Administration Guide

Section 76

76

CPC Delay

3,4

Delay in seconds after caller hangs up when

the PHONE ADAPTER will start removing the

tip-and-ring voltage to the attached equipment

of the called party.

Range= 0 to 255(s)

Resolution = 1 (s)

2

CPC Duration

3,4

Duration in seconds for which the tip-to-ring

voltage is removed after the caller hangs up.

After that tip-to-ring voltage is restored and dial

tone will apply if the attached equipment is still

off hook. CPC is disabled if this value is set to

0.

Range= 0 to 1.000 (s)

Resolution = 0.001 (s)

0 (CPC

disable

d)

Notes:

1. The Call Progress Tones and DTMF playback level are not affected by the <FXS Port Output

Gain>.

2. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used

after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. The

Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete

as dialed, but more dialed digits would match other as yet incomplete sequences.

3. PHONE ADAPTER has had polarity reversal feature since release 1.0 which can be applied to

both the caller and the callee end. This feature is generally used for answer supervision on the

caller side to signal to the attached equipment when the call has been connected (remote end

has answered) or disconnected (remote end has hung up). This feature should be disabled for

the called party (ie by using the same polarity for connected and idle state) and the CPC feature

should be used instead.

4. Without CPC enabled, reorder tone will is played after a configurable delay. If CPC is enabled,

dial tone will be played when tip-to-ring voltage is restored.

4.10.5. Miscellaneous Parameters

Parameter Name

Description

Type

Default

Set Local Date

(mm/dd/yyyy)

Setting the local date; year is optional and

can be 2-digit or 4-digit

Str10

Local Time (HH/mm/ss) Setting the local time; second is optional.

Str8

Time Zone

Number of hours to add to GMT to form local

time for caller-id generation. Choices: GMT-

12:00, GMT-11:00,…, GMT, GMT+01:00,

GMT+02:00, …, GMT+13:00

Choice

GMT-07:00

FXS Port Impedance

Electrical impedance of the FXS port.

{600,

900, 600+2.16uF,

900+2.16uF,

270+750||150nF,

220+820||120nF,

220+820||115nF,

370+620||310nF}

600

FXS Port Input Gain

Input Gain in dB. Valid values are 6.0 to –

infinity. Up to 3 decimal places

dB

-3

FXS Port Output Gain

Similar to <FXS Port Input Gain> but apply to

the output signal

dB

-3

77

DTMF Playback Level

Local DTMF playback level in dBm (up to 1

decimal place)

PwrLevel

-10.0

DTMF Playback Length

Local DTMF playback duration in ms

Time3

.1

Detect ABCD

Enable local detection of DTMF ABCD

Bool

Yes

Playback ABCD

Enable local playback of OOB DTMF ABCD

Bool

Yes

Caller ID Method

The following choices are available:

• Bellcore (N.Amer,China): CID, CIDCW,

and VMWI. FSK sent after 1st ring (same as

ETSI FSK sent after 1st ring) (no polarity

reversal or DTAS)

• DTMF (Finland,Sweden): CID only. DTMF

sent after polarity reversal (and no DTAS)

and before 1st ring

• DTMF (Denmark): CID only. DTMF sent

after polarity reversal (and no DTAS) and

before 1st ring

• ETSI DTMF: CID only. DTMF sent after

DTAS (and no polarity reversal) and before

1st ring

• ETSI DTMF With PR: CID only. DTMF sent

after polarity reversal and DTAS and before

1st ring

• ETSI DTMF After Ring: CID only. DTMF

sent after 1st ring (no polarity reversal or

DTAS)

• ETSI FSK: CID, CIDCW, and VMWI. FSK

sent after DTAS (but no polarity reversal) and

before 1st ring. Will wait 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 1st ring. Will wait for ACK

from CPE after DTAS for CIDCW. Polarity

reversal is applied only if equipment is on

hook.

Choice

Bellcore

FXS Port Power Limit

Options: 1, 2, 3, 4, 5, 6, 7, 8

Choice

3

Notes:

1. It should be noted that the choice of CID method will affect the following features:

• 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 figure below (a) – (c). All CID methods can be applied for this

type of caller-id

• On Hook Caller ID Not Associated with Ringing – This feature is used for 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 caller-id methods: “Bellcore”, “ETSI FSK”, and “ETSI FSK With PR”

• Off Hook Caller ID – This is used to delivery caller-id on incoming calls when the attached phone is

off hook. See figure below (f). 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 only available if

the caller-id method is one of “Bellcore”, “ETSI FSK”, or “ETSI FSK With PR”.

78

Polarity

Reversal

First

Ring

CAS

(DTAS)

DTMF/

FSK

Polarity

Reversal

CAS

(DTAS)

FSK

CAS

(DTAS)

Wait For

ACK

FSK

First

Ring

FSK

OSI

FSK

a) Bellcore/ETSI Onhook Post-Ring FSK

d) Bellcore Onhook FSK w/o Ring

f) Bellcore/ETSI Offhook FSK

c) ETSI Onhook Pre-Ring FSK/DTMF

e) ETSI Onhook FSK w/o Ring

DTMF

b) ETSI Onhook Post-Ring DTMF

First

Ring

Figure: PHONE ADAPTER Caller ID Delivery Architecture

79

5. Expected Feature Behavior

The PHONE ADAPTER can be configured to the custom requirements of the service provider, so that

from the subscriber’s point of view, the service behaves exactly as the service provider wishes – with

varying degrees of control left with the end user. This means that a service provider can leverage the

programmability of the PHONE ADAPTER to offer sometimes subtle yet continually valuable and

differentiated services optimized for the network environment or target market(s).

This section of the Administration Guide, describes how some of the supported basic and enhanced,

or supplementary services could be implemented. The implementations described below by no

means are the only way to achieve the desired service behavior.

To understand the specific implementation options of the below features, including parameters,

requirements and contingencies please refer the section Configuration Parameters, section Error!

Reference source not found..

5.1.

Originating a Phone Call

Service Description

Placing telephone a call to another telephone

or telephony system (IVR, conference bridge,

etc.). This is the most basic service.

User Action Required to Activate or Use

When the user picks up the handset, the

PHONE ADAPTER provides dial tone and is

ready to collect dialing information via DTMF

digits from the telephone Touchtone key pad.

Expected Call and Network Behavior

While it is possible to support overlapped

dialing within the context of SIP, the PHONE

ADAPTER collects a complete phone number

and sends the full number in a SIP INVITE

message to the proxy server for further call

processing. In order to minimize dialing delay,

the PHONE ADAPTER maintains a dial plan

and matches it against the cumulative number

entered by the user. The PHONE ADAPTER

also detects invalid phone numbers not

compatible with the dial plan and alerts the

user via a configurable tone (Reorder) or

announcement.

User Action Required to Deactivate or End

Hang-up the telephone.

5.2.

Receiving a Phone Call

Service Description

The PHONE ADAPTER can receive calls from

the PSTN or other IP Telephony subscribers

User Action Required to Activate or Use

When the telephone rings, pick up the handset

and begin talking.

Expected Call and Network Behavior

Each subscriber is assigned an E.164 ID

(phone number) so that they may be reached

80

from wired or wireless callers on the PSTN or

IP network. The PHONE ADAPTER supplies

ring voltage to the attached telephone set to

alert the user of incoming calls.

User Action Required to Deactivate or End

Hang-up the telephone.

5.3.

Caller ID

Service Description

If available, the PHONE ADAPTER supports

the generation and pass through of Caller ID

information.

User Action Required to Activate or Use

No user action required. The user’s telephone

equipment must support Caller ID to display

the caller’s name and/or number.

Expected Call and Network Behavior

In between ringing bursts, the PHONE

ADAPTER can generate a Caller-ID signal to

the attached phone when the phone is on-

hook.

As part of the INVITE message, the PHONE

ADAPTER sends the caller’s name and

number as it is configured in the profile.

User Action Required to Deactivate or End

No user action required. See CLIP and CLIR.

5.4.

Calling Line Identification Presentation (CLIP)

Service Description

Some users will elect to block their Caller ID

information for all outgoing calls. However,

there may be circumstances where sending

Caller ID information for a call is desired, i.e.

trying to reach a party that does not accept

Caller ID blocked calls.

User Action Required to Activate or Use

Lift the receiver

Listen for dial tone

Press *__

Listen for dial tone

Dial the telephone number you are calling

Expected Call and Network Behavior

Caller ID will be sent to the distant party for this

call only. Users must repeat this process at the

start of each call.

User Action Required to Deactivate or End

No action required. This service is only in

81

effect for the duration of the current call.

5.5.

Calling Line Identification Restriction (CLIR) – Caller ID Blocking

Service Description

This feature allows the user to block the

delivery of their Caller ID to the number they

are calling. This feature must be activated prior

to dialing each call and is only in effect for the

duration of each call.

User Action Required to Activate or Use

Lift the receiver

Listen for dial tone

Press *__

Listen for dial tone

Dial the telephone number you are calling

You must repeat this process at the start of

each call

Expected Call and Network Behavior

The user activates this service to hide his

Caller ID when making an outgoing call.

User Action Required to Deactivate or End

No action required. This service is only in

effect for the duration of the current call.

5.6.

Call Waiting

Service Description

The user can accept a call from a 3rd party

while engaging in an active call. The PHONE

ADAPTER shall alert the subscriber of the 2nd

incoming call by playing a call waiting tone.

User Action Required to Activate or Use

If the you choose to answer the second call

either:

Press and release your phone's switch hook

(the button you release when you take your

phone off the hook) or

Press the flash button (if your phone has one).

This puts your first call on hold and

automatically connects you to your second call.

To put your second caller back on hold and

return to your first caller, press the switch hook

or flash button again. (You can alternate

between calls as often as you like.)

Expected Call and Network Behavior

If the user is on a call when another call comes

in they will hear a series of beeps / tones

82

alerting them to the second call. The person

calling will hear normal ringing.

User Action Required to Deactivate or End

See Cancel Call Waiting.

5.7.

Disable or Cancel Call Waiting

Service Description

The PHONE ADAPTER supports disabling of

call waiting permanently or on a per call basis.

User Action Required to Activate or Use

To temporarily disable Call Waiting (for the

length of one call):

Before placing a call:

Lift Receiver

Press *__

Listen for dial tone then dial the number you

want to call.

Call Waiting is now disabled for the duration of

this call only.

To deactivate Call Waiting while on a call:

Press the switch hook or flash button briefly.

This puts the first call on hold.

Listen for three short tones and then a dial

tone.

Press *__

Listen for dial tone then return to your call by

pressing the switch hook or flash button. Call

Waiting is now disabled for the duration of this

call.

To deactivate Call Waiting while on a

permanent basis (until cancelled):

Lift the receiver

Listen for dial tone

Press *__

You will hear a confirmation tone signaling your

request to cancel Call Waiting has been

accepted.

Expected Call and Network Behavior

Callers who dial your number will receive a

busy signal or, if available, the caller will be

forwarded

to

voice

mail

or

another

predetermined forwarding number.

User Action Required to Deactivate or End

If you have cancelled Call Waiting temporarily,

83

no user action is required.

If you deactivated call waiting and wish to

reinstate the service, do the following:

Lift the receiver

Listen for dial tone

Press *__

You will hear a confirmation tone signaling your

request to cancel Call Waiting has been

accepted.

5.8.

Call-Waiting with Caller ID

Service Description

When the user is on the phone and has Call

Waiting active, the new caller’s Caller ID

information will be displayed on the users

phone display screen at the same time the user

is hearing the Call Waiting beeps / tones.

User Action Required to Activate or Use

The telephone equipment connected to the

PHONE ADAPTER must support Call-Waiting

with Caller ID.

Expected Call and Network Behavior

In between call waiting tone bursts, the

PHONE ADAPTER can generate a Caller-ID

signal to the attached phone when it is off

hook.

User Action Required to Deactivate or End

Not applicable.

5.9.

Voice Mail

Service Description

Service Providers may provide voice mail

service to their subscribers.

Users have the

ability to retrieve voice mail via the telephone

connected to the PHONE ADAPTER.

User Action Required to Activate or Use

The PHONE ADAPTER indicates that a

message is waiting by, playing stuttered dial

tone when the user picks up the handset.

To retrieve messages:

Lift the receiver

Listen for dial tone

Dial the phone number assigned to the PHONE

ADAPTER

You will be connected to the voice mail server

and prompted by a voice response system with

84

instructions to listen to your messages.

Expected Call and Network Behavior

When voice mail is available for a subscriber, a

notification message will be sent from the

Voice Mail server to the PHONE ADAPTER.

When the user dials their own phone number,

the

PHONE

ADAPTER

connects

the

subscriber their voice mail system which can

then connect them to their individual voice mail

box.

User Action Required to Deactivate or End

Follow instructions of the voice mail system or

simply hang-up the telephone.

5.10. Attendant Call Transfer

Service Description

Attendant Call Transfer lets a customer use

their Touchtone phone to send a call to any

other phone, inside or outside their business,

including a wireless phones.

User Action Required to Activate or Use

While in a call with the party to be transferred:

Press the switch hook or flash button on the

phone to place the party on hold

Listen for three short tones followed by dial

tone

Dial the number to which you will transfer the

caller

Stay on the line until the called number

answers

Announce the call

Press the switch hook or flash button adding

the held party to the call

Hang up to connect the two parties and

transfer the call

Note: You can hook flash while the 3

rd

party is

ringing to start an early conference. Then hang

up to complete the transfer without waiting for

the 3

rd

party to answer first.

Expected Call and Network Behavior

When the user presses the switch hook or flash

button, the transferee is placed on hold. When

the user successfully dials the transfer number

and the party answers the transferee can be

added to the call by pressing the switch hook

or

flash

button

creating

a

three-way

conference.

When the user hangs up the

phone the transferee and the called party

85

remain in a call.

User Action Required to Deactivate or End

Not applicable.

5.11. Unattended or “Blind” Call Transfer

Service Description

Unattended or “Blind” Call Transfer lets a

customer use their Touchtone phone to send a

call to any other phone, inside or outside their

business, including a wireless phones.

User Action Required to Activate or Use

While in a call with the party to be transferred:

Press the switch hook or flash button on the

phone to place the party on hold

Enter *__

Dial the number to which you will transfer the

caller

The call is transferred when a complete

number is entered. You will hear a short

confirmation tone, followed by regular dial tone

Expected Call and Network Behavior

When the user presses the switch hook or flash

button, the transferee is placed on hold. When

the user successfully dials the transfer number,

the transferee will automatically call the dialed

number.

User Action Required to Deactivate or End

No applicable.

5.12. Call Hold

Service Description

Call Hold lets you put a caller on hold for an

unlimited period of time. It is especially useful

on phones without the hold button. Unlike a

hold button, this feature provides access to a

dial tone while the call is being held.

User Action Required to Activate or Use

Press the switch hook or flash button on the

phone to place the first party on hold. You will

hear a dial tone.

To make another call:

Enter the new number

To return to call on hold:

Hang up and the phone set will ring with the

first call on the line (or Hook Flash again)

86

Expected Call and Network Behavior

User Action Required to Deactivate or End

Hang-up the telephone.

5.13. Three-Way Calling

Service Description

The user can originate a call to a 3rd party

while engaging in an active call.

User Action Required to Activate or Use

Press the switch hook or flash button on the

phone to place the first party on hold

Listen for three short tones followed by dial

tone

Dial the number of the 3

rd

party.

When the 3

rd

party answers you may have a

conversation with them while the other party is

on hold.

To hold a conference with the party on hold

and the 3

rd

party, simply press the switch hook

or flash button

Expected Call and Network Behavior

The PHONE ADAPTER supports up to two

calls per line. The PHONE ADAPTER can

conference two calls by bridging the 2

nd

and 3

rd

parties.

User Action Required to Deactivate or End

Hang-up the telephone.

5.14. Three-Way Ad-Hoc Conference Calling

Service Description

This feature allows the user to conference up

to two other numbers on the same line to

create a three-way call.

User Action Required to Activate or Use

If you are already on a call and wish to add a

third party:

Press the switch hook or flash button

Listen for dial tone

Dial the third party normally

When the third party number starts to ring

press the switch hook or flash button again

You now have the original caller and the third

party together with you on the same call.

If you want to initiate a new Three Way Call:

87

Call the first party in the normal manner

Follow the directions for adding a third party

(see instructions above)

Expected Call and Network Behavior

The PHONE ADAPTER can host a 3-way

conference and perform 3-way audio mixing

(without the need of an external conference

bridge device or service).

If you also have Call Transfer you can also

hang up at any time to transfer the original

caller to the third party

User Action Required to Deactivate or End

5.15. Call Return

Service Description

The PHONE ADAPTER supports a service that

allows the PHONE ADAPTER to automatically

dial the last caller’s number.

User Action Required to Activate or Use

Pick up the receiver

Listen for dial tone

Press *__ to dial back the last caller that tried

to reach you.

Expected Call and Network Behavior

This service gives the user the convenience of

recalling the last incoming call to their number

automatically.

User Action Required to Deactivate or End

No user action required

5.16. Automatic Call Back

Service Description

This feature allows the user to place a call to

the last number they tried to reach whether the

call was answered, unanswered or busy by

dialing an activation code.

User Action Required to Activate or Use

Pick up the receiver

Listen for dial tone

Press *__

Expected Call and Network Behavior

If the number called is idle the call will ring

through and complete normally. If the called

number is busy the user will hear a special

announcement and the feature will monitor the

called number for up to 30 minutes. When both

88

lines are idle, the user hears a special ring.

During the monitoring process the user can

continue to originate and receive calls without

affecting the Call Return on Busy request. Call

Return on Busy requests can be canceled by

dialing the deactivation code.

User Action Required to Deactivate or End

Lift the receiver

Listen for dial tone

Press *__

5.17. Call FWD – Unconditional

Service Description

All calls are immediately forwarded to the

designated forwarding number. The PHONE

ADAPTER will not ring or provide call waiting

when Call FWD – Unconditional is activated.

User Action Required to Activate or Use

Lift the receiver

Listen for dial tone

Press *__

Listen for dial tone and enter the telephone

number you are forwarding your call to.

Activation will be confirmed with three short

bursts of tone and your forwarding will be

activated.

Alternatively, the user can activate this feature

from a web browser interface.

Expected Call and Network Behavior

This feature allows a user the option to divert

(forward) all calls to their telephone number to

any number using the touchtone keypad of

their telephone or web browser interface. This

service is activated or deactivated from the

phone being forwarded or the web browser

interface.

User Action Required to Deactivate or End

Lift the receiver

Listen for dial tone

Press *__

You will hear a confirmation tone signaling your

change has been accepted.

Alternatively, the user can deactivate this

feature from a web browser interface.

89

5.18. Call FWD – Busy

Service Description

Calls are forwarded to the designated

forwarding number if the subscriber’s line is

busy because of the following; Primary line

already in a call, primary and secondary line in

a call or conference.

User Action Required to Activate or Use

Lift the receiver

Listen for dial tone

Press *__

Listen for dial tone and enter the telephone

number you are forwarding your call to.

Activation will be confirmed with three short

bursts of tone and your forwarding will be

activated.

Alternatively, the user can activate this feature

from a web browser interface.

Expected Call and Network Behavior

This feature allows a user the option to divert

(forward) calls to their telephone number to any

number when their phone is busy or in

conference by using the touchtone keypad of

their telephone or web browser interface. This

service is activated or deactivated from the

phone being forwarded or the web browser

interface.

User Action Required to Deactivate or End

Lift the receiver

Listen for dial tone

Press *__

You will hear a confirmation tone signaling your

change has been accepted.

Alternatively, the user can deactivate this

feature from a web browser interface.

5.19. Call FWD - No Answer

Service Description

Calls are forwarded to the designated

forwarding number after a configurable time

period elapses while the PHONE ADAPTER is

ringing and does not answer.

User Action Required to Activate or Use

Lift the receiver

Listen for dial tone

Press *__

90

Listen for dial tone and enter the telephone

number you are forwarding your call to.

Activation will be confirmed with three short

bursts of tone and your forwarding will be

activated.

Alternatively, the user can activate this feature

from a web browser interface.

Note: The forward delay is entered from the

web interface. Default is 20s

Expected Call and Network Behavior

This feature allows a user the option to divert

(forward) calls to their telephone number to any

other dialable number when their phone is not

answered by using the touchtone keypad of

their telephone or web browser interface. This

service is activated or deactivated from the

phone being forwarded or the web browser

interface.

User Action Required to Deactivate or End

Lift the receiver

Listen for dial tone

Press *__

You will hear a confirmation tone signaling your

change has been accepted.

Alternatively, the user can deactivate this

feature from a web browser interface.

5.20. Anonymous Call Blocking

Service Description

By setting the corresponding configuration

parameter on the PHONE ADAPTER, the

subscriber has the option to block incoming

calls that do not reveal the caller’s Caller ID.

User Action Required to Activate or Use

Pick up the receiver

Listen for dial tone

To Activate Press *__

Expected Call and Network Behavior

When activated by the user, callers will hear

(busy) tone.

User Action Required to Deactivate or End

To De-activate Press *__

5.21. Distinctive / Priority Ringing and Call Waiting Tone

Service Description

The PHONE ADAPTER supports a number of

91

ringing and call waiting tone patterns to be

played when incoming calls arrive. The choice

of alerting pattern to use is carried in the

incoming SIP INVITE message inserted by the

SIP Proxy Server (or other intermediate

application server in the Service Provider’s

domain).

User Action Required to Activate or Use

Pick up the receiver

Listen for dial tone

Press *__

Expected Call and Network Behavior

With this service, incoming calls from up to __

telephone numbers can be automatically

identified by distinctive ringing. A distinctive

ringing

pattern

(i.e.

short-long-short)

accompanies

incoming

calls

from

the

designated telephone numbers.

If the user is engaged in conversation and a

call from one of the designated numbers

arrives, a distinctive call waiting tone (i.e. short-

long-short) accompanies the incoming call.

Calls from other telephone numbers ring

normally.

User Action Required to Deactivate or End

5.22. Speed Calling – Up to Eight (8) Numbers or IP Addresses

Service Description

The

PHONE

ADAPTER

supports

user

programming of up to 8 long distance, local,

international or emergency numbers and/or IP

addresses for fast and easy access.

User Action Required to Activate or Use

Pick up the receiver

Listen for dial tone

Press *__

Dial the single digit code under which the

number is to be stored (2-9)

Dial the complete number to be stored just as if

you were going to dial it yourself

Listen for Confirmation tone (two short beeps)

Hang up or repeat the sequence

Note: To enter IP addresses, a graphical user

interface like a web browser must be used.

92

Expected Call and Network Behavior

Pick up the receiver

Listen for dial tone

Press single digit code assigned to the stored

number (2-9)

Press # to signal dialing complete

The number is automatically dialed normally.

User Action Required to Deactivate or End

None

6. Troubleshooting

6.1.

Call Statistics Reporting

The following lists the statistics collected by the PHONE ADAPTER during normal operation. These

statistics are presented in the PHONE ADAPTER web-page (under the “Info” tab). Line status is

reported for each line (1 and 2). Each line maintains up to 2 calls: Call 1 and 2.

System Status

Current Time

Current time and date. E.g., 10/3/2003 16:43:00

Elapsed Time

Total time elapsed since last reboot. E.g., 25 days and 18:12:36

Broadcast Pkts Sent

Total number of broadcast packets sent

Broadcast Pkts Recv

Total number of broadcast packets received

Broadcast Bytes Sent

Total number of broadcast bytes sent

Broadcast Bytes Recv

Total number of broadcast bytes received and processed

Broadcast Packets Dropped

Total number of broadcast packets received but not processed

Broadcast Bytes Dropped

Total number of broadcast bytes received but not processed

RTP Packets Sent

Total number of RTP packets sent (including redundant packets)

RTP Packets Received

Total number of RTP packets received (including redundant packets)

RTP Bytes Sent

Total number of RTP bytes sent

RTP Bytes Received

Total number of RTP bytes received

SIP Messages Sent

Total number of SIP messages sent (including retransmissions)

SIP Messages Received

Total number of SIP messages received (including retransmissions)

SIP Bytes Sent

Total number of bytes of SIP messages sent (including retransmissions)

SIP Bytes Received

Total number of bytes of SIP messages received (including retransmissions)

External IP

External IP address used for NAT mapping

Line 1/2 Status

Hook State

State of the hook switch: On or Off

Registration State

Registration state of the line: Not Registered, Registered or Failed

Last Registration At

Local time of the last successful registration

Next Registration In

Number of seconds before the next registration renewal

Message Waiting

Indicate whether new voice mails available: Yes or No

Call Back Active

Indicate whether a call back request is in progress: Yes or No

Last Called Number

The last number called

Last Caller Number

The number of the last caller

Mapped SIP Port

NAT Mapped SIP Port

Call 1/2 Status

State

State of the call: Idle, Dialing, Calling, Proceeding, Ringing, Answering,

Connected, Hold, Holding, Resuming, or Reorder

Tone

Tone playing for this call: Dial, 2

nd

Dial, Outside Dial, Ring Back, Ring,

Busy, Reorder, SIT1– 4, Call Waiting, Call Forward, Conference,

93

Prompt, Confirmation, or Message-Waiting

Encoder

Encoder in use: G711u, G711a, G726-16/24/32/40, G729a, or G729ab

Decoder

Decoder in use: G711u, G711a, G726-16/24/32/40, G729a, or G729ab

FAX

Indicate whether FAX pass-through mode has been initiated: Yes or No

Type

Indicate the call type: Inbound or Outbound

Remote Hold

Indicate whether the remote end has placed the call on hold: Yes or No

Call Back

Indicate whether the call is triggered by a call back request: Yes or No

Peer Name

Name of the peer

Peer Phone

Phone number of the peer

Duration

Duration of the call in hr/min/sec format

Packets Sent

Number of RTP packets sent

Packets Recv

Number of RTP packets received

Bytes Sent

Number of RTP bytes sent

Bytes Recv

Number of RTP bytes received

Decode Latency

Decoder latency in milliseconds

Jitter

Receiver jitter in milliseconds

Round Trip Delay

Network round trip delay (ms); available if the peer supports RTCP

Packets Lost

Total number of packets lost

Packet Error

Number of RTP packets received that are invalid

Mapped RTP Port

NAT mapped RTP port

6.2.

Enabling Logging and Debugging

The PHONE ADAPTER uses the following parameters to enable logging and debugging (both using

the syslog protocol over UDP.)

• Syslog_Server

• Debug_Server

• Debug_Level

6.3.

Error and Log Reporting

The PHONE ADAPTER Error Status Code (ESC) is used to indicate the current operation status of

the PHONE ADAPTER unit. An error state can be a relatively long transient state or a steady state.

The state is also represented by a special blinking pattern of the Status LED (next to the RJ-11 ports).

The Error Status Code is a 4 digit number. The first digit indicates the error class: 1xxx represents

normal operation states while 2xxx – 9xxx represent error states that must be fixed for the unit to

function properly. The status code values can be read from the IVR option XXX or from the PHONE

ADAPTER web-page.

6.4.

Internal Error Codes

The PHONE ADAPTER defines a number of internal error codes (X00–X99) to facilitate configuration

in providing finer control over the behavior of the unit under certain error conditions. They can be

viewed as extensions to the SIP response codes 100–699. The definitions are shown below

Error Code

Description

X00

Transport layer (or ICMP) error when sending a SIP request

X20

SIP request times out while waiting for a response

94

X40

General SIP Protocol Error (e.g., unacceptable codec in SDP in 200 and

ACK messages, or times out while waiting for ACK)

X60

Dialed number invalid according to given dial plan

6.5.

Provisioning and Upgrade result codes

The $PRVST and $UPGST macro variables expand to integer codes which report the state of a

resync or upgrade attempt. They are typically used within triggers and resync/upgrade conditions.

The values of these variables is as follows:

-1 = explicit request (resync/upgrade url or sip)

0 = just rebooted (resync only)

1 = triggered from configured trigger or rule

2 = error retry

6.6.

Table of SIP Response Codes (Error Codes)

For convenience, below is a list of SIP error codes at the time of this printing which incorporates

response codes from the IANA (Internet Assigned Numbers Authority) SIP parameter registry

(http://www.iana.org/assignments/sip-parameters), and additional response codes defined in Internet-

drafts which are implemented by the PHONE ADAPTER.

Provisional 1xx

100 Trying

180 Ringing

181 Call Is Being Forwarded

182 Queued

183 Session Progress

Successful 2xx

200 OK

202 Accepted

Redirection 3xx

300 Multiple Choices

301 Moved Permanently

302 Moved Temporarily

305 Use Proxy

380 Alternative Service

Request Failure 4xx

400 Bad Request

401 Unauthorized

402 Payment Required

403 Forbidden

404 Not Found

405 Method Not Allowed

406 Not Acceptable

407 Proxy Authentication Required

408 Request Timeout

95

410 Gone

412 Conditional Request Failed

413 Request Entity Too Large

414 Request-URI Too Long

415 Unsupported Media Type

416 Unsupported URI Scheme

420 Bad Extension

421 Extension Required

423 Interval Too Brief

429 Provide Referrer Identity

480 Temporarily Unavailable

481 Call/Transaction Does Not Exist

482 Loop Detected

483 Too Many Hops

484 Address Incomplete

485 Ambiguous

486 Busy Here

487 Request Terminated

488 Not Acceptable Here

489 Bad Event

491 Request Pending

493 Undecipherable

494 Security Agreement Required

Server Failure 5xx

500 Server Internal Error

501 Not Implemented

502 Bad Gateway

503 Service Unavailable

504 Server Time-out

505 Version Not Supported

513 Message Too Large

580 Precondition Failure

Global Failures 6xx

600 Busy Everywhere

603 Decline

604 Does Not Exist Anywhere

606 Not Acceptable

7. Summary of Implemented Features and Specifications

The PHONE ADAPTER is a full featured, fully programmable phone adapter that can be custom

provisioned within a wide range of configuration parameters. The below feature descriptions are

written as a high-level overview to provide a basic understanding of the feature breadth and

capabilities of the PHONE ADAPTER. To understand the specific implementation of the below

features, including parameters, requirements and contingencies please refer the section PHONE

ADAPTER Feature Configuration Parameters, section Error! Reference source not found..

7.1.

Data Networking Features

7.1.1.

MAC Address (IEEE 802.3)

96

7.1.2.

IPv4 – Internet Protocol Version 4 (RFC 791) upgradeable to v6 (RFC 1883)

7.1.3.

ARP – Address Resolution Protocol

7.1.4.

DNS – A Record (RFC 1706), SRV Record (RFC 2782)

7.1.5.

DiffServ (RFC 2475) and ToS – Type of Service (RFC 791/1349)

7.1.6.

DHCP Client – Dynamic Host Configuration Protocol (RFC 2131)

7.1.7.

ICMP – Internet Control Message Protocol (RFC792)

7.1.8.

TCP – Transmission Control Protocol (RFC793)

7.1.9.

UDP – User Datagram Protocol (RFC768)

7.1.10. RTP – Real Time Protocol (RFC 1889) (RFC 1890)

7.1.11. RTCP – Real Time Control Protocol (RFC 1889)

7.2.

Voice Features

7.2.1.

SIPv2 – Session Initiation Protocol Version 2 (RFC 3261-3265)

7.2.1.1. SIP Proxy Redundancy – Static or Dynamic via DNS SRV

In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server.

An average SIP proxy server may handle tens of thousands subscribers. It is important that a backup

server is available so that an active server can be temporarily switched out for maintenance. The

PHONE ADAPTER supports the use of backup SIP proxy servers so that service disruption should be

next to non-existent.

Static Redundancy:

A relatively simple way to support proxy redundancy is to configure a static list of SIP proxy servers to

the PHONE ADAPTER in its configuration profile where the list is arranged in some order of priority.

The PHONE ADAPTER will attempt to contact the highest priority proxy server whenever possible.

When the currently selected proxy server is not responding, the PHONE ADAPTER automatically

retries the next proxy server in the list.

Dynamic Redundancy:

The dynamic nature of SIP message routing makes the use of a static list of proxy servers inadequate

in some scenarios. In deployments where user agents are served by different domains, for instance, it

would not be feasible to configure one static list of proxy servers per covered domain into an PHONE

ADAPTER. One solution to this situation is through the use DNS SRV records. The PHONE

ADAPTER can be instructed to contact a SIP proxy server in a domain named in SIP messages. The

PHONE ADAPTER shall consult the DNS server to get a list of hosts in the given domain that

provides SIP services. If an entry exists, the DNS server will return a SRV record which contains a list

of SIP proxy servers for the domain, with their host names, priority, listening ports, etc. The PHONE

ADAPTER shall try to contact the list of hosts in the order of their stated priority.

7.2.1.2. Re-registration with Primary SIP Proxy Server

If the PHONE ADAPTER is currently using a lower priority proxy server, it should periodically probe

the higher priority proxy to see if it is back on line and attempt to switch back to the higher priority

proxy whenever possible. It is very important that switching proxy server should not affect calls that

are already in progress.

7.2.1.3. SIP Support in Network Address Translation Networks – NAT

7.2.2.

Codec Name Assignment

97

Negotiation of the optimal voice codec is sometimes dependent on the PHONE ADAPTER device’s

ability to “match” a codec name with the far-end device/gateway codec name.

The PHONE

ADAPTER allows the network administrator to individually name the various codecs that are

supported such that the correct codec successfully negotiates with the far end the equipment.

7.2.3.

Secure Calls

A user (if enabled by service provider or administrator) has the option to make an outbound call

secure in the sense that the audio packets in both directions are encrypted.

7.2.4.

Voice Algorithms:

7.2.4.1. G.711 (A-law and mµ-law)

This very low complexity codec supports uncompressed 64 kbps digitized voice transmission at one

through ten 5 ms voice frames per packet. This codec provides the highest voice quality and uses the

most bandwidth of any of the available codecs.

7.2.4.2. G.726

This low complexity codec supports compressed 16, 24, 32 and 40 kbps digitized voice transmission

at one through ten 10 ms voice frames per packet. This codec provides the high voice quality.

7.2.4.3. G.729A

The ITU G.729 voice coding algorithm is used to compress digitized speech. Linksys supports

G.729. G.729A is a reduced complexity version of G.729. It requires about half the processing power

to code G.729. The G.729 and G.729A bit streams are compatible and interoperable, but not

identical.

7.2.4.4. G.723.1

The PHONE ADAPTER supports the use of ITU G.723.1 audio codec at 6.4 kbps. Up to 2 channels

of G.723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1

simultaneously, or Line 1 or Line 2 can initiate a 3-way conference with both call legs using G.723.1.

7.2.5.

Codec Selection

The administrator can select which low-bit-rate codec to be used for each line. G711a and G711u

are always enabled.

7.2.6.

Dynamic Payload

When no static payload value is assigned per RFC 1890, the PHONE ADAPTER can support

dynamic payloads for G.726.

7.2.7.

Adjustable Audio Frames Per Packet

This feature allows the user to set the number of audio frames contained in one RTP packet. Packets

can be adjusted to contain from 1 – 10 audio frames. Increasing the number of packets decreases the

bandwidth utilized – but it also increases delay and may affect voice quality.

7.2.8.

Fax Tone Detection Pass-Through

Users can connect a fax terminal to the PHONE ADAPTER telephone port(s). Fax terminals transmit

a single tone when they answer a call. The PHONE ADAPTER detects the type of equipment in use

on the basis of its answer tone. When it detects the equipment answering the call, the PHONE

ADAPTER performs a switchover from the current audio codec to G.711 codec.

7.2.9.

DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO *)

98

The PHONE ADAPTER may relay DTMF digits as out-of-band events to preserve the fidelity of the

digits. This can enhance the reliability of DTMF transmission required by many IVR applications such

as dial-up banking and airline information.

7.2.10. Call Progress Tone Generation

The PHONE ADAPTER has configurable call progress tones. Parameters for each type of tone may

include number of frequency components, frequency and amplitude of each component, and cadence

information.

7.2.11. Call Progress Tone Pass Through

This feature allows the user to hear the call progress tones (such as ringing) that are generated from

the far-end network.

7.2.12. Jitter Buffer – Dynamic (Adaptive)

The PHONE ADAPTER can buffer incoming voice packets to minimize out-of-order packet arrival.

This process is known as jitter buffering. The Jitter Buffer size will proactively adjust or adapt in size

depending on changing network conditions.

The PHONE ADAPTER has a Network Jitter Level control setting for each line of service. The jitter

level decides how aggressively the PHONE ADAPTER will try to shrink the jitter buffer over time to

achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is

lower, it shrinks more quickly.

7.2.13. Full Duplex Audio

Full-duplex is the ability to communicate in two directions simultaneously so that more than one

person can speak at a time. Half-duplex means that only one person can talk at a time – like a CB

radio or walkie-talkie, which is unnatural in normal free-flowing two-way communications. The

PHONE ADAPTER supports full-duplex audio.

7.2.14. Echo Cancellation – Up to 8 ms Echo Tail

The PHONE ADAPTER supports hybrid line echo cancellation. This feature uses the G.165 echo

canceller to eliminate up to 8 ms of line echo. This feature does not provide acoustic echo

cancellation on endpoint devices – that is, an end user’s speakerphone.

7.2.15. Voice Activity Detection with Silence Suppression & Comfort Noise Generation

Voice Activity Detection (VAD) and Silence Suppression is a means of increasing the number of calls

supported by the network by reducing the required bi-directional bandwidth for a single call. VAD

uses a very sophisticated algorithm to distinguish between speech and non-speech signals. Based

upon 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 2-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 since 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.

7.2.16. Attenuation / Gain Adjustment

7.2.17. Signaling Hook Flash Event

99

The PHONE ADAPTER can signal hook flash events to the remote party on a connected call. This

feature can be used to provide advanced mid-call services with third-party-call-control. Depending on

the features that the service provider will offer using third-party-call-control, the following three

PHONE ADAPTER features may be disabled to correctly signal a hook-flash event to the softswitch:

1. Call Waiting Service

2. Three Way Call Service

3. Three Way Conf Service

7.2.18. Configurable Flash / Switch Hook Timer

7.2.19. Configurable Dial Plan with Interdigit Timers

The PHONE ADAPTER has three configurable interdigit timers:

• Initial timeout (T) = handset off hook, no digit pressed yet.

• Long timeout (L) = one or more digits pressed, more digits needed to reach a valid number

(as per the dial plan).

• Short timeout (S) = current dialed number is valid, but more digits would also lead to a valid

number.

7.2.20. Message Waiting Indicator Tones – MWI

7.2.21. Polarity Control

The PHONE ADAPTER allows the polarity to be set when a call is connected and when a call is

disconnected. This feature is required to support some pay phone system and answering machines.

7.2.22. Calling Party Control – CPC

CPC signals to the called party equipment that the calling party has hung up during a connected call

by removing the voltage between the tip and ring momentarily. This feature is useful for auto-answer

equipment which then knows when to disengage.

7.2.23. International Caller ID Delivery

In addition to support of the Bellcore (FSK) and Swedish/Danish (DTMF) methods of Caller ID (CID)

delivery, release 2.0 adds a large subset of ETSI compliant methods to support international CID

equipment. The figure below shows the CID/CIDCW architecture used in the PHONE ADAPTER.

Different flavors of CID delivery method can be obtained by mixing-and-matching some of the steps

as shown.

It should be noted that the choice of CID method will affect the following features:

• 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 Figure 1 (a) – (c). All PHONE ADAPTER CID methods can be

applied for this type of caller-id

• On Hook Caller ID Not Associated with Ringing – In the PHONE ADAPTER this feature is used for

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 caller-id methods: “Bellcore”, “ETSI FSK”, and “ETSI FSK With

PR”

• Off Hook Caller ID – This is used to delivery caller-id on incoming calls when the attached phone is

off hook (see Figure 1 (f)). 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 only available if the

caller-id method is one of “Bellcore”, “ETSI FSK”, or “ETSI FSK With PR”.

100

Polarity

Reversal

First

Ring

CAS

(DTAS)

DTMF/

FSK

Polarity

Reversal

CAS

(DTAS)

FSK

CAS

(DTAS)

Wait For

ACK

FSK

First

Ring

FSK

OSI

FSK

a) Bellcore/ETSI Onhook Post-Ring FSK

d) Bellcore Onhook FSK w/o Ring

f) Bellcore/ETSI Offhook FSK

c) ETSI Onhook Pre-Ring FSK/DTMF

e) ETSI Onhook FSK w/o Ring

DTMF

b) ETSI Onhook Post-Ring DTMF

First

Ring

PHONE ADAPTER Caller ID Delivery Architecture

7.2.24. Streaming Audio Server – SAS

This feature allows one to attach an audio source to one of the PHONE ADAPTER FXS ports and

use it as a streaming audio source device. The corresponding Line (1 or 2) can be configured as a

streaming audio server (SAS) such that when the Line is called, the PHONE ADAPTER answers the

call automatically and starts streaming audio to the calling party provided the FXS port is off-hook. If

the FXS port is on-hook when the incoming call arrives, the PHONE ADAPTER replies with a SIP 503

response code to indicate “Service Not Available.” If an incoming call is auto-answered, but later the

FXS port becomes on-hook, the PHONE ADAPTER 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 PHONE ADAPTER replies with a SIP 486 response code to indicate “Busy Here”.

The SAS line can be setup to refresh each streaming audio session periodically (via SIP re-INVITE)

to detect if the connection to the caller is down. If the caller does not respond to the refresh message,

the SAS line will terminate the call so that the streaming resource can be used for other callers.

7.2.25. Music On Hold – MOH

On a connected call, the PHONE ADAPTER may place the remote party on call (the only way to do

this on te PHONE ADAPTER is to perform a hook-flash to initiate a 3-way call or to swap 2 calls

during call-waiting). If the remote party indicates that they can still receive audio while the call is

holding, the PHONE ADAPTER can be setup to contact an auto-answering SAS as described in

Section 4 and have it stream audio to the holding party. When used this way, the SAS is referred to

as a MOH Server.

101

MSA

CD Player,

Radio, etc.

Line

In

Phone 1

Phone 2

Phone 1

Phone 2

IP

Network

IP

Network

PA1:

IP=192.168.2.100

User ID[1]=1001, SIP Port[1]=5060

User ID[2]=1002, SIP Port[2]=5061

PA2:

IP=192.168.2.200

User ID[1]=2001, SIP Port[1]=5060

User ID[2]=2002, SIP Port[2]=5061

Example configuration for MOH application with a PHONE ADAPTER line configured as a SAS

SAS Configuration Examples:

The following configuration examples are based on the setup as depicted in Figure.

Example 1: SAS Line not registered with the Proxy Server for the other subscribers

On PHONE ADAPTER 1:

SAS Enable[1] = no

MOH Server [1] = 1002@192.168.2.100:5061 or 1002@127.0.0.1:5061

SAS Enable[2] = yes

On PHONE ADAPTER 2:

SAS Enable[1] = no

MOH Server [1] = 1002@192.168.2.100:5061

SAS Enable[2] = no

MOH Server [2] = 1002@192.168.2.100:5061

Example 2: SAS Line registered with the Proxy Server as the other subscribers

On PHONE ADAPTER 1:

SAS Enable[1] = no

MOH Server [1] = 1002

102

SAS Enable[2] = yes

On PHONE ADAPTER 2:

SAS Enable[1] = no

MOH Server [1] = 1002

SAS Enable[2] = no

MOH Server [2] = 1002

7.3.

Security Features

7.3.1.

Multiple Administration Layers (Levels and Permissions)

7.3.2.

HTTP Digest – Encrypted Authentication via MD5 (RFC 1321)

7.3.3.

HTTPS with Client Certificate

7.4.

Administration and Maintenance Features

7.4.1.

Web Browser Administration and Configuration via Integral Web Server

7.4.2.

Telephone Key Pad Configuration with Interactive Voice Prompts

7.4.3.

Automated Provisioning & Upgrade via TFTP, HTTP and HTTPS

7.4.4.

Periodic Notification of Upgrade Availability via NOTIFY or HTTP

7.4.5.

Non-Intrusive, In-Service Upgrades

7.4.6.

Report Generation and Event Logging

The PHONE ADAPTER reports a variety of status and error reports to assist service providers to

diagnose problems and evaluate the performance of their services. The information can be queried

by an authorized agent (using HTTP with digested authentication, for instance). The information may

be organized as an XML page or HTML page.

7.4.7.

Syslog and Debug Server Records

The PHONE ADAPTER supports detailed logging of all activities for further debugging. The debug

information may be sent to a configured Syslog server. Via the configuration parameters, the PHONE

ADAPTER allows some settings to select which type of activity/events should be logged – for

instance, a debug level setting.

8. List of all configuration parameters

Below is a list of all the configuration parameters for this software version (2.0.9). To obtain this list for

another version of software, run the profile compiler utility (spc).

# ***

# *** Linksys PHONE ADAPTER Series Configuration Parameters

# ***

# *** System Configuration

Restricted_Access_Domains

"" ;

Enable_Web_Server

"Yes" ;

Web_Server_Port

"80" ;

103

Enable_Web_Admin_Access

"Yes" ;

Admin_Passwd

"" ;

User_Password

! "" ;

# *** Internet Connection Type

DHCP

! "Yes" ;

Static_IP

! "" ;

NetMask

! "" ;

Gateway

! "" ;

# *** Optional Network Configuration

HostName

! "" ;

Domain

! "" ;

Primary_DNS

! "" ;

Secondary_DNS

! "" ;

DNS_Server_Order

"Manual" ; # options:

Manual/Manual,DHCP/DHCP,Manual

DNS_Query_Mode

"Parallel" ; # options: Parallel/Sequential

Syslog_Server

"" ;

Debug_Server

"" ;

Debug_Level

"0" ; # options: 0/1/2/3

Primary_NTP_Server

"" ;

Secondary_NTP_Server

"" ;

# *** Configuration Profile

Provision_Enable

"Yes" ;

Resync_On_Reset

"Yes" ;

Resync_Random_Delay

"2" ;

Resync_Periodic

"3600" ;

Resync_Error_Retry_Delay

"3600" ;

Forced_Resync_Delay

"14400" ;

Resync_From_SIP

"Yes" ;

Resync_After_Upgrade_Attempt

"Yes" ;

Resync_Trigger_1

"" ;

Resync_Trigger_2

"" ;

Resync_Fails_On_FNF

"No" ;

Profile_Rule

"/init.cfg" ;

Profile_Rule_B

"" ;

Profile_Rule_C

"" ;

Profile_Rule_D

"" ;

Log_Resync_Request_Msg

"$PN $MAC -- Requesting resync

$SCHEME://$SERVIP:$PORT$PATH" ;

Log_Resync_Success_Msg

"$PN $MAC -- Successful resync

$SCHEME://$SERVIP:$PORT$PATH" ;

Log_Resync_Failure_Msg

"$PN $MAC -- Resync failed: $ERR" ;

# *** Firmware Upgrade

Upgrade_Enable

"Yes" ;

Upgrade_Error_Retry_Delay

"3600" ;

Downgrade_Rev_Limit

"" ;

Upgrade_Rule

"" ;

Log_Upgrade_Request_Msg

"$PN $MAC -- Requesting upgrade

$SCHEME://$SERVIP:$PORT$PATH" ;

Log_Upgrade_Success_Msg

"$PN $MAC -- Successful upgrade

$SCHEME://$SERVIP:$PORT$PATH -- $ERR" ;

Log_Upgrade_Failure_Msg

"$PN $MAC -- Upgrade failed: $ERR" ;

# *** General Purpose Parameters

GPP_A

"" ;

GPP_B

"" ;

GPP_C

"" ;

GPP_D

"" ;

GPP_E

"" ;

GPP_F

"" ;

104

GPP_G

"" ;

GPP_H

"" ;

GPP_I

"" ;

GPP_J

"" ;

GPP_K

"" ;

GPP_L

"" ;

GPP_M

"" ;

GPP_N

"" ;

GPP_O

"" ;

GPP_P

"" ;

GPP_SA

"" ;

GPP_SB

"" ;

GPP_SC

"" ;

GPP_SD

"" ;

# *** SIP Parameters

Max_Forward

"70" ;

Max_Redirection

"5" ;

Max_Auth

"2" ;

SIP_User_Agent_Name

"$VERSION" ;

SIP_Server_Name

"$VERSION" ;

SIP_Accept_Language

"" ;

DTMF_Relay_MIME_Type

"application/dtmf-relay" ;

Hook_Flash_MIME_Type

"application/hook-flash" ;

Remove_Last_Reg

"No" ;

Use_Compact_Header

"No" ;

# *** SIP Timer Values (sec)

SIP_T1

".5" ;

SIP_T2

"4" ;

SIP_T4

"5" ;

SIP_Timer_B

"32" ;

SIP_Timer_F

"32" ;

SIP_Timer_H

"32" ;

SIP_Timer_D

"32" ;

SIP_Timer_J

"32" ;

INVITE_Expires

"240" ;

ReINVITE_Expires

"30" ;

Reg_Min_Expires

"1" ;

Reg_Max_Expires

"7200" ;

Reg_Retry_Intvl

"30" ;

Reg_Retry_Long_Intvl

"1200" ;

# *** Response Status Code Handling

SIT1_RSC

"" ;

SIT2_RSC

"" ;

SIT3_RSC

"" ;

SIT4_RSC

"" ;

Try_Backup_RSC

"" ;

Retry_Reg_RSC

"" ;

# *** RTP Parameters

RTP_Port_Min

"16384" ;

RTP_Port_Max

"16482" ;

RTP_Packet_Size

"0.030" ;

Max_RTP_ICMP_Err

"0" ;

RTCP_Tx_Interval

"0" ;

# *** SDP Payload Types

NSE_Dynamic_Payload

"100" ;

AVT_Dynamic_Payload

"101" ;

G726r16_Dynamic_Payload

"98" ;

G726r24_Dynamic_Payload

"97" ;

G726r40_Dynamic_Payload

"96" ;

105

G729b_Dynamic_Payload

"99" ;

NSE_Codec_Name

"NSE" ;

AVT_Codec_Name

"telephone-event" ;

G711u_Codec_Name

"PCMU" ;

G711a_Codec_Name

"PCMA" ;

G726r16_Codec_Name

"G726-16" ;

G726r24_Codec_Name

"G726-24" ;

G726r32_Codec_Name

"G726-32" ;

G726r40_Codec_Name

"G726-40" ;

G729a_Codec_Name

"G729a" ;

G729b_Codec_Name

"G729ab" ;

G723_Codec_Name

"G723" ;

# *** NAT Support Parameters

Handle_VIA_received

"No" ;

Handle_VIA_rport

"No" ;

Insert_VIA_received

"No" ;

Insert_VIA_rport

"No" ;

Substitute_VIA_Addr

"No" ;

Send_Resp_To_Src_Port

"No" ;

STUN_Enable

"No" ;

STUN_Test_Enable

"No" ;

STUN_Server

"" ;

EXT_IP

"" ;

EXT_RTP_Port_Min

"" ;

NAT_Keep_Alive_Intvl

"15" ;

# ***

Line_Enable[1]

"Yes" ;

# *** Streaming Audio Server (SAS)

SAS_Enable[1]

"No" ;

SAS_DLG_Refresh_Intvl[1]

"30" ;

SAS_Inbound_RTP_Sink[1]

"" ;

# *** NAT Settings

NAT_Mapping_Enable[1]

"No" ;

NAT_Keep_Alive_Enable[1]

"No" ;

NAT_Keep_Alive_Msg[1]

"$NOTIFY" ;

NAT_Keep_Alive_Dest[1]

"$PROXY" ;

# *** Network Settings

SIP_TOS/DiffServ_Value[1]

"0x68" ;

Network_Jitter_Level[1]

"high" ; # options: low/medium/high/very high

RTP_TOS/DiffServ_Value[1]

"0xb8" ;

# *** SIP Settings

SIP_Port[1]

"5060" ;

SIP_100REL_Enable[1]

"No" ;

EXT_SIP_Port[1]

"" ;

Auth_Resync-Reboot[1]

"Yes" ;

SIP_Debug_Option[1]

"none" ; # options: none/1-line/1-line excl.

OPT/1-line excl. NTFY/1-line excl. REG/1-line excl. OPT|NTFY|REG/full/full excl.

OPT/full excl. NTFY/full excl. REG/full excl. OPT|NTFY|REG

# *** Call Feature Settings

Blind_Attn-Xfer_Enable[1]

"No" ;

MOH_Server[1]

"" ;

Xfer_When_Hangup_Conf[1]

"Yes" ;

# *** Proxy and Registration

106

Proxy[1]

"" ;

Use_Outbound_Proxy[1]

"No" ;

Outbound_Proxy[1]

"" ;

Use_OB_Proxy_In_Dialog[1]

"Yes" ;

Register[1]

"Yes" ;

Make_Call_Without_Reg[1]

"No" ;

Register_Expires[1]

"3600" ;

Ans_Call_Without_Reg[1]

"No" ;

Use_DNS_SRV[1]

"No" ;

DNS_SRV_Auto_Prefix[1]

"No" ;

Proxy_Fallback_Intvl[1]

"3600" ;

Voice_Mail_Server[1]

"" ;

# *** Subscriber Information

Display_Name[1]

"" ;

User_ID[1]

"" ;

Password[1]

"" ;

Use_Auth_ID[1]

"No" ;

Auth_ID[1]

"" ;

Mini_Certificate[1]

"" ;

SRTP_Private_Key[1]

"" ;

# *** Supplementary Service Subscription

Call_Waiting_Serv[1]

"Yes" ;

Block_CID_Serv[1]

"Yes" ;

Block_ANC_Serv[1]

"Yes" ;

Dist_Ring_Serv[1]

"Yes" ;

Cfwd_All_Serv[1]

"Yes" ;

Cfwd_Busy_Serv[1]

"Yes" ;

Cfwd_No_Ans_Serv[1]

"Yes" ;

Cfwd_Sel_Serv[1]

"Yes" ;

Cfwd_Last_Serv[1]

"Yes" ;

Block_Last_Serv[1]

"Yes" ;

Accept_Last_Serv[1]

"Yes" ;

DND_Serv[1]

"Yes" ;

CID_Serv[1]

"Yes" ;

CWCID_Serv[1]

"Yes" ;

Call_Return_Serv[1]

"Yes" ;

Call_Back_Serv[1]

"Yes" ;

Three_Way_Call_Serv[1]

"Yes" ;

Three_Way_Conf_Serv[1]

"Yes" ;

Attn_Transfer_Serv[1]

"Yes" ;

Unattn_Transfer_Serv[1]

"Yes" ;

MWI_Serv[1]

"Yes" ;

VMWI_Serv[1]

"Yes" ;

Speed_Dial_Serv[1]

"Yes" ;

Secure_Call_Serv[1]

"Yes" ;

Referral_Serv[1]

"Yes" ;

Feature_Dial_Serv[1]

"Yes" ;

# *** Audio Configuration

Preferred_Codec[1]

"G711u" ; # options: G711u/G711a/G726-16/

G726-24/G726-32/G726-40/G729a/G723

Silence_Supp_Enable[1]

"No" ;

Use_Pref_Codec_Only[1]

"No" ;

Echo_Canc_Enable[1]

"Yes" ;

G729a_Enable[1]

"Yes" ;

Echo_Canc_Adapt_Enable[1]

"Yes" ;

G723_Enable[1]

"Yes" ;

Echo_Supp_Enable[1]

"Yes" ;

G726-16_Enable[1]

"Yes" ;

FAX_CED_Detect_Enable[1]

"Yes" ;

G726-24_Enable[1]

"Yes" ;

FAX_CNG_Detect_Enable[1]

"Yes" ;

G726-32_Enable[1]

"Yes" ;

FAX_Passthru_Codec[1]

"G711u" ; # options: G711u/G711a

107

G726-40_Enable[1]

"Yes" ;

FAX_Codec_Symmetric[1]

"Yes" ;

DTMF_Tx_Method[1]

"Auto" ; # options: InBand/AVT/INFO/Auto

FAX_Passthru_Method[1]

"NSE" ; # options: None/NSE/ReINVITE

Hook_Flash_Tx_Method[1]

"None" ; # options: None/AVT/INFO

FAX_Process_NSE[1]

"Yes" ;

Release_Unused_Codec[1]

"Yes" ;

# *** Dial Plan

Dial_Plan[1] "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ;

Enable_IP_Dialing[1]

"No" ;

# *** FXS Port Polarity Configuration

Idle_Polarity[1]

"Forward" ; # options: Forward/Reverse

Caller_Conn_Polarity[1]

"Forward" ; # options: Forward/Reverse

Callee_Conn_Polarity[1]

"Forward" ; # options: Forward/Reverse

# *** Call Forward Settings

Cfwd_All_Dest[1]

! "" ;

Cfwd_Busy_Dest[1]

! "" ;

Cfwd_No_Ans_Dest[1]

! "" ;

Cfwd_No_Ans_Delay[1]

! "20" ;

# *** Selective Call Forward Settings

Cfwd_Sel1_Caller[1]

! "" ;

Cfwd_Sel1_Dest[1]

! "" ;

Cfwd_Sel2_Caller[1]

! "" ;

Cfwd_Sel2_Dest[1]

! "" ;

Cfwd_Sel3_Caller[1]

! "" ;

Cfwd_Sel3_Dest[1]

! "" ;

Cfwd_Sel4_Caller[1]

! "" ;

Cfwd_Sel4_Dest[1]

! "" ;

Cfwd_Sel5_Caller[1]

! "" ;

Cfwd_Sel5_Dest[1]

! "" ;

Cfwd_Sel6_Caller[1]

! "" ;

Cfwd_Sel6_Dest[1]

! "" ;

Cfwd_Sel7_Caller[1]

! "" ;

Cfwd_Sel7_Dest[1]

! "" ;

Cfwd_Sel8_Caller[1]

! "" ;

Cfwd_Sel8_Dest[1]

! "" ;

Cfwd_Last_Caller[1]

! "" ;

Cfwd_Last_Dest[1]

! "" ;

Block_Last_Caller[1]

! "" ;

Accept_Last_Caller[1]

! "" ;

# *** Speed Dial Settings

Speed_Dial_2[1]

! "" ;

Speed_Dial_3[1]

! "" ;

Speed_Dial_4[1]

! "" ;

Speed_Dial_5[1]

! "" ;

Speed_Dial_6[1]

! "" ;

Speed_Dial_7[1]

! "" ;

Speed_Dial_8[1]

! "" ;

Speed_Dial_9[1]

! "" ;

# *** Supplementary Service Settings

CW_Setting[1]

! "Yes" ;

Block_CID_Setting[1]

! "No" ;

Block_ANC_Setting[1]

! "No" ;

DND_Setting[1]

! "No" ;

CID_Setting[1]

! "Yes" ;

CWCID_Setting[1]

! "Yes" ;

Dist_Ring_Setting[1]

! "Yes" ;

108

Secure_Call_Setting[1]

"No" ;

# *** Distinctive Ring Settings

Ring1_Caller[1]

! "" ;

Ring2_Caller[1]

! "" ;

Ring3_Caller[1]

! "" ;

Ring4_Caller[1]

! "" ;

Ring5_Caller[1]

! "" ;

Ring6_Caller[1]

! "" ;

Ring7_Caller[1]

! "" ;

Ring8_Caller[1]

! "" ;

# *** Ring Settings

Default_Ring[1]

! "1" ; # options: 1/2/3/4/5/6/7/8

Default_CWT[1]

! "1" ; # options: 1/2/3/4/5/6/7/8

Hold_Reminder_Ring[1]

! "8" ; # options: 1/2/3/4/5/6/7/8/none

Call_Back_Ring[1]

! "7" ; # options: 1/2/3/4/5/6/7/8

Cfwd_Ring_Splash_Len[1]

! "0" ;

Cblk_Ring_Splash_Len[1]

! "0" ;

VMWI_Ring_Splash_Len[1]

! ".5" ;

VMWI_Ring_Policy[1]

"New VM Available" ; # options: New VM

Available/New VM Becomes Available/New VM Arrives

Ring_On_No_New_VM[1]

"No" ;

# ***

Line_Enable[2]

"Yes" ;

# *** Streaming Audio Server (SAS)

SAS_Enable[2]

"No" ;

SAS_DLG_Refresh_Intvl[2]

"30" ;

SAS_Inbound_RTP_Sink[2]

"" ;

# *** NAT Settings

NAT_Mapping_Enable[2]

"No" ;

NAT_Keep_Alive_Enable[2]

"No" ;

NAT_Keep_Alive_Msg[2]

"$NOTIFY" ;

NAT_Keep_Alive_Dest[2]

"$PROXY" ;

# *** Network Settings

SIP_TOS/DiffServ_Value[2]

"0x68" ;

Network_Jitter_Level[2]

"high" ; # options: low/medium/high/very high

RTP_TOS/DiffServ_Value[2]

"0xb8" ;

# *** SIP Settings

SIP_Port[2]

"5061" ;

SIP_100REL_Enable[2]

"No" ;

EXT_SIP_Port[2]

"" ;

Auth_Resync-Reboot[2]

"Yes" ;

SIP_Debug_Option[2]

"none" ; # options: none/1-line/1-line excl.

OPT/1-line excl. NTFY/1-line excl. REG/1-line excl. OPT|NTFY|REG/full/full excl.

OPT/full excl. NTFY/full excl. REG/full excl. OPT|NTFY|REG

# *** Call Feature Settings

Blind_Attn-Xfer_Enable[2]

"No" ;

MOH_Server[2]

"" ;

Xfer_When_Hangup_Conf[2]

"Yes" ;

# *** Proxy and Registration

Proxy[2]

"" ;

Use_Outbound_Proxy[2]

"No" ;

109

Outbound_Proxy[2]

"" ;

Use_OB_Proxy_In_Dialog[2]

"Yes" ;

Register[2]

"Yes" ;

Make_Call_Without_Reg[2]

"No" ;

Register_Expires[2]

"3600" ;

Ans_Call_Without_Reg[2]

"No" ;

Use_DNS_SRV[2]

"No" ;

DNS_SRV_Auto_Prefix[2]

"No" ;

Proxy_Fallback_Intvl[2]

"3600" ;

Voice_Mail_Server[2]

"" ;

# *** Subscriber Information

Display_Name[2]

"" ;

User_ID[2]

"" ;

Password[2]

"" ;

Use_Auth_ID[2]

"No" ;

Auth_ID[2]

"" ;

Mini_Certificate[2]

"" ;

SRTP_Private_Key[2]

"" ;

# *** Supplementary Service Subscription

Call_Waiting_Serv[2]

"Yes" ;

Block_CID_Serv[2]

"Yes" ;

Block_ANC_Serv[2]

"Yes" ;

Dist_Ring_Serv[2]

"Yes" ;

Cfwd_All_Serv[2]

"Yes" ;

Cfwd_Busy_Serv[2]

"Yes" ;

Cfwd_No_Ans_Serv[2]

"Yes" ;

Cfwd_Sel_Serv[2]

"Yes" ;

Cfwd_Last_Serv[2]

"Yes" ;

Block_Last_Serv[2]

"Yes" ;

Accept_Last_Serv[2]

"Yes" ;

DND_Serv[2]

"Yes" ;

CID_Serv[2]

"Yes" ;

CWCID_Serv[2]

"Yes" ;

Call_Return_Serv[2]

"Yes" ;

Call_Back_Serv[2]

"Yes" ;

Three_Way_Call_Serv[2]

"Yes" ;

Three_Way_Conf_Serv[2]

"Yes" ;

Attn_Transfer_Serv[2]

"Yes" ;

Unattn_Transfer_Serv[2]

"Yes" ;

MWI_Serv[2]

"Yes" ;

VMWI_Serv[2]

"Yes" ;

Speed_Dial_Serv[2]

"Yes" ;

Secure_Call_Serv[2]

"Yes" ;

Referral_Serv[2]

"Yes" ;

Feature_Dial_Serv[2]

"Yes" ;

# *** Audio Configuration

Preferred_Codec[2]

"G711u" ; # options: G711u/G711a/G726-16/

G726-24/G726-32/G726-40/G729a/G723

Silence_Supp_Enable[2]

"No" ;

Use_Pref_Codec_Only[2]

"No" ;

Echo_Canc_Enable[2]

"Yes" ;

G729a_Enable[2]

"Yes" ;

Echo_Canc_Adapt_Enable[2]

"Yes" ;

G723_Enable[2]

"Yes" ;

Echo_Supp_Enable[2]

"Yes" ;

G726-16_Enable[2]

"Yes" ;

FAX_CED_Detect_Enable[2]

"Yes" ;

G726-24_Enable[2]

"Yes" ;

FAX_CNG_Detect_Enable[2]

"Yes" ;

G726-32_Enable[2]

"Yes" ;

FAX_Passthru_Codec[2]

"G711u" ; # options: G711u/G711a

G726-40_Enable[2]

"Yes" ;

FAX_Codec_Symmetric[2]

"Yes" ;

110

DTMF_Tx_Method[2]

"Auto" ; # options: InBand/AVT/INFO/Auto

FAX_Passthru_Method[2]

"NSE" ; # options: None/NSE/ReINVITE

Hook_Flash_Tx_Method[2]

"None" ; # options: None/AVT/INFO

FAX_Process_NSE[2]

"Yes" ;

Release_Unused_Codec[2]

"Yes" ;

# *** Dial Plan

Dial_Plan[2] "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ;

Enable_IP_Dialing[2]

"No" ;

# *** FXS Port Polarity Configuration

Idle_Polarity[2]

"Forward" ; # options: Forward/Reverse

Caller_Conn_Polarity[2]

"Forward" ; # options: Forward/Reverse

Callee_Conn_Polarity[2]

"Forward" ; # options: Forward/Reverse

# *** Call Forward Settings

Cfwd_All_Dest[2]

! "" ;

Cfwd_Busy_Dest[2]

! "" ;

Cfwd_No_Ans_Dest[2]

! "" ;

Cfwd_No_Ans_Delay[2]

! "20" ;

# *** Selective Call Forward Settings

Cfwd_Sel1_Caller[2]

! "" ;

Cfwd_Sel1_Dest[2]

! "" ;

Cfwd_Sel2_Caller[2]

! "" ;

Cfwd_Sel2_Dest[2]

! "" ;

Cfwd_Sel3_Caller[2]

! "" ;

Cfwd_Sel3_Dest[2]

! "" ;

Cfwd_Sel4_Caller[2]

! "" ;

Cfwd_Sel4_Dest[2]

! "" ;

Cfwd_Sel5_Caller[2]

! "" ;

Cfwd_Sel5_Dest[2]

! "" ;

Cfwd_Sel6_Caller[2]

! "" ;

Cfwd_Sel6_Dest[2]

! "" ;

Cfwd_Sel7_Caller[2]

! "" ;

Cfwd_Sel7_Dest[2]

! "" ;

Cfwd_Sel8_Caller[2]

! "" ;

Cfwd_Sel8_Dest[2]

! "" ;

Cfwd_Last_Caller[2]

! "" ;

Cfwd_Last_Dest[2]

! "" ;

Block_Last_Caller[2]

! "" ;

Accept_Last_Caller[2]

! "" ;

# *** Speed Dial Settings

Speed_Dial_2[2]

! "" ;

Speed_Dial_3[2]

! "" ;

Speed_Dial_4[2]

! "" ;

Speed_Dial_5[2]

! "" ;

Speed_Dial_6[2]

! "" ;

Speed_Dial_7[2]

! "" ;

Speed_Dial_8[2]

! "" ;

Speed_Dial_9[2]

! "" ;

# *** Supplementary Service Settings

CW_Setting[2]

! "Yes" ;

Block_CID_Setting[2]

! "No" ;

Block_ANC_Setting[2]

! "No" ;

DND_Setting[2]

! "No" ;

CID_Setting[2]

! "Yes" ;

CWCID_Setting[2]

! "Yes" ;

Dist_Ring_Setting[2]

! "Yes" ;

Secure_Call_Setting[2]

"No" ;

111

# *** Distinctive Ring Settings

Ring1_Caller[2]

! "" ;

Ring2_Caller[2]

! "" ;

Ring3_Caller[2]

! "" ;

Ring4_Caller[2]

! "" ;

Ring5_Caller[2]

! "" ;

Ring6_Caller[2]

! "" ;

Ring7_Caller[2]

! "" ;

Ring8_Caller[2]

! "" ;

# *** Ring Settings

Default_Ring[2]

! "1" ; # options: 1/2/3/4/5/6/7/8

Default_CWT[2]

! "1" ; # options: 1/2/3/4/5/6/7/8

Hold_Reminder_Ring[2]

! "8" ; # options: 1/2/3/4/5/6/7/8/none

Call_Back_Ring[2]

! "7" ; # options: 1/2/3/4/5/6/7/8

Cfwd_Ring_Splash_Len[2]

! "0" ;

Cblk_Ring_Splash_Len[2]

! "0" ;

VMWI_Ring_Splash_Len[2]

! ".5" ;

VMWI_Ring_Policy[2]

"New VM Available" ; # options: New VM

Available/New VM Becomes Available/New VM Arrives

Ring_On_No_New_VM[2]

"No" ;

# *** Call Progress Tones

Dial_Tone

"350@-19,440@-19;10(*/0/1+2)" ;

Second_Dial_Tone

"420@-19,520@-19;10(*/0/1+2)" ;

Outside_Dial_Tone

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

Prompt_Tone

"520@-19,620@-19;10(*/0/1+2)" ;

Busy_Tone

"480@-19,620@-19;10(.5/.5/1+2)" ;

Reorder_Tone

"480@-19,620@-19;10(.25/.25/1+2)" ;

Off_Hook_Warning_Tone

"480@-10,620@0;10(.125/.125/1+2)" ;

Ring_Back_Tone

"440@-19,480@-19;*(2/4/1+2)" ;

Confirm_Tone

"600@-16;1(.25/.25/1)" ;

SIT1_Tone

"985@-16,1428@-16,1777@-16;

20(.380/0/1,.380/0/2,.380/0/3,0/4/0)" ;

SIT2_Tone

"914@-16,1371@-16,1777@-16;

20(.274/0/1,.274/0/2,.380/0/3,0/4/0)" ;

SIT3_Tone

"914@-16,1371@-16,1777@-16;

20(.380/0/1,.380/0/2,.380/0/3,0/4/0)" ;

SIT4_Tone

"985@-16,1371@-16,1777@-16;

20(.380/0/1,.274/0/2,.380/0/3,0/4/0)" ;

MWI_Dial_Tone

"350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)" ;

Cfwd_Dial_Tone

"350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)" ;

Holding_Tone

"600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)" ;

Conference_Tone

"350@-19;20(.1/.1/1,.1/9.7/1)" ;

Secure_Call_Indication_Tone

"397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)" ;

# *** Distinctive Ring Patterns

Ring1_Cadence

"60(2/4)" ;

Ring2_Cadence

"60(.3/.2,1/.2,.3/4)" ;

Ring3_Cadence

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

Ring4_Cadence

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

Ring5_Cadence

"60(.2/.2,.2/.2,.2/.2,1/4)" ;

Ring6_Cadence

"60(.2/.4,.2/.4,.2/4)" ;

Ring7_Cadence

"60(.4/.2,.4/.2,.4/4)" ;

Ring8_Cadence

"60(0.25/9.75)" ;

# *** Distinctive Call Waiting Tone Patterns

CWT1_Cadence

"30(.3/9.7)" ;

CWT2_Cadence

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

CWT3_Cadence

"30(.1/.1, .3/.1, .1/9.3)" ;

CWT4_Cadence

"30(.1/.1,.1/.1,.1/9.5)" ;

CWT5_Cadence

"30(.3/.1,.1/.1,.3/9.1)" ;

CWT6_Cadence

"30(.1/.1,.3/.2,.3/9.1)" ;

CWT7_Cadence

"30(.3/.1,.3/.1,.1/9.1)" ;

112

CWT8_Cadence

"2.3(.3/2)" ;

# *** Distinctive Ring/CWT Pattern Names

Ring1_Name

"Bellcore-r1" ;

Ring2_Name

"Bellcore-r2" ;

Ring3_Name

"Bellcore-r3" ;

Ring4_Name

"Bellcore-r4" ;

Ring5_Name

"Bellcore-r5" ;

Ring6_Name

"Bellcore-r6" ;

Ring7_Name

"Bellcore-r7" ;

Ring8_Name

"Bellcore-r8" ;

# *** Ring and Call Waiting Tone Spec

Ring_Waveform

"Sinusoid" ; # options: Sinusoid/Trapezoid

Ring_Frequency

"25" ;

Ring_Voltage

"70" ;

CWT_Frequency

"440@-10" ;

# *** Control Timer Values (sec)

Hook_Flash_Timer_Min

".1" ;

Hook_Flash_Timer_Max

".9" ;

Callee_On_Hook_Delay

"0" ;

Reorder_Delay

"5" ;

Call_Back_Expires

"1800" ;

Call_Back_Retry_Intvl

"30" ;

Call_Back_Delay

".5" ;

VMWI_Refresh_Intvl

"30" ;

Interdigit_Long_Timer

"10" ;

Interdigit_Short_Timer

"3" ;

CPC_Delay

"2" ;

CPC_Duration

"0" ;

# *** Vertical Service Activation Codes

Call_Return_Code

"*69" ;

Blind_Transfer_Code

"*98" ;

Call_Back_Act_Code

"*66" ;

Call_Back_Deact_Code

"*86" ;

Cfwd_All_Act_Code

"*72" ;

Cfwd_All_Deact_Code

"*73" ;

Cfwd_Busy_Act_Code

"*90" ;

Cfwd_Busy_Deact_Code

"*91" ;

Cfwd_No_Ans_Act_Code

"*92" ;

Cfwd_No_Ans_Deact_Code

"*93" ;

Cfwd_Last_Act_Code

"*63" ;

Cfwd_Last_Deact_Code

"*83" ;

Block_Last_Act_Code

"*60" ;

Block_Last_Deact_Code

"*80" ;

Accept_Last_Act_Code

"*64" ;

Accept_Last_Deact_Code

"*84" ;

CW_Act_Code

"*56" ;

CW_Deact_Code

"*57" ;

CW_Per_Call_Act_Code

"*71" ;

CW_Per_Call_Deact_Code

"*70" ;

Block_CID_Act_Code

"*67" ;

Block_CID_Deact_Code

"*68" ;

Block_CID_Per_Call_Act_Code

"*81" ;

Block_CID_Per_Call_Deact_Code

"*82" ;

Block_ANC_Act_Code

"*77" ;

Block_ANC_Deact_Code

"*87" ;

DND_Act_Code

"*78" ;

DND_Deact_Code

"*79" ;

CID_Act_Code

"*65" ;

CID_Deact_Code

"*85" ;

CWCID_Act_Code

"*25" ;

CWCID_Deact_Code

"*45" ;

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Dist_Ring_Act_Code

"*26" ;

Dist_Ring_Deact_Code

"*46" ;

Speed_Dial_Act_Code

"*74" ;

Secure_All_Call_Act_Code

"*16" ;

Secure_No_Call_Act_Code

"*17" ;

Secure_One_Call_Act_Code

"*18" ;

Secure_One_Call_Deact_Code

"*19" ;

Referral_Services_Codes

"" ;

Feature_Dial_Services_Codes

"" ;

# *** Outbound Call Codec Selection Codes

Prefer_G711u_Code

"*017110" ;

Force_G711u_Code

"*027110" ;

Prefer_G711a_Code

"*017111" ;

Force_G711a_Code

"*027111" ;

Prefer_G723_Code

"*01723" ;

Force_G723_Code

"*02723" ;

Prefer_G726r16_Code

"*0172616" ;

Force_G726r16_Code

"*0272616" ;

Prefer_G726r24_Code

"*0172624" ;

Force_G726r24_Code

"*0272624" ;

Prefer_G726r32_Code

"*0172632" ;

Force_G726r32_Code

"*0272632" ;

Prefer_G726r40_Code

"*0172640" ;

Force_G726r40_Code

"*0272640" ;

Prefer_G729a_Code

"*01729" ;

Force_G729a_Code

"*02729" ;

# *** Miscellaneous

Set_Local_Date_(mm/dd)

"" ;

Set_Local_Time_(HH/mm)

"" ;

Time_Zone

"GMT-07:00" ; # options: GMT-12:00/

GMT-11:00/GMT-10:00/GMT-09:00/GMT-08:00/GMT-07:00/GMT-06:00/GMT-05:00/

GMT-04:00/GMT-03:30/GMT-03:00/GMT-02:00/GMT-01:00/GMT/GMT+01:00/

GMT+02:00/GMT+03:00/GMT+03:30/GMT+04:00/GMT+05:00/GMT+05:30/GMT+05:45/

GMT+06:00/GMT+06:30/GMT+07:00/GMT+08:00/GMT+09:00/GMT+09:30/GMT+10:00/

GMT+11:00/GMT+12:00/GMT+13:00

FXS_Port_Impedance

"600" ; # options: 600/900/600+2.16uF/

900+2.16uF/270+750||150nF/220+820||120nF/220+820||115nF/370+620||310nF

FXS_Port_Input_Gain

"-3" ;

FXS_Port_Output_Gain

"-3" ;

DTMF_Playback_Level

"-16" ;

DTMF_Playback_Length

".1" ;

Detect_ABCD

"Yes" ;

Playback_ABCD

"Yes" ;

Caller_ID_Method

"Bellcore(N.Amer,China)" ; # options:

Bellcore(N.Amer,China)/DTMF(Finland,Sweden)/DTMF(Denmark)/ETSI DTMF/

ETSI DTMF With PR/ETSI DTMF After Ring/ETSI FSK/ETSI FSK With PR(UK)

FXS_Port_Power_Limit

"3" ; # options: 1/2/3/4/5/6/7/8

Protect_IVR_FactoryReset

"No" ;

9. Acronyms

A/D

Analog To Digital Converter

ANC

Anonymous Call

B2BUA

Back to Back User Agent

Bool

Boolean Values. Specified as “yes” and “no”, or “1” and “0” in the profile

CA

Certificate Authority

CAS

CPE Alert Signal

CDR

Call Detail Record

CID

Caller ID

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CIDCW

Call Waiting Caller ID

CNG

Comfort Noise Generation

CPC

Calling Party Control

CPE

Customer Premises Equipment

CWCID

Call Waiting Caller ID

CWT

Call Waiting Tone

D/A

Digital to Analog Converter

dB

decibel

dBm

dB with respect to 1 milliwatt

DHCP

Dynamic Host Configuration Protocol

DNS

Domain Name Server

DRAM

Dynamic Random Access Memory

DSL

Digital Subscriber Loop

DSP

Digital Signal Processor

DTAS

Data Terminal Alert Signal (same as CAS)

DTMF

Dual Tone Multiple Frequency

ETSI

European Telecommunication Standard

FQDN

Fully Qualified Domain Name

FSK

Frequency Shift Keying

FXS

Foreign eXchange Station

GW

Gateway

ITU

International Telecommunication Union

HTML

Hypertext Markup Language

HTTP

Hypertext Transfer Protocol

HTTPS

HTTP over SSL

ICMP

Internet Control Message Protocol

IGMP

Internet Group Management Protocol

ILEC

Incumbent Local Exchange Carrier

IP

Internet Protocol

ISP

Internet Service Provider

ITSP

IP Telephony Service Provider

IVR

Interactive Voice Response

LAN

Local Area Network

LBR

Low Bit Rate

LBRC

Low Bit Rate Codec

MC

Mini-Certificate

MGCP

Media Gateway Control Protocol

MOH

Music On Hold

MOS

Mean Opinion Score (1-5, the higher the better)

ms

Millisecond

MSA

Music Source Adaptor

MWI

Message Waiting Indication

OSI

Open Switching Interval

PCB

Printed Circuit Board

PR

Polarity Reversal

PS

Provisioning Server

PSQM

Perceptual Speech Quality Measurement (1-5, the lower the better)

PSTN

Public Switched Telephone Network

NAT

Network Address Translation

OOB

Out-of-band

REQT

(SIP) Request Message

RESP

(SIP) Response Message

RSC

(SIP) Response Status Code, such as 404, 302, 600

RTP

Real Time Protocol

115

RTT

Round Trip Time

SAS

Streaming Audio Server

SDP

Session Description Protocol

SDRAM

Synchronous DRAM

sec

seconds

SIP

Session Initiation Protocol

SLIC

Subscriber Line Interface Circuit

SP

Service Provider

PAP2

Phone Adaptor Ports 2 (Linksys Phone Adaptor)

SSL

Secure Socket Layer

TFTP

Trivial File Transfer Protocol

TCP

Transmission Control Protocol

UA

User Agent

uC

Micro-controller

UDP

User Datagram Protocol

URL

Uniform Resource Locator

VM

Voice Mail

VMWI

Visual Message Waiting Indication/Indicator

VQ

Voice Quality

WAN

Wide Area Network

XML

Extensible Markup Language

10. Glossary

ACD (Automatic Call Distribution): A switching system designed to allocate incoming calls to certain

positions or agents in the order received and to hold calls not ready to be handled (often with a

recorded announcement).

Area Code: A 3-digit code used in North America to identify a specific geographic telephone location.

The first digit can be any number between 2 and 9. The second and third digits can be any number.

Billing Increment: The division by which the call is rounded. In the field it is common to see full-minute

billing on the local invoice while 6-second rounding is the choice of most long-distance providers that

bill their customers directly.

Blocked Calls: Caused by an insufficient network facility that does not have enough lines to allow

calls to reach a given destination. May also pertain to a call from an originating number that is

blocked by the receiving telephone number.

Bundled Service: Offering various services as a complete package.

Call Completion: The point at which a dialed number is answered.

Call Termination: The point at which a call is disconnected.

CDR (Call Detail Records): A software program attached to a VoIP/telephone system that records

information about the telephone number’s activity.

Carrier’s Carrier: Companies that build fiber optic and microwave networks primarily selling to

resellers and carriers. Their main focus is on the wholesale and not the retail market.

Casual Access: Casual Access is when customers choose not to use their primary carriers to process

the long-distance call being made. The customer dials the carrier’s 101XXXX number.

CO (Central Office): Switching center for the local exchange carrier.

Centrex: This service is offered by the LEC to the end user. The feature-rich Centrex line offers the

same features and benefits as a PBX to a customer without the capital investment or maintenance

charges. The LEC charges a monthly fee to the customer, who must agree to sign a term agreement.

116

Circuits: The communication path(s) that carry calls between two points on a network.

Customer Premise Equipment: The only part of the telecommunications system that the customer

comes into direct contact with. Example of such pieces of equipment are: telephones, key systems,

PBXs, voicemail systems and call accounting systems as well as wiring telephone jacks. The

standard for this equipment is set by the FCC, and the equipment is supplied by an interconnect

company.

Dedicated Access: Customers have direct access to the long-distance provider via a special circuit

(T1 or private lines). The circuit is hardwired from the customer site to the POP and does not pass

through the LEC switch. The dial tone is provided from the long-distance carrier.

Dedicated Access Line (DAL): Provided by the local exchange carrier. An access line from the

customer’s telephone equipment directly to the long-distance company’s switch or POP.

Demarcation Point: This is where the LEC’s ownership and responsibility (wiring, equipment) ends

and the customer’s responsibilities begin.

Direct Inward Dialing (DID): Allows an incoming call to bypass the attendant and ring directly to an

extension. Available on most PBX systems and a feature of Centrex service.

Dual Tone Multifrequency (DTMF): Better known as the push button keypad. DTMF replaces dial

pulses with electronically produced tones for network signaling.

Enhanced Service: Services that are provided in addition to basic long distance and accessed by way

of a touchtone phone through a series of menus.

Exchange Code (NXX): The first three digits of a phone number.

Flat-rate Pricing: The customer is charged one rate (sometimes two rates, one for peak and one for

off-peak) rather than a mileage-sensitive program rate.

IXC (Interexchange Carrier): A long-distance provider that maintains its own switching equipment.

IVR (Interactive Voice Response): Provides mechanism for information to be stored and retrieved

using voice and a touchtone telephone.

Local Loop: The local telephone company provides the transmission facility from the customer to the

telephone company’s office, which is engineered to carry voice and/or data.

North American Numbering Plan (NANP): How we identify telephone numbers in North America. We

can identify the telephone number based on their three separate components (NPA) (NXX) (XXXX).

PIN (Personal Identification Code): A customer calling/billing code for prepaid and pay-as-you-go

calling cards.

Private Branch Exchange: Advanced phone system commonly used by the medium to larger

customer. It allows the customer to perform a variety of in-house routing (inside calling). The dial tone

that is heard when the customer picks up the phone is an internal dial tone.

SS7 (System Signaling Number 7): Technology used by large carriers to increase the reliability and

speed of transmission between switches.

Switch (Switching): Equipment that connects and routes calls and provides other interim functions

such as least cost routing, IVR, and voicemail. It performs the “traffic cop” function of

telecommunications via automated management decisions.

Touchtone (DTMF): The tone recognized by a push button (touchtone) telephone.

Unified Messaging: Platform that lets users send, receive, and manage all email, voice, and fax

messages from any telephone, PC, or information device.

Voice Mail: A system that allows storage and retrieval of voice messages through voicemail boxes.