Sipura Technology SPA2002-ER - Earthlink Truevoice Phone Adpt, SPA-2000, SPA-1000, SPA-3000 User Manual

Sipura Technology, Inc.

SPA User Guide

July 2004

Disclaimer – Please Read:

This document contains implementation examples and techniques using Sipura Technology, Inc. and, in som e instances, other company’s techno logy and products and is a recommendation only and does not constitute any legal arrangement between Sipura Technology, Inc. and the reader, either written or implied. The conclusions reached and recommendations and statements made are based on generic network , service and application requir ements and should be r egarded as a guide to assist you in forming your own opinions and decision regarding your particular situation. As well, Sipura Technology reserves the right to change the features and functionalities for products described in this document at any time. These changes may involve changes to the described solutions over time.

Use of Proprietary Information and Copyright Notice:

This document contains proprietary information that is to be used only by Sipura Technology custom ers. Any unau thorized dis closure, copying, dis tribution, or use of this information is prohibited.

Sipura Technology, Inc. SPA User Guide

Table of Contents

1. Product Description ....................................................................................................................... 6

1.1. SPA Hardware Overview...................................................................................................... 6

2. Installation Overview ..................................................................................................................... 7

3. Software Configuration.................................................................................................................. 8

3.1.1.1. Firmware Upgrade................................................................................................................8

3.2. IVR Interface......................................................................................................................... 8

3.3. Web Interface ..................................................................................................................... 11

3.3.1. Web Interface Conventions.......................................................................................................... 11

3.3.2. Administration Privileges.............................................................................................................. 12

3.3.3. Basic and Advanced Views.......................................................................................................... 12

3.3.3.1. Resync URL........................................................................................................................ 12

3.3.3.2. Reboot URL........................................................................................................................ 13

Through the Reboot URL, you can reboot the SPA............................................................................... 13

Note: Upon request, the SPA will reboot only when it is idle..................................................................13

3.4. Configuration Parameters................................................................................................... 13

3.4.1. System Parameters...................................................................................................................... 13

System Configuration................................................................................................................................. 13

Network Configuration................................................................................................................................ 13

3.4.2. Provisioning Parameters.............................................................................................................. 14

3.4.3. Upgrade Parameters.................................................................................................................... 15

3.4.4. Protocol Parameters.....................................................................................................................15

3.4.4.1. Dynamic Payload Types..................................................................................................... 17

3.4.4.2. SDP Audio Codec Names................................................................................................... 18

3.4.4.3. NAT Support....................................................................................................................... 18

3.4.5. Line 1 and Line 2 Parameters ...................................................................................................... 19

3.4.5.1. User Account Information................................................................................................... 19

3.4.5.2. Supplementary Services Enablement................................................................................. 22

3.4.5.3. Audio Settings..................................................................................................................... 23

3.4.5.4. Dial Plan............................................................................................................................. 25

3.4.5.5. Polarity Settings.................................................................................................................. 25

3.4.6. User 1 and User 2 Parameters..................................................................................................... 25

3.4.6.1. Call Forward And Selective Call Forward/Blocking Settings............................................... 26

3.4.6.2. Speed Dial Settings............................................................................................................ 26

3.4.6.3. Supplementary Service Settings......................................................................................... 26

3.4.6.4. Distinctive Ring and Ring Settings...................................................................................... 27

3.4.7. Regional Parameters....................................................................................................................28

3.4.7.1. Call Progress Tones........................................................................................................... 28

3.4.7.2. Ring and CWT Cadence..................................................................................................... 29

3.4.7.3. Control Timer Values (sec)................................................................................................. 30

3.4.7.4. Vertical Service Code Assignment......................................................................................31

3.4.7.5. Outbound Call Codec Selection Codes: ............................................................................. 34

3.4.7.6. Miscellaneous Parameters.................................................................................................. 34

3.5. Call Statistics Reporting...................................................................................................... 36

4. SPA-3000 Configuration.............................................................................................................. 38

4.1. Overview............................................................................................................................. 38

4.2. SPA-3000 Voice Configur ation Organization ..................................................................... 39

4.2.1. FXS Interface............................................................................................................................... 40

4.2.2. FXO Interface............................................................................................................................... 41

4.2.3. VoIP Interfaces............................................................................................................................. 42

4.2.4. Call Types.................................................................................................................................... 42

4.2.5. Determining the Availability of the PSTN line............................................................................... 43

4.3. Gateway Call Restriction by Dial Plan................................................................................ 43

4.4. Authentication Methods ...................................................................................................... 44

4.5. VoIP-To-PSTN Calls (Call Type #4)................................................................................... 46

4.5.1. One-Stage Dialing ........................................................................................................................ 46

4.5.2. Two-Stage Dialing........................................................................................................................ 47

4.6. PSTN-To-VoIP Calls (C al l Type #3)................................................................................... 48

4.7. Terminating Gateway Calls................................................................................................. 50

4.8. Line 1 VoIP Outbound Call Routing (Call Type #7)............................................................ 51

4.9. Line 1 VoIP Fallback to PSTN............................................................................................ 52

4.10. VoIP-To-PSTN Calls Via VoIP1 Interface (Call Type #5)................................................... 52

4.11. PSTN Call Ring Thru Line 1 (Call Type #6)........................................................................ 53

4.12. Symmetric RTP...................................................................................................................54

4.13. Configuration Examples and Call Scenarios...................................................................... 54

4.13.1. Setup VoIP1 and VoIP2 With Separate VoIP Accounts........................................................... 54

4.13.2. Setup VoIP1 and VoIP2 with Same VoIP Account.................................................................. 55

4.13.3. PSTN-To-VoIP Call Without Ringing Thru Line 1.................................................................... 55

4.13.4. PSTN Call Answered By Line 1............................................................................................... 56

4.13.5. VoIP-to-PSTN Call via VoIP2 Interface With PIN Authentication............................................. 57

4.13.6. VoIP-to-PSTN Call via VoIP2 Interface With HTTP Digest Authentication:............................. 57

4.13.7. Line 1 Forward-On-No-Answer to PSTN Gateway.................................................................. 58

4.13.8. Line 1 Forward-All to PSTN Gateway...................................................................................... 59

4.13.9. Line 1 Forward-On-No-Answer to a Particular PSTN Number................................................. 59

4.13.10. Line 1 Forward-Selective to PSTN Gateway or Number ......................................................... 59

4.13.11. From Line 1 Dials 9 to Access PSTN-Gateway for Local Calls................................................ 59

4.13.12. From Line 1 Route 311 and 911 Calls to PSTN-Gateway....................................................... 60

4.14. Summary of SPA-3000 Configuration Parameters............................................................. 60

4.14.1. PSTN Line – Dial Plans........................................................................................................... 60

4.14.2. PSTN Line – VoIP-To-PSTN Gateway Setup.......................................................................... 60

4.14.3. PSTN Line – VoIP Users and Passwords (HTTP Authentication) ........................................... 61

4.14.4. PSTN Line – PSTN-To-VoIP Gateway Setup.......................................................................... 62

4.14.5. PSTN Line – FXO Timer Values – In seconds......................................................................... 63

4.14.6. PSTN Line – PSTN Disconnect Detection............................................................................... 64

4.14.7. PSTN Line – International Control........................................................................................... 65

4.14.8. Line 1 and PSTN Line – Audio Configuration.......................................................................... 66

4.14.9. Line 1 – Gateway Accounts..................................................................................................... 66

4.14.10. Line 1 – VoIP Fallback To PSTN............................................................................................. 67

4.14.11. Line 1 – Dial Plan .................................................................................................................... 67

4.14.12. User1 – Call Forward Settings................................................................................................. 67

4.14.13. User1 – Selective Call Forward Settings................................................................................. 68

4.14.14. Regional – Call Progress Tones .............................................................................................. 68

4.14.15. PSTN User – PSTN-To-VoIP Selective Call Forward Settings................................................ 68

4.14.16. PSTN User – PSTN-To-VoIP Speed Dial Settings.................................................................. 68

4.14.17. PSTN User – PSTN Ring Thru Line 1 Distinctive Ring Settings.............................................. 68

4.14.18. PSTN User – PSTN Ring Thru Line 1 Ring Settings............................................................... 69

4.14.19. Info – PSTN Line Status.......................................................................................................... 69

4.14.20. PSTN/VoIP Caller Commands via DTMF................................................................................ 70

5. User Guidelines........................................................................................................................... 70

5.1. Basic Services .................................................................................................................... 71

5.1.1. Originating a Phone Call.............................................................................................................. 71

5.1.2. Receiving a Phone Call................................................................................................................ 71

5.2. Enhanced Services............................................................................................................. 71

5.2.1. Caller ID....................................................................................................................................... 72

5.2.2. Calling Line Identification Presentation (CLIP)............................................................................. 72

5.2.3. Calling Line Identification Restriction (CLIR) – Caller ID Blocking................................................ 72

5.2.4. Call Waiting.................................................................................................................................. 73

5.2.5. Disable or Cancel Call Waiting..................................................................................................... 73

5.2.6. Call-Waiting with Caller ID............................................................................................................ 75

5.2.7. Voice Mail..................................................................................................................................... 75

5.2.8. Attendant Call Transfer................................................................................................................ 76

5.2.9. Unattended or “Blind” Call Transfer.............................................................................................. 76

5.2.10. Call Hold.................................................................................................................................. 77

5.2.11. Three-Way Calling................................................................................................................... 77

5.2.12. Three-Way Ad-Hoc Conference Calling.................................................................................. 78

5.2.13. Call Return............................................................................................................................... 78

5.2.14. Automatic Call Back................................................................................................................ 79

5.2.15. Call FWD – Unconditional....................................................................................................... 79

5.2.16. Call FWD – Busy..................................................................................................................... 80

5.2.17. Call FWD - No Answer ............................................................................................................81

5.2.18. Anonymous Call Blocking........................................................................................................ 82

5.2.19. Distinctive / Priority Ringing and Call Waiting Tone................................................................. 82

5.2.20. Speed Calling – Up to Eight (8) Numbers or IP Addresses..................................................... 83

6. Appendix I: Dial Plan .................................................................................................................. 83

1. Product Description

This guide describes basic use of the Sipura Technology SPA phone adapter – an intelligent lowdensity Voice over IP (VoIP) gateway. The SPA e nables carrier class residentia l and business IP Telephony services de livered over broadband or high-speed Internet connect ions. By intelligent, we mean the SPA maintains the states of all the calls it terminates. It is capable of making proper decisions in reaction to user input events (such as on/off hook or hook flash) with little or no involvement by a ‘middle-man’ server or media gateway controller.

Examples of proper reactions are: playing dial tone, collecting DTMF digits, comparing them against a dial plan and term inating a call. With inte lligent endpoints at the e dges of a network , performing the bulk of the call pr ocessing duties, the dep loyment of a large networ k with thousands of subsc ribers can scale quickl y without the introduction of complic ated, expensive servers. As desc ribed later in this section, the S ession Initiation Protocol ( SIP) is a good choice of call s ignaling protocol for the implementation of such a device in this type of network.

1.1. SPA Hardware Overview

The SPA has one of the smalles t f orm f actor s on the mar k et. It can be ins tall ed in minutes as a tabletop or wall mount CPE device. The images belo w show the SPA-2000. The SPA- 1000 and SPA3000 are similar to size and shape – the only difference being the color of the adapter.

Figures Figure 1, Figur e 2, Figure 3 and Fi gure 4 show the front, re ar, left side and right s ide of the SPA-2000, respectively.

Figure 1 – SPA-2000 Front

Figure 3 – SPA-2000 Rear

Figure 2 – SPA-2000 Left Side

Figure 4 – SPA-2000 Right Side

The SPA has the following interfaces for networking, power and visual status indication:

1. Two (2) RJ-11 Type Analog Telephone Jack Interfaces (Figure 4, above): These interfaces ac cept standard RJ-11 telephone c onnectors. An Analog touc htone telephone or

fax machine m ay be conne cted to either int erf ace. If the s ervic e sup ports o nl y one incoming line, the analog telephone or f ax machine sho uld be connec ted to port one ( 1) of the SPA. Port one (1) is the outermost telephone port on the SPA and is labeled “Phone 1.”

The SPA-3000 has an RJ-11 interfac e labeled “Line” which can be used to connect the adapter to a PSTN analog telephone circuit.

2. One LED for Unit Status (Figure 4, above):

3. One Ethernet 10baseT RJ-45 Jack Interface ( Figure 2, above): This interface acc epts a standard or crossover Ethernet cable with s tandard RJ-45 connector. For

optimum perf ormanc e, Si p ura Technology recommends that a Category 5 cable or greater b e used in conjunction with the SPA.

4. One LED for Data Link and Activity ( Figure 2, above):

5. One 5 Volt Power Adapter Interface ( Figure 2, above) This interface acc epts the SPA power adapter that c ame with the unit. Sipura T echnology does not

support the use of any other power adapters other then the power a dapter that was s hipped with the SPA unit.

2. Installation Overview

Please check to make sure that you have the following package contents:

1. Sipura Phone Adapter Unit

2. Ethernet Cable

3. RJ-11 Phone Cable (SPA-3000 Only)

4. SPA Quickstart Guide5.

5. Volt Power Adapter You will also need:

1. One or Two Analog Touch Tone Telephones (or Fax Machine)

2. Access to an IP Network via an Ethernet Connection

3. Access to a PSTN network connection – SPA-3000 only. Please observe the following steps to install the SPA.From the Left Side of the SPA:1. Insert a

standard RJ-45 Ether ne t c a ble (inc lud ed) into th e LAN port.2. Insert t he power adapter cabl e into th e 5V power adapter cable r eceptacle. Ensure that th e power adapter jack is snugly attached to the SPA.From the Right Sid e of the SPA:1. Insert a standard RJ-11 te lephone cable into the Phone 1 port.2. Connect the other end of the cable to an analog telephone or fax machine.3. Insert a standard RJ-11 telephone cable into the Phone 2 port (Optional).4. Connect the other end of the cable to an analog telephone or fax machine.

Note: Do not conn ect RJ-11 telephone cable from the SPA-1000 or SPA-2000 to th e wall jack to prevent any chance of connec tion to the circuit switched telco ne twork.You may now insert the plu g end of the power adapter into a live power outlet which will power up the SPA.

3. Software Configuration

3.1.1.1. Firmware Upgrade Firmware Upgrade via PC Utility Program:

From time to time, Sipura Technology will make available a PC executable file that will facilitate the upgrade of a SPA. In order to upgrade a device via this method, the end user must have administrative permission (via password protected log-in) to perform this upgrade.

Once the user has obtained the proper firmware upgrade executable, the user simply runs the program from a file location on their local PC. The PC program walks the user through the upgrade process via a graphical user interface. Generally, the entire upgrade process should take no more than five minutes to complete.

Please note: Some end-users who have obtained their SPA directly from a service provider will never need to manually upgrade their device. Via the remote upgrade process, Sipura Technology provides capability for the SPA to be maintained from a remote location (e.g. a service provider network server), using the Internet connection of the end-user as the conduit through which profile updates and firmware upgrades are performed.

3.2. IVR Interface

Administrators and/or users can chec k (read) and set ( write) basic net work configuration s ettings via a touchtone telephone connected to one of the RJ-11 phone ports of the SPA.

Please Note: Service Providers of fering service using the SPA may restrict, protect or turn of f certain aspects of the

unit’s IVR and web configuration capabilities. The Interactive Voice Res ponse (IVR) capabi lities of the SPA are des igned to give the adminis trator

and/or user basic rea d/write capabilities such that the unit c an attain basic IP network connectivit y and the more advanced browser-based configuration menu may be accessed.

1. The SPA IVR uses the following conventions: By factory default there is no password and no password authentication is prompted for all the IVR settings. If administrator password is set, password authentic ation will be prompted f or certain I VR settings . See 3.4.2 f or detailed inform ation about administrator password.

To input the password using the phone keypad, the following translation convention applies:

o To input: A, B, C, a, b, c -- press ‘2’ o To input: D, E, F, d, e, f -- press ‘3’ o To input: G, H, I, g, h, i -- press ‘4’ o To input: J, K, L, j, k, l -- press ‘5’ o To input: M, N, O, m, n, o -- press ‘6’ o To input: P, Q, R, S, p, q, r, s -- press ‘7’ o To input: T, U, V, t, u, v -- press ‘8’ o To input: W, X, Y, Z, w, x, y, z -- press ‘9’ o To input all other characters in the administrator password, press ‘0’

Note: This translation convention only applies to the password input. For example: to input password “test#@1234” by phone keypad, you need to press the following

sequence of digits: 8378001234.

2. After entering a value, press the # (pound) key to indicate end of input.

o To Save value, press ‘1’ o To Review the value, press ‘2’ o To Re-enter the value, press ‘3’

o To Cancel the value entry and return to the main configuration menu, press ‘

Notes:

o The final ‘#’ key won’t be counted into value. o Saved settings will take effect when the telephone is hung-up and if necessary, the S PA will

automatically reboot.

3. After one minute of inactivity, the un it times out. T he user will need to re-en ter the configurat ion menu from the beginning by pressing * * * *.

4. If, while entering a valu e (lik e an IP addr ess) and you d ecid e to exit with out enter ing an y changes , you may do so by pressing the * (star) key twice within a half second window of tim e. Otherwise, the entry of the * (star) key will be treated as a dot (decimal point).

Example: To enter IP addres s, use numbers 0 – 9 on the telephone ke y pad and use the * (star) key to enter a decimal point.

To enter the following IP address value: 192.168.2.215 A. Use the touchtone key pad to enter: 192*168*2*215# B. When prompted, enter 1 to save setting to configuration. C. Hang-up the phone to cause setting to take effect.

- or D. Enter the value of the next setting category to modify . . .

*’ (star)

5. Hang-up the phone to cause all settings to take effect.

SPA Interactive Voice Response (IVR) Menu:

IVR Action IVR Menu Choice Parameter(s) Notes:

Enter IVR Menu

* * * *

Exit IVR Menu Check DHCP

3948 100

None Ignore SIT or other tones

until you hear, “Sipura configuration menu. Please enter option followed by the pound key

or hang-up to exit.” None None IVR will announce if DHCP

in enabled or disabled.

Enable/Disable DHCP

Check IP Address

Set Static IP Address

Check Network Mask

Set Network Mask

Check Static Gateway IP Address

101

110

111

120

121

130

Enter 1 to enable Enter 0 to disable

Requires Password

None IVR will announce the

current IP address of SPA. Enter IP address

using numbers on the telephone key pad. Use the * (star) key when entering a decimal

DHCP must be “Disabled”

otherwise you will hear,

“Invalid Option,” if you try

to set this value.

Requires Password point.

None IVR will announce the

current network mask of

SPA. Enter value using

numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

DHCP must be “Disabled”

otherwise you will hear,

“Invalid Option,” if you try

to set this value.

Requires Password

None IVR will announce the

current gateway IP

address of SPA.

Set Static Gateway IP Address

Check MAC Address

Check Firmware Version

Check Primary DNS Server Setting

Set Primary DNS Server

131

140

150

160

161

Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal

DHCP must be “Disabled”

otherwise you will hear,

“Invalid Option,” if you try

to set this value.

Requires Password point.

None IVR will announce the

MAC address of SPA in

hex string format. None IVR will announce the

version of the firmware

running on the SPA. None

IVR will announce the

current setting in the

Primary DNS field.

Enter IP address

Requires Password

using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Check SPA’s Web Server Port

170

None

IVR will announce the

port that the web server

is listening on. (Default is

80)

Enable/Di sable Web Server of SPA

Manual Reboot of Unit

7932

732668

Enter 1 to enable Enter 0 to disable

None After you hear “Option

Requires Password

Successful,” hang-up. Unit

will reboot automatically.

User Factory Reset of Unit

WARNING: ALL “User-Changeable” NON-

DEFAULT SETTINGS WILL BE LOST!

This might include network and service provider data.

877778

Enter 1 to confirm Enter *(star) to cancel operation

SPA will prompt for

confirmation. After

confirming, you will hear

“Option Successful.” Hang-

up. Unit will reboot and all

“User Changeable”

configuration param eter s

will be reset to factory

default values.

Factory Reset of Unit

WARNING: ALL NON-DEFAULT SETTINGS

WILL BE LOST! This includes network and

service provider data.

73738

Enter 1 to confirm Enter * (star) to cancel operation

SPA will prompt for

confirmation. After

confirming, you will hear

“Option Successful.” Hang-

up. Unit will reboot and all

configuration param eter s

will be reset to factory

default values.

Note: If the Administrator password is not set or the user is allowed to change it, the items marked with “Requires Password” will not require a password.

3.3. Web Interface

The SPA provides a built-in web server. C onfigurati on and adm inistration c an be perfor med through this convenient web interf ac e.

3.3.1. Web Interface Conventions

The SPA uses the following conventions with the web administration capabilities:

o The SPA web adm inistration supports two privil ege levels: Administrator and U ser. To use

the User privilege, simply point a web browser at the IP address of the SPA; to use the administrator privilege, use URL http://IP_Address_Of_SPA/admin information about administration privileges.

o Version 1.0 of the SPA supports Internet Explorer 5.5 and above and Netscape 7.0 and

above.

o The web configuration pages can be password protected. See 3.3.2 for more information

about password protect.

o The user name of web Administrator is : admin o The user name of web User is : user

/. See 3.3.2 for more

o Note: The user names for both administrator and User are fixed and cannot be changed. o After making changes to SPA configuration parameters, pressing “Submit All Changes”

button will apply all th e changes and if necessar y, automatically reboot the devi ce. Multiple changes may be made on multiple page tabs of the web int erf ac e at the s ame time. Pressing “Submit All Changes” will apply all the modifications.

Important Note: switching between page tabs won’t apply the changes to SPA, The only way to apply the changes is to press the “Submit All Changes” button.

o If the “Undo All Changes” button is clicked, an y modifications to prof ile parameters on any and

all pages will be reset back to their original values before modification.

NOTE: Pressing the “Undo All Changes” has no effect on the SPA; it will only reset the values on the web page.

3.3.2. Administration Privileges

The SPA supports t wo levels of adm inistration pr ivileg es: Adm inistrator a nd User, both pr ivileges can be password protecte d. Im portant note: b y factory default, there ar e n o pass words assigned for b oth Administrator and User.

The Administrator h as the privil ege to modif y all the web prof ile parameters and can also m odify the passwords of both Administrator and User. A User only has the pr ivilege to access part of the web profile parameter s ; the par ameter group that User c an ac ces s is s pec if ie d by the Administrator , which can only be done through provisioning.

To access the Administrator level privilege, use URL: http://IP_Address_Of_SPA/admin

/. If the password has been set f or Administrator, the browser will prom pt for authentication. The username for Administrator is “admin” and cannot be changed.

To access the User lev el privilege, use URL: http://IP_Address_Of_SPA/

. If the password has be en set for User, the br owser will prompt for User auth entication. The username f or User is “user” and cannot be changed.

When browsing Adm inistrator pag es, one can s witch to User pri vileges b y click the link “User Login”. (Note: if User password was set, the br owser wil l prompt for User authenticatio n when you clic k “User Login” link). On the other side, from the User pages you can switch to Administrator privilege by clicking the link “Admin Login.” Authentication is needed if Administrator password has been set.

Warning: S witching between the User and Administrator will discard the uncom mitted changes that have already been made on the web pages.

3.3.3. Basic and Advanced Views

The web configuration interface provides a Basic and an Advanced view from which the various configuration parameters can be accessed. The SPA Provisioning tab is only visible from the Advanced Administr ator vie w of the web interf ace.

Warning: Switching between the bas ic and a dv anc ed vie w will d isc ar d the u ncommitted changes that have already been made on the web pages.

3.3.3.1. Resync URL

Through Resync URL you can force the SPA to do a resync to a profile specified in the URL. Note: The SPA will resync only when it is idle. The syntax of Resync URL is:

http://<spa-ip-addr>/resync?[[protocol://][server-name[:port]]/profile-pathname]

If no parameter follows “/resync?”, the profile rule setting in provisioning is used. See Error! Reference source not found. for detailed information about profile rule in provisioning

If no protocol is specif ied, TFTP protocol is assumed. Note: O nly TFTP is supported in the current release.

If no server-name is specified, the host that requests the URL is used as server-name. If no port specified, default port of the protocol is used – 69 for TFTP. The profile-path is the path to the new profile to resync with. For example: http://192.168.2.217/upgrade?tftp://192.168.2.251/spaconf.scf

3.3.3.2. Reboot URL

Through the Reboot URL, you can reboot the SPA. Note: Upon request, the SPA will reboot only when it is idle. The Reboot URL is: http://<spa-ip-addr>/admin/reboot

3.4. Configuration Parameters

3.4.1. System Parameters

System Configuration

Parameter Name Description Default

Restricted Access Domains Enable Web Server Enable/disable web server of SPA

Enable Web Admin Access Admin Password The password for administrator User Password The password for User

Parameter Name Description Default

DHCP Enable/Disable DHCP Yes Host Name Host Name of SPA Domain The network domain of SPA Static IP Static IP address of SPA, which will take effect if DHCP

NetMask The NetMask used by SPA when DHCP is disabled 255.255.255. Gateway The default gateway used by SPA when DHCP is Primary DNS DNS server used by SPA in addition to DHCP supplied

Secondary DNS DNS server used by SPA in addition to DHCP supplied

This feature is used when implementing software customization.

This feature should only be used on firmware version 1.0.9 or later.

Enable/disable Admin pages of web server of SPA

Network Configuration

is disabled

disabled DNS servers if DHCP is enabled; when DHCP is

disabled, this will be the primary DNS server. DNS servers if DHCP is enabled; when DHCP is

disabled, this will be the secondary DNS server.

Yes Yes

0.0.0.0

DNS Query Mode Do parallel or sequential DNS Query Parallel Syslog Server Specify the Syslog server name and port. This feature

specifies the server for logging SPA system information and critical events.

Debug Server The debug server name and port. This feature

specifies the server for logging SPA debug information. The level of detailed output depends on the debug level parameter setting.

Debug Level The higher the debug level, the more debug

0 information will be generated. Zero (0) means no debug information will be generat ed.

Primary NTP

IP address or name of primary NTP server.

Server Secondary NTP

IP address or name of secondary NTP server

Server Web Server Port TCP port through which the SPA web server will

80 communicate

Notes:

- Parallel DNS query mode: SPA will send the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply will be accepted by SPA.

- To log SIP messages, Debug Level must be set to at least 2.

- If both Debug Server and Syslog Server are specified, _Syslog messages are also logged to the

Debug Server.

3.4.2. Provisioning Parameters

Provisioning operations are gated by the Provision_Enable parameter.

Parameter Name Description Default

Provision Enable yes Resync On Reset yes Resync Random Delay Resync Periodic 3600 Resync Error Retry Delay Resync From SIP Yes Resync After Upgrade Attempt Resync Trigger 1 Resync Trigger 2 Profile Rule /spa.cfg Profile Rule B Profile Rule C Profile Rule D Log Resync Request Msg

Log Resync Success Msg

3600

Yes

See

provisioning discussion section

See

provisioning discussion

section Log Resync Failure Msg

See

provisioning

discussion

section GPP A thru GPP P empty GPP SA thru GPP SD empty

Note: In a customized SPA, the profile ru le wou ld po int to a service pr ov id er’s serv er .

3.4.3. Upgrade Parameters

Parameter Name Description Default

Upgrade Enable Yes Upgrade Error Retry Delay Upgrade Rule empty Log Upgrade Request Msg

Log Upgrade Success Msg

Log Upgrade Failure Msg

Note: In a customized SPA, the upgrade rule would point to a service provider’s server.

3600

See

provisioning discussion section

See

provisioning discussion section

See

provisioning discussion section

3.4.4.

Protocol Parameters

Parameter Name Description Default

Max Forward SIP Max-Forward value. Range: 1 – 255 70 Max Redirection Number of times to allow an INVITE to be

5 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.

Max Auth Maximum number of times a request may be

2 challenged (0-255)

SIP User Agent Name

User-Agent Header to be used by the unit in outbound requests. If empty, the header is not

Sipura/

$version included.

SIP Server Name Server Header to used by the unit in

responses to inbound responses. If empty,

Sipura/

$version the header is not included.

SIP Accept Language

Accept-Language Header to be used b y the unit. If empty, the header is not included.

Remove Last Reg Remove last registration before registering a

no new one if value is different one.

DTMF Relay MIME Type Hook Flash MIME Type

This is the MIME Type to be used in a SIP INFO message used to signal DTMF event. This is the MIME Type to be used in a SIP INFO message used to signal hook flash

application/dtmf-relay

application/hook-flash event.

Use Compact Header

If set to yes, the SPA will use compact SIP headers in outbound SIP messages. If set to

no no the SPA will use normal SIP headers.

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

.5 – 64 sec

SIP T2 RFC 3261 T2 value (Maximum retransmit

4 interval for non-INVITE requests and INVITE responses). Range: 0 – 64 sec

SIP T4 RFC 3261 T4 value (Maximum duration a

5 message will remain in the network). Range: 0 – 64 sec

SIP Timer B INVITE time out value. Range: 0 – 64 sec 32 SIP Timer F Non-INVITE time out value. Range: 0 – 64

32 sec

SIP Timer H INVITE final response time out value. Range:

32 0 – 64 sec

SIP Timer D ACK hang around time. Range: 0 – 64 sec 32 SIP Timer J Non-INVITE response hang around time.

32 Range: 0 – 64 sec

INVITE Expires INVITE request Expires header value in sec.

0 = do not include Expires header in INVITE. Range: 0 – (2

– 1)

ReINVITE Expires ReINVITE request Expires header value in

sec. 0 = do not include Expires header in the request. Range: 0 – (2

– 1)

Reg Min Expires Minimum registration expiration time allowed

180

1 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.

Reg Max Expires Maximum registration expiration time allowed

7200 from the proxy in the Min-Expires header. If value is larger than this, then the maximum value is used

Reg Retry Intvl Interval to wait before the SPA retries

30 registration again after encountering a failure condition during last registration

Reg Retry Long Interval

When Registration fails with a SIP response code that does no match <Retry Reg RSC>,

1200 the SPA will wait for the delay specified in this

parameter before retrying. If this parameter is 0, the SPA will stop retrying. This value should be much larger than <Reg Retry Intvl> which should not be 0.

SIT1 RSC1 SIP response status code to INVITE on which

to play the SIT1 Tone

SIT2 RSC1 SIP response status code to INVITE on which

to play the SIT2 Tone

SIT3 RSC1 SIP response status code to INVITE on which

to play the SIT3 Tone

SIT4 RSC1 SIP response status code to INVITE on which

to play the SIT4 Tone

Try Backup RSC SIP response status code on which to retry a

backup server for the current request

Retry Reg RSC Interval to wait before the SPA retries

30 registration again after encountering a failure condition during last registration

RTP Port Min2 Minimum port number for RTP transmission

16384 and reception

RTP Port Max2 Maximum port number for RTP transmission

16482 and reception

RTP Packet Size Packet size in sec. Valid values must be

0.02

multiple of 0.01s. Range: 0.01 – 0.16

RTCP Tx Interval4 Controls the interval (sec) to send out RTCP

0 sender report on an active connection. Range: 0 – 255 (s)

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, SPA will reuse the same compact headers when generating th e respons e regardless the set tings of the <Use Compact Hea der> param eter. If inbound SIP requests contain norm al headers, SPA will substitute th ose headers with compact headers (if defined by RFC 261) if <Use Compact Header> parameter is set to “yes.”

4. During an active connec tion, the SPA can be pr ogrammed to send out compound RTCP p acket 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 ex cept the last one conta ins exac t l y 1 RR (R ec ei ver Report); the last S R car r ies no RR. The SDES cont ains CNAME, NAME, and TOOL ident ifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> ( or “Anonymous” if user block s caller ID), and TOOL is set to the Verdor/Hardware-platform-software-version (such as Sipura/SPA2000-

1.0.31(b)). The NT P timestam p used in the SR is a snaps hot of the S PA’s loc al ti m e, not the tim e reported by an NTP server. If the SPA rec eives a RR f rom the peer, it will a ttem pt to com pute the round trip delay and show it as t he <Cal l Ro un d T r ip Dela y> value (ms) in the Info section of SPA web page.

3.4.4.1. Dynamic Payload Types

Parameter Name Description Default

NSE Dynamic Payload AVT Dynamic Payload G726r16 Dynamic Payload G726r24 Dynamic Payload G726r40 Dynamic Payload G729b Dynamic Payload

1,2

NSE dynamic payload type 100

1,2

AVT dynamic payload type 101

1,2

G726-16 dynamic payload type 98

1,2

G726-24 dynamic payload type 97

1,2

G726-40 dynamic payload type 96

1,2

G729b dynamic payload type 99

Notes:

1. Valid range is 96 – 127

2. The configured d ynamic payloads are used for outbou nd calls only where the SPA prese nts the

SDP offer. For inbound calls with a SDP offer, SPA will follow the caller’s dynamic payload type assignments

3.4.4.2. SDP Audio Codec Names

Parameter Name Description Default

NSE Codec Name NSE Codec name used in SDP NSE AVT Codec Name AVT Codec name used in SDP telephone-event G711a Codec Name G711a Codec name used in SDP PCMA G711u Codec Name G711u Codec name used in SDP PCMU G726r16 Codec Name G726-16 Codec name used in SDP G726-16 G726r24 Codec Name G726-24 Codec name used in SDP G726-24 G726r32 Codec Name G726-32 Codec name used in SDP G726-32 G726r40 Codec Name G726-40 Codec name used in SDP G726-40 G729a Codec Name G729a Codec name used in SDP G729a G729b Codec Name G729b Codec name used in SDP G729ab G723 Codec Name G723 Codec name used in SDP G723

Notes:

1. SPA uses the configured codec names in its outbound SDP

2. SPA ignores the codec names in incoming SDP for standard payload types (0 – 95).

3. For dynamic payloa d t ypes, S PA ide ntif ies t he c od ec by the configured codec names. Comparison

is case-insensitive.

3.4.4.3. NAT Support

Parameter Name Description Default

Handle_VIA_received If set to “yes”, the SPA will process the “received”

No parameter in the VIA header inserted by the server in a response to any one of its request. Else the parameter is ignored.

Handle_VIA_rport If set to “yes”, the SPA will process the “rport”

No parameter in the VIA header inserted by the server in a response to any one of its request. Else the parameter is ignored.

Insert VIA received Insert received parameter in VIA header in SIP

No responses if received from IP and VIA sent-by IP differ

Insert VIA rport Insert rport parameter in VIA header in SIP

No responses if received-from port and VIA sent-by port differ

Substitute VIA addr Use nat-mapped IP:port values in VIA header No Send Resp To Src Port Send response to the request source port instead of

No the VIA sent-by port

STUN Server STUN server to contact for NAT mapping discovery STUN Enable Enable the use of STUN to discover NAT mapping No STUN Test Enable If enabled with <STUN Enable> = “yes” and a valid

No <STUN Server>, the SPA will perform a NAT type discovery operation when first power on by contacting the configured STUN server. The result of the discovery will be reported in a Warning

header in all subsequent REGISTER requests – “Warning: 399 spa <stun type>”, where <stun type> is one of the following: "Unknown NAT Type", "STUN Server Not Reachable", "STUN Server Not Responding", "Open Internet Detected", "Symmetric Firewall Detected", "Full Cone NAT Detected", "Restricted Cone NAT Detected", "Symmetric NAT Detected"; If the SPA detects Symmetric Nat or Symmetric Firewall, Nat Mapping will be disabled (th at is , no substitution of IP address and port with external IP address an nat-mapped port)

Ext IP External IP address to substitute for the actual IP

address of the unit in all outgoing SIP messages. If “0.0.0.0” is specified, no IP address substitution is performed.

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 s ubstituted by the corresponding port value in the externa l RTP port range.

0.0.0.0

NAT Keep Alive Intvl Interval between sending NAT-mapping keep alive

15 message in sec

Notes:

3.4.5. Line 1 and Line 2 Parameters

Per line parameter tags must be appended with [1] or [2] (corresponding to lines 1 or 2) in the configuration profile. It is omitted below for readability.

3.4.5.1. User Account Information Parameter Name Description Default

Line Enable Enable this line for service Yes MOH Server2 The User ID or URL of the auto-answering SAS to

contact for MOH services. Examples: 5000, 1001@music.sipura.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). SIP Port SIP message listening port and transmission port 5060 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. SIP TOS/DiffServ Value RTP TOS/DiffServ Value

TOS/DiffServ field value in UDP IP Pac kets

carrying a SIP Message

TOS/DiffServ field value in UDP IP Pac kets

carrying a RTP data

Empty

0x68 0xb8

SAS Enable3 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. SAS DLG Refresh

Intvl

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, SPA

will terminate this call with a SIP BYE message.

The default = 0 (Session refresh disabled)

Range = 0-255 (s) SAS Inbound RTP

Sink

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 SPA 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 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]. NAT Mapping Enable Enable the use of externally mapped of IP address

and SIP/RTP ports in SIP messages . The mappin g

may be discovered by any of the supported

methods. NAT Keep Alive Enable

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

Msg> is sent periodicall y ever y <NAT Keep Al i ve

Intvl> seconds. 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. 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 SIP Debug Option None, 1-line, full, exclude OPTIONS, exclude

low. This parameter affects how jitter buffer size is

adjusted in the SPA. 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

$NOTIFY

$PROXY

none High

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). SIP 100REL Enable Enable the support or the 100rel SIP extension for

No reliable transmission of provisional responses (18x) and the use of PRACK requests.

Blind Attn-Xfer Enable

If enabled, the SPA performs an attended transfer operation by terminating the current call leg, and

No blind transferring the other call leg. If disabled, the

SPA performs an attended transfer by referring the other call leg to the current call leg while maintaining both call legs.

Proxy SIP Proxy Server for all outbound requests Use Outbound Proxy Enable the use of <Outbound Proxy>. If set to “no”,

No <Outbound Proxy> and <Use OB Proxy in Dialog) is ignored.

Outbound Proxy SIP Outbound Proxy Server where all outbound

No requests are sent as the first hop.

Use OB Proxy In Dialog

Whether to forcer SIP requests to be sent to the outbound proxy within a dialog. Ignored if <Use

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

empty

Yes parameter is ignored if <Proxy> is not specified.

Make Call Without Reg

Allow making outbound calls without successful (dynamic) registration by the unit. If “No”, dial tone

No will not play unless registration is successful

Ans Call Without Reg Allow answering inbound calls without successful

No (dynamic) registrat ion b y the unit

3600 will periodically renew registration shortly before the current registration expired. This parameter is ignored if <Register> is “no”. Range: 0 – (2

– 1)

sec

Use DNS SRV Whether to use DNS SRV lookup for Proxy and

No Outbound Proxy

DNS SRV Auto Prefix If enabled, the SPA will autom atic ally prepend the

No Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name

Proxy Fallback Intvl This parameter sets the delay (sec) after which the

3600 SPA will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the SPA via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the SPA will not attempt to fall back after a fail over)

Display Name Subscriber’s display name to appear in caller-id

User ID Subscriber’s user-id. Usually a E.164 number Password Subscriber’s a/c password Auth ID Subscriber’s authentication ID Use Auth ID If set to “yes”, the pair <Auth ID> and <Password>

No are used for SIP authentication. Else the pair <User ID> and <Password> are used.

Mini Certificate Base64 encoded of Mini-Certificate concatenated

Empty with the 1024-bit public key of the CA signing the MC of all subscribers in the group.

SRTP Private Key Base64 encoded of the 512-bit private key per

Empty subscriber for establishment of a secure call.

Notes:

1. If proxy responded to REGISTER with a smaller Expires valu e, the SPA will renew registration based on this sm aller value inste ad of the configured value. If regist ration failed with an “Expires too brief” error respons e, the SPA will retry with the value given in the M in-Expires header in the error response.

2. MOH Notes:

• The remote party mus t indic ate t hat it c an r ece iv e au dio whi le h ol din g MO H t o work. That is the SIP 2xx response from the remote party in reply to the re-INVITE fr om the SPA to put the call on hold must have the SDP i ndicate a sendrec v or recvo nly attribut e and the r emote des tinatio n 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 SPA 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 audi o source m ust be off-hook for the stream ing to occur. Othe rwise incom ing 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 sess ion, batter y is removed from tip-and-ring of the FXS port. Some audio so urce devices have an L ED to indicate the b attery status. T his can be used as a visual indication whether any audio streaming is in progress.

• IVR can still be used on an SAS line, but th e user needs to f ollo w som e simple s teps: a) Connect a phone to the port and make sur e the phone is on-hook , b) power on the SPA and c ) pick up handset and press * * * * to invoke IVR in the usual way. The idea behin d this is that if th e SPA boots up and finds that the SAS line is on-hook, it will not r emove battery from the line so that IVR ma y be used. But if the SPA boots up and f inds that the SAS line is off-hook, it will remove batter y from the line since no audio session is in progress.

• Set up the Proxy and Su bscr iber Inf orm ation for the SAS Li ne as you norm all y would with a re gular user account.

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

3.4.5.2. Supplementary Services Enablement

The SPA provides nati ve support of a large s et of enhanced or suppl ementary services. Al l of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplem entary service. A supplementar y service should be disabled if a) the user has not subscribed for it, or b) the Ser vice Pro vider inten ds to support s imilar s ervice using oth er means than relying on the SPA.

Parameter Name Description Default

Call Waiting Serv Enable Call Waiting Service Yes Block CID Serv Enable Block Caller ID Service Yes Block ANC Serv Enable Block Anonymous Calls Service Yes Dist Ring Serv Enable Distinctive Ringing Service Yes Cfwd All Serv Enable Call Forward All Service Yes Cfwd Busy Serv Enable Call Forward Busy Servic e Yes

Cfwd No Ans Serv Enable Call Forward No Answer Service Yes Cfwd Sel Serv Enable Call Forward Selective Service Yes Cfwd Last Serv Enable Forward Last Call Service Yes Block Last Serv Enable Block Last Call Service Yes Accept Last Serv Enable Accept Last Call Service Yes DND Serv Enable Do Not Disturb Service Yes CID_Serv Enable Caller ID Service Yes CWCID Serv Enable Call Waiting Caller ID Service Yes Call Return Serv Enable Call Return Service Yes Call Back Serv Enable Call Back Service Yes Three Way Call Serv1 Enable Three W ay Calling Servic e Yes Three Way Conf

1,2

Serv Attn Transfer Serv Unattn Transfer Serv Enable Unattended (Blind) Call Transfer

Enable Three Way Conference Service Yes

1,2

Enable Attended Call Transfer Service Yes

Yes

Service

MWI Serv3 Enable MWI Service Yes VMWI Serv Enable VMWI Service (FSK) Yes Speed Dial Serv Enable Speed Dial Service Yes Secure Call Serv Enable Secure Call Service Yes Referral Serv Enable Referral Service. See <Referral

Yes

Services Codes> for more details

Feature Dial Serv Enable Feature Dial Service. See <Feature

Dial Services Codes> for more details

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.

3.4.5.3. Audio Settings

Parameter Name Description Default

Preferred Codec Select a preferred codec for all calls. However, the

G711u 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

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

No fail if the far end does not support this codec.

LBR Codec Enable *** This parameter has been removed. *** Silence Supp Enable Enable silence suppression so that silent audio

No frames are not transmitted

Echo Canc Enable Enable the use of echo canceller Yes Echo Canc Adapt

Enable echo canceller to adapt Yes

Enable Echo Supp Enable Enable the use of echo suppressor. If <Echo Canc

Yes Enable> is “no”, this parameter is ignored

G729a Enable1 Enable the use of G729a codec at 8 kbps. Yes G723 Enable1 Enable the use of G723 codec at 6.3 kbps Yes G726-16 Enable1 Enable the use of G726 codec at 16 kbps Yes G726-24 Enable1 Enable the use of G726 codec at 24 kbps Yes G726-32 Enable1 Enable the use of G726 codec at 32 kbps Yes G726-40 Enable1 Enable the use of G726 codec at 40 kbps Yes FAX Passthru Enable *** This parameter has been removed. *** Yes FAX CED Detect Enable Enable detection of FAX tone. Yes FAX CNG Detect

Yes

Enable FAX Passthru Codec Codec to use for fax passthru G711u FAX Codec Symmetric Force unit to use symmetric codec during FAX

Yes passthru

FAX Passthru Method Choices: None / NSE / ReINVITE NSE FAX Process NSE Yes DTMF Tx Method Method to transmit DTMF signals to the far end:

Auto 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

Hook Flash Tx Method Select the method to signal Hook Flash events:

None

• 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

Release Unused Codec Yes

Notes:

1. A codec resource is co nsidered as allocated if it has been included in the SDP codec lis t of an active call, even though it even tuall y ma y not be the one chos en for the connec tion. So, if the G.72 9a 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 ac tually uses G.729a. If the G729a res ource is already allocated and si nce only one G.729a resource is allowed per SPA, 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 availab le per SPA. Therefore it is im portant to disable the use of G.729a in ord er to guarantee the support of 2 simultaneous G.723/G.726 codec.

3.4.5.4. Dial Plan

See section 6 for additional information regarding the configuration of the SPA dial plan.

Parameter Name Description Default

Dial Plan Per-line dial plan script See below

Enable IP Dialing Enable IP Dialing no See the previous section for explanation of Dial Plan Script syntax. Default Dial Plan script for each line: “(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.)” Explanation of Default Dial Plan:

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, ...) Note: If IP dialing is enabled, one c an dial [user -id@]a.b.c .d[:port], wh ere ‘@’, ‘.’, and ‘:’ are di aled b y

entering “*”, user-id m ust be numeric (like a phone n umber) and a, b, c, d must be bet ween 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 d ial plan, then it is interpreted as a r egular phone number according to the dial p lan. T he INVIT E m essage, howe ver, is still s ent to th e outbou nd proxy if it is enabled.

3.4.5.5. Polarity Settings

Parameter Name Description Default

Idle Polarity Polarity before call connected Forward

Caller Conn Polarity Polarity after outbound call connected Reverse

Callee Conn Polarity Polarity after inbound call connected Reverse Notes:

3.4.6. User 1 and User 2 Parameters

User 1/2 refers to the subscriber of Line 1/2. When a call is made from Line 1/2, SPA shall use the user and line settings for that Line; there is no user login support in SPA v1.0. Per user parameter tags must be appended with [1] or [2] (corresponding to line 1 or 2) in the configuration profile. It is omitted below for readability.

3.4.6.1. Call Forward And Selective Call Forward/Blocking Settings

Parameter Name Description Default

Cfwd All Dest Forward number for Call Forward All Service

Cfwd Busy Dest Forward number for Call Forward Busy Service

Cfwd No Ans Dest Forward number for Call Forward No Answer Service

Cfwd No Ans Delay Delay in sec before Call Forward No Answer triggers 20

Cfwd Sel1 Caller Caller number pattern to trigger Call Forward Selective 1

Cfwd Sel2 Caller Caller number pattern to trigger Call Forward Selective 2

Cfwd Sel3 Caller Caller number pattern to trigger Call Forward Selective 3

Cfwd Sel4 Caller Caller number pattern to trigger Call Forward Selective 4

Cfwd Sel5 Caller Caller number pattern to trigger Call Forward Selective 5

Cfwd Sel6 Caller Caller number pattern to trigger Call Forward Selective 6

Cfwd Sel7 Caller Caller number pattern to trigger Call Forward Selective 7

Cfwd Sel8 Caller Caller number pattern to trigger Call Forward Selective 8

Cfwd Sel1 Dest Forward number for Call Forward Selective 1

Cfwd Sel2 Dest Forward number for Call Forward Selective 2

Cfwd Sel3 Dest Forward number for Call Forward Selective 3

Cfwd Sel4 Dest Forward number for Call Forward Selective 4

Cfwd Sel5 Dest Forward number for Call Forward Selective 5

Cfwd Sel6 Dest Forward number for Call Forward Selective 6

Cfwd Sel7 Dest Forward number for Call Forward Selective 7

Cfwd Sel8 Dest Forward number for Call Forward Selective 8

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

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

Cfwd Last Caller The Caller number that is actively forwarded to <Cfwd

Last Dest> by using the Call Forward Last activation code

Cfwd Last Dest Forward number for the <Cfwd Last Caller>

3.4.6.2. Speed Dial Settings

Parameter Name Description Default

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

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

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

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

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

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

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

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

3.4.6.3. Supplementary Service Settings

Parameter Name Description Default

CW Setting Call Waiting on/off for all calls Yes

Block CID Setting Block Caller ID on/off for all calls No

Block ANC Setting Block Anonymous Calls on or off No

DND Setting DND on or off No

CID Setting Caller ID Generation on or off Yes

CWCID Setting Call Waiting Caller ID Generation on or off Yes

Dist Ring Setting Distinctive Ring on or off Yes

Secure Call Setting If yes, all outbound calls are secure calls by default No

3.4.6.4. Distinctive Ring and Ring Settings

Parameter Name Description Default

Ring 1 Caller Caller number pattern to play Distinctive Ring/CWT 1

Ring 2 Caller Caller number pattern to play Distinctive Ring/CWT 2

Ring 3 Caller Caller number pattern to play Distinctive Ring/CWT 3

Ring 4 Caller Caller number pattern to play Distinctive Ring/CWT 4

Ring 5 Caller Caller number pattern to play Distinctive Ring/CWT 5

Ring 6 Caller Caller number pattern to play Distinctive Ring/CWT 6

Ring 7 Caller Caller number pattern to play Distinctive Ring/CWT 7

Ring 8 Caller Caller number pattern to play Distinctive Ring/CWT 8

Default Ring Default ringing pattern, 1 – 8, for all callers 1

Default CWT Default CWT pattern, 1 – 8, for all callers 1

Hold Reminder Ring Ring pattern for reminder of a holding call when the

phone is on-hook Call Back Ring Ring pattern for call back notification None Cfwd Ring Splash

Len Cblk Ring Splash

Len VMWI Ring Splash Len

Duration of ring splash when a call is forwarded

(0 – 10.0s)

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

10.0s)

Duration of ring splash when new messages arrive

before the VMWI signal is applied (0 – 10.0s) VMWI Ring Policy The parameter controls when a ring splash is played

when a the VM server sends a SIP NOTIFY message

to the SPA 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 Ring On No New VM If enabled, the SPA will play a ring splash when the

VM server sends SIP NOTIFY message to the SPA

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

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.

2. Feature not yet available.

None

0 0 .5 New VM

Available

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Sipura Technology SPA2002-ER - Earthlink Truevoice Phone Adpt, SPA-2000, SPA-1000, SPA-3000 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual