additional

BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR

73

carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to

<User ID>@<Proxy>, NAME is set to <Display Name> (or “Anonymous” if user blocks caller ID),

and TOOL is set to the Verdor/Hardware-platform-software-version (such as Linksys/PHONE

ADAPTER2000-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the PHONE

ADAPTER’s local time, not the time reported by an NTP server. If the PHONE ADAPTER

receives a RR from the peer, it will attempt to compute the round trip delay and show it as the

<Call Round Trip Delay> value (ms) in the Info section of PHONE ADAPTER web page.

4.10.2. Additional User Account Information

Parameter Name

Description

Type

Default

Line Enable

Enable this line for service

Bool

Yes

MOH Server

2

The User ID or URL of the auto-answering SAS to

contact for MOH services. Examples: 5000,

1001@music.Linksys.com, 66.12.123.15:5061.

Note: When only a user-id is given, the current

proxy or outbound proxy will be contacted as in the

making of a regular outbound call. MOH is disabled

if this parameter is not specified (empty).

Str127

Empty

SIP Port

SIP message listening port and transmission port

Port

5060

SIP TOS/DiffServ

Value

TOS/DiffServ field value in UDP IP Packets

carrying a SIP Message

Byte

0x68

RTP TOS/DiffServ

Value

TOS/DiffServ field value in UDP IP Packets

carrying a RTP data

Byte

0xb8

SAS Enable

3

Enables the FXS Line to act as a Streaming Audio

Source (SAS). If enabled, the line cannot be used

for making outgoing calls. Instead, it auto-answers

incoming calls and streams audio RTP packets to

the calling party.

Bool

No

SAS DLG Refresh

Intvl

3

If non-zero, this is the interval at which SAS sends

out session refresh (SIP re-INVITE) messages to

detect if connection to the caller is still up. If the

caller does not respond to refresh message,

PHONE ADAPTER will terminate this call with a

SIP BYE message. The default = 0 (Session

refresh disabled)

Range = 0-255 (s)

0

SAS Inbound RTP

Sink

3

The purpose of this parameter is to work around

devices that do not play inbound RTP if the SAS

line declares itself as a “sendonly” device and tells

the client not to stream out audio. This parameter is

a FQDN or IP address of a RTP sink to be used by

the PHONE ADAPTER SAS line in the SDP of its

200 response to inbound INVITE from a client. It

will appear in the c = line and the port number and,

if specified, in the m = line of the SDP. If this value

is not specified or equal to 0, then c = 0.0.0.0 and

a=sendonly will be used in the SDP to tell the SAS

client to not to send any RTP to this SAS line. If a

non-zero value is specified, then a=sendrecv and

the SAS client will stream audio to the given

address. Special case: If the value is $IP, then the

Str63

74

SAS line’s own IP address is used in the c = line

and a=sendrecv. In that case the SAS client will

stream RTP packets to the SAS line. The default

value is [empty].

SIP Debug Option

None, 1-line, full, exclude OPTIONS, exclude

REGISTER, exclude NOTIFY, …

Choice

none

Network Jitter Level

4 settings are available: very high, high, medium,

low. This parameter affects how jitter buffer size is

adjusted in the PHONE ADAPTER. Jitter buffer

size is adjusted dynamically. The minimum jitter

buffer size is 30 ms or (10 ms + current RTP frame

size), which ever is larger, for all jitter level settings.

But the starting jitter buffer size value is larger for

higher jitter levels. This parameter controls the rate

at which to adjust the jitter buffer size to reach the

minimum. If the jitter level is set to high, then the

rate of buffer size decrement is slower (more

conservative), else faster (more aggressive).

Choice

High

SIP 100REL Enable

Enable the support or the 100rel SIP extension for

reliable transmission of provisional responses (18x)

and the use of PRACK requests.

Bool

No

Blind Attn-Xfer

Enable

If enabled, the PHONE ADAPTER performs an

attended transfer operation by terminating the

current call leg, and blind transferring the other call

leg. If disabled, the PHONE ADAPTER performs an

attended transfer by referring the other call leg to

the current call leg while maintaining both call legs.

Bool

No

Notes:

1. If proxy responded to REGISTER with a smaller Expires value, the PHONE ADAPTER will renew

registration based on this smaller value instead of the configured value. If registration failed with an

“Expires too brief” error response, the PHONE ADAPTER will retry with the value given in the Min-

Expires header in the error response.

2. MOH Notes:

• The remote party must indicate that it can receive audio while holding MOH to work. That is the SIP

2xx response from the remote party in reply to the re-INVITE from the PHONE ADAPTER to put the

call on hold must have the SDP indicate a sendrecv or recvonly attribute and the remote destination

address and port must not be 0

3. SAS Notes:

• Either or both of lines 1 and 2 can be configured as an SAS server.

• Each server can maintain up to 5 simultaneous calls. If the second line on the PHONE ADAPTER is

disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls will

receive a busy signal (SIP 486 Response).

• The streaming audio source must be off-hook for the streaming to occur. Otherwise incoming calls

will get a error response (SIP 503 Response). The SAS line will not ring for incoming calls even if the

attached equipment is on-hook

• 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 whether

any audio streaming is in progress.

75

• IVR can still be used on an SAS line, but the user needs to follow some simple steps: a) Connect a

phone to the port and make sure the phone is on-hook, b) power on the PHONE ADAPTER and c)

pick up handset and press * * * * to invoke IVR in the usual way. The idea behind this is that if the

PHONE ADAPTER boots up 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 PHONE ADAPTER boots up and finds that the SAS line is

off-hook, it will remove battery from the line since no audio session 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.

4.10.3. Per-Line Polarity Settings

Parameter Name

Description

Type

Default

Idle Polarity

Polarity before call connected

{Forward,Reverse}

Forward

Caller Conn Polarity

Polarity after outbound call connected

{Forward,Reverse}

Reverse

Callee Conn Polarity

Polarity after inbound call connected

{Forward,Reverse}

Reverse

4.10.4. Additional Timer Values (sec)

Parameter Name

Description

Type

Default

Hook Flash Timer Min

Minimum on-hook time before off-hook to

qualify as hook-flash. Less than this the on-

hook event is ignored. Range: 0.1 – 0.4 sec

Time3

0.1

Hook Flash Timer Max

Maximum on-hook time before off-hook to

qualify as hook-flash. More than this the on-

hook event is treated as on-hook (no hook-

flash event). Range: 0.4 – 1.6 sec

Time3

0.9

Callee On Hook Delay

The phone must be on-hook for at this time in

sec before the PHONE ADAPTER will tear

down the current inbound call. It does not apply

to outbound calls. Range: 0 – 255 sec

Time0

0

Reorder Delay

Delay after far end hangs up before reorder

tone is played. 0 = plays immediately, inf =

never plays. Range: 0 – 255 sec

Time0

5

Call Back Expires

Expiration time in sec of a call back activation.

Ragne: 0 – 65535 sec

Time0

1800

Call Back Retry Intvl

Call back retry interval in sec. Range: 0 – 255

sec

Time0

30

Call Back Delay

Delay after receiving the first SIP 18x response

before declaring the remote end is ringing. If a

busy response is received during this time, the

PHONE ADAPTER still considers the call as

failed and keeps on retrying.

Time3

0.5

VMWI Refresh Intvl

Interval between VMWI refresh to the CPE

Time3

0.5

Interdigit Long Timer

2

Long timeout between entering digits when

dialing. Range: 0 – 64 sec

Time0

10

Interdigit Short Timer

2

Short timeout between entering digits when

dialing. Range: 0 – 64 sec

Time0

3