YV2 Administration

YV2 Administration Jan.
2007
YV2 Administration
[1/129]
YV2 Administration Jan. 2007
Directory
Directory..............................................................................................................................................2
0. Main Menu.......................................................................................................................................9
1 Address Book..............................................................................................................................10
1.1 Find....................................................................................................................................10
1.1.1 Find.【OK】.................................................................................................................10
1.1.1.1 View Contact.........................................................................................................11
1.1.1.1.1 Edit Contact.Display........................................................................................11
1.1.1.1.2 Edit Contct.Protocol........................................................................................11
1.1.1.1.3 Edit Contact.Email-Like Address....................................................................12
1.2 Add....................................................................................................................................12
1.2.1 Add Contact.Display......................................................................................................12
1.2.2 Add Contact.Protocol....................................................................................................13
1.2.3 Add Contact.Email-Like Address..................................................................................13
1.3 Del......................................................................................................................................13
1.3.1 Clear...............................................................................................................................14
1.4 Address Book.【OK】......................................................................................................14
1.4.1 Speed Dials....................................................................................................................14
1.4.2 Call Screen.....................................................................................................................14
1.4.3 Favorite..........................................................................................................................15
1.4.4 IMPP..............................................................................................................................16
1.4.4.1 State.......................................................................................................................16
1.4.4.2 IMPP......................................................................................................................17
1.4.4.2.1 Add..................................................................................................................20
1.4.4.2.2 View Status......................................................................................................21
2 Call History.................................................................................................................................22
2.1 Missed/Dialed/Received Calls...........................................................................................22
2.1.1 View Record..................................................................................................................22
3 Terminal Settings........................................................................................................................24
3.1 Password............................................................................................................................24
3.1.1 Admin/User....................................................................................................................24
3.1.1.1 Passward.Admin.【OK】.....................................................................................25
3.2 Programable Keys..............................................................................................................25
3.2.1 Programmable Keys.【OK】........................................................................................25
3.2.1.1 DTMF....................................................................................................................26
3.2.1.1.1 View DTMF.....................................................................................................26
3.2.1.2 Set Programmable Keys........................................................................................27
3.2.1.2.1 Address Book...................................................................................................28
3.2.1.2.2 Auto-Answer....................................................................................................28
[2/129]
3.2.1.2.3 Call History......................................................................................................28

3.2.1.2.4 Call Detail........................................................................................................28
3.2.1.2.5 Call Return.......................................................................................................29
3.2.1.2.6 Channel Info....................................................................................................29
3.2.1.2.7 Do Not Disturb (DND)....................................................................................30
3.2.1.2.8 DTMF List.......................................................................................................30
3.2.1.2.9 DTMF Relay by INFO.....................................................................................30
3.2.1.2.10 Favorite..........................................................................................................30
3.2.1.2.11 Forward..........................................................................................................30
3.2.1.2.12 IMPP..............................................................................................................30
3.2.1.2.13 Logout............................................................................................................30
3.2.1.2.14 Message.........................................................................................................30
3.2.1.2.15 Network Status...............................................................................................30
3.2.1.2.16 No Function...................................................................................................31
3.2.1.2.17 Register..........................................................................................................31
3.2.1.2.18 Reject.............................................................................................................31
3.2.1.2.19 SIP Domain Status.........................................................................................31
3.3 Date/Time..........................................................................................................................31
3.3.1 Zone...............................................................................................................................32
3.3.2 FMT...............................................................................................................................32
3.3.3 DST................................................................................................................................32
3.3.3.1 DST Offset.............................................................................................................33
3.3.3.2 DST Start Date.......................................................................................................33
3.3.3.2.1 DST Start Date.Month.....................................................................................33
3.3.3.2.2 DST Start Date.Day.........................................................................................33
3.3.3.2.3 DST Start Date.Day.Day-Of-Month................................................................33
3.3.3.2.4 DST Start Date.Day.Week-Of-Month.............................................................34
3.3.3.2.5 DST Start Date.Time.......................................................................................34
3.3.4 Set Date/Time................................................................................................................35
3.4 Ringer................................................................................................................................35
3.4.1 Type...............................................................................................................................35
3.4.2 Vol..................................................................................................................................35
3.5 Alert-Info...........................................................................................................................36
3.5.1 Tag.................................................................................................................................36
3.5.2 Ring................................................................................................................................36
3.6 LCD Back-Light................................................................................................................37
3.7 Language............................................................................................................................37
3.8 Alarm.................................................................................................................................37
3.8.1 Regular Alarm................................................................................................................38
3.8.1.1 Time.......................................................................................................................38
3.8.1.2 Regularity..............................................................................................................38
3.8.2 Ringer............................................................................................................................38
3.9 Lock...................................................................................................................................39
[3/129]
4 Call Forward...............................................................................................................................40
4.1 Target Number...................................................................................................................40
4.2 Call Forward.All Calls Forward........................................................................................40
4.3 Call Forward.Busy Forward..............................................................................................41
4.4 Call Forward.No Answer Forward....................................................................................41
5 Preference...................................................................................................................................42
5.1 Call Waiting.......................................................................................................................42
5.2 Dial Timeout......................................................................................................................42
5.3 Hold Recall Timer..............................................................................................................43
5.4 Auto-Redial........................................................................................................................43
5.4.1 Stop Criterion.................................................................................................................43
5.4.2 Retry Interval.................................................................................................................44
5.4.3 Totoal Duration..............................................................................................................44
5.5 Dial Plan............................................................................................................................44
5.5.1 Inter-digit Timeout.........................................................................................................44
5.5.2 LAN Dial.......................................................................................................................45
5.5.3 Call Command...............................................................................................................45
5.5.4 Dial Plan........................................................................................................................51
5.5.5 Hot Line.........................................................................................................................51
5.6 Message Alert....................................................................................................................52
5.7 Auto-Answer......................................................................................................................52
5.7.1 Auto-Answer Control List.............................................................................................53
5.7.1.1 Level......................................................................................................................53
5.7.1.2 AoR........................................................................................................................54
5.8 Unhold on Transfer............................................................................................................54
5.9 Conference Alert................................................................................................................55
Default is enabled..................................................................................................................55
5.10 Xfer on Exit Conf..............................................................................................................55
6 Network......................................................................................................................................57
6.1 Active Status......................................................................................................................57
6.2 General...............................................................................................................................58
6.2.1 Mode..............................................................................................................................58
6.2.2 Static Settings................................................................................................................59
6.2.3 PPPoE Settings..............................................................................................................59
6.2.3.1 User Name.............................................................................................................59
6.2.3.2 Password................................................................................................................59
6.2.3.3 Service Name.........................................................................................................60
[4/129]
6.2.4 DNS Server....................................................................................................................60

6.2.5 Host Name.....................................................................................................................61
6.2.6 Time-to-Live..................................................................................................................61
6.3 RTP Settings.......................................................................................................................62
6.3.1 Port Base........................................................................................................................62
6.3.2 Port Range.....................................................................................................................62
6.3.3 PTP IP ToS.....................................................................................................................62
6.3.4 Symmetric Flow.............................................................................................................63
6.4 Network & Firewall...........................................................................................................64
6.4.1 STUN.............................................................................................................................64
6.4.1.1 STUN Server.........................................................................................................65
6.4.1.2 Diagnose NAT.......................................................................................................65
6.4.2 UDP Traversal................................................................................................................66
6.4.3 Static NAT IP.................................................................................................................67
6.4.4 Dynamic NAT IP...........................................................................................................67
6.4.5 Diagnose UPnP..............................................................................................................67
6.5 SNMP................................................................................................................................68
6.6 VLAN................................................................................................................................69
6.6.1 VLAN ID.......................................................................................................................69
6.6.2 VLAN CoS....................................................................................................................70
7 SIP Settings.................................................................................................................................71
7.1 SIP Service.........................................................................................................................71
7.1.1 Transport........................................................................................................................71
7.1.2 SIP Port..........................................................................................................................71
7.1.3 Expires...........................................................................................................................72
7.1.4 rport................................................................................................................................72
7.1.5 Hold by RFC3261..........................................................................................................73
7.1.6 Signal IP ToS.................................................................................................................73
7.1.7 Session Timer.................................................................................................................74
7.1.8 Keep Alive Timer...........................................................................................................74
7.1.9 Expose SW Ver..............................................................................................................74
7.1.10 SIP Timer T1..............................................................................................................75
7.2 Service Domain.................................................................................................................75
7.2.1 N-th Domain..................................................................................................................76
7.2.1.1 Activation..............................................................................................................77
7.2.1.2 Authentication........................................................................................................77
7.2.1.2.1 Server Login....................................................................................................77
7.2.1.2.2 Server Password...............................................................................................77
[5/129]
7.2.1.2.3 qop Support......................................................................................................78

7.2.1.3 Address-of-Record.................................................................................................78
7.2.1.4 Proxy Server..........................................................................................................78
7.2.1.4.1 Mode................................................................................................................79
7.2.1.4.2 Default Proxy FQDN.......................................................................................79
7.2.1.4.3 Transport..........................................................................................................80
7.2.1.4.4 UDP Port..........................................................................................................80
7.2.1.4.5 TCP Port..........................................................................................................80
7.2.1.5 Registrar.................................................................................................................81
7.2.1.5.1 Mode................................................................................................................81
7.2.1.5.2 Registrar FQDN...............................................................................................81
7.2.1.5.3 Transport..........................................................................................................82
7.2.1.5.4 UDP Port..........................................................................................................82
7.2.1.5.5 TCP Port..........................................................................................................82
7.2.1.6 Auto-Answer..........................................................................................................82
7.2.1.7 Keep NAT Alive....................................................................................................83
7.3 ENUM & E.164.................................................................................................................83
7.3.1 ENUM DNS Suffix........................................................................................................84
7.3.2 Min ENUM Length........................................................................................................84
7.3.3 URI Format....................................................................................................................84
7.3.4 Int’l Access Code...........................................................................................................85
8 Service........................................................................................................................................86
8.1 MWI...................................................................................................................................86
8.1.1 Voice Mailbox AoR.......................................................................................................87
8.1.2 Unsolicited NOTIFY.....................................................................................................88
8.2 Message.............................................................................................................................88
8.2.1 Write Message...............................................................................................................88
8.2.1.1 Compose................................................................................................................88
8.2.1.2 Finish Writing........................................................................................................88
8.2.1.3 Send.......................................................................................................................89
8.2.2 Inbox..............................................................................................................................89
8.2.2.1 Read.......................................................................................................................89
8.2.2.1.1 From/Edit.........................................................................................................90
8.2.2.1.2 Content.............................................................................................................90
8.2.2.1.3 Reply................................................................................................................90
8.2.2.1.4 Forward............................................................................................................91
8.2.3 Outbox...........................................................................................................................91
8.2.3.1 Read.......................................................................................................................92
8.2.3.1.1 To/Edit.............................................................................................................92
8.2.3.1.2 Content.............................................................................................................92
8.2.3.1.3 Resend..............................................................................................................92
8.2.3.1.4 Forward............................................................................................................93
8.2.4 Draft...............................................................................................................................93
8.3 SNTP..................................................................................................................................94
[6/129]
8.3.1 Mode..............................................................................................................................94
8.3.2 Server.............................................................................................................................95
8.4 Auto-Provision...................................................................................................................95
8.4.1 Protocol........................................................................................................................105
8.4.2 Server...........................................................................................................................106
8.4.3 Batch Default...............................................................................................................106
8.4.4 Terminal-Specific.........................................................................................................107
8.4.5 Security........................................................................................................................107
9 Advanced..................................................................................................................................108
9.1 CODEC............................................................................................................................108
9.1.1 Preferences...................................................................................................................108
9.1.2 Packetization................................................................................................................109
9.1.3 Comfort Noise..............................................................................................................110
9.1.4 RFC2833 PT................................................................................................................111
9.1.5 DTMF Relay by INFO.................................................................................................111
9.2 System Admin..................................................................................................................112
9.3 System Status...................................................................................................................112
9.3.1 Network Status.............................................................................................................113
9.3.2 System Up Time...........................................................................................................113
9.3.3 Call Statistics...............................................................................................................113
9.3.3.1 Lastest 50 Calls....................................................................................................113
9.3.3.2 Last 72 Hours/Accumulated................................................................................114
9.3.3.2.1 Connected Calls.............................................................................................114
9.3.3.2.2 Call Attempts.................................................................................................115
9.3.4 System Information......................................................................................................116
9.3.5 Version.........................................................................................................................116
9.4 Log...................................................................................................................................116
9.4.1 Log Component...........................................................................................................117
9.4.2 Logger Type.................................................................................................................117
9.4.3 Network Logger...........................................................................................................118
Appendix A – “Dial Plan”................................................................................................................119
1. Dial Plan...............................................................................................................................119
1.1. Dial Plan Commands.......................................................................................................119
1.2. Dial Plan Rules................................................................................................................120
1.2.1. (In Rule) for Dial Plan Blocking.................................................................................120
1.2.2. ‘P’ Rule for Dial Prefix................................................................................................121
1.2.3. ‘R’ Rule for Enhanced Prefix.......................................................................................121
[7/129]
1.2.4. ‘A’ Rule for Suffix Appending.....................................................................................122

1.2.5. ‘C’ Rule for Call Blocking...........................................................................................122
1.2.6. ‘X’ Rule for Call Blocking and Call Forwarding Blocking.........................................122
1.2.7. ‘D’ Rule for Displaying Caller ID...............................................................................123
1.3. Dial Plan Examples..............................................................................................................123
1.3.1. Example 1....................................................................................................................123
1.3.2. Example 2....................................................................................................................123
1.3.3. Example 3....................................................................................................................124
Appendix B – “Available NTP Servers”..........................................................................................125
[8/129]
0. Main Menu
1 . A d d r e s s B o o k
2 . C a l l H i s t o r y
3 . T e r m i n a l S e t t i n g s
4 . C a l l F o r w a r d
5 . P r e f e r e n c e s
6 . N e t w o r k
7 . S I P S e t t i n g s
8 . S e r v i c e
9 . A d v a n c e d
Find Add Del Back
 【Menu】 : Enter or exit menu mode.
 【↑】and【↓】: Navigate through menu items.
 【OK】: Traverse into selected menu or return to previous menu if no specified function.
 [Back]: return.
 4 soft-buttons on the bottom line: Context-sensitive menus.
Ring Lamp
4x21 LCD with back-light Context-Sensitive

Handset (128x32 pixels) Soft-button
Navigation Keys
Speaker
OK
Hold
Keypad
Transfer MWI
Mute
Redial
Hands-free
Speed Dial
Volume
Menu
Micro-phone
【Context-sensitive Soft-Keys】: 4 context-sensitive soft-keys

【Navigation Keys】
【OK】：Enter or Confirm
【↑】：Move up (per line or per item)
【↓】：Move down (per line or per item)
【←】：Move left (per characters)
[9/129]
【→】：Move right (per characters)

【Hold】：Call Hold
【Transfer】：Call Transfer
【Redial】：Redial the last dialed number.
【Speed Dial】：Speed Dial
【Menu】：Menu
【MWI】：Message Waiting Indication, MWI: Access to voice mail system
【MUTE】： Mute
【SPK】：Hands-free
【↕】: Volume adjustment for ringer, headset, handset, speaker.
1 Address Book
4 . C a l L F o r w a r d
6 . N e t w o r k
8 . S e r v i c e
9 . A d v a n c e d
Find Add Del Back
Max 500 entries.
 [Add]: Add new contact into address book.
 [Find]: Search a contact in address book.
 [Del]: Delete contacts from address book.
 【OK】: Enter address-book sub-menu.
1.1 Find
S e a r c h N a m e :
M i c h a e l
Del Abc../123.. Clear Back

Search for a specific entry on address book. The search criterion is a longest prefix match. If no character is
entered, then it will position on the 1st entry on address book.
 [Del]: Delete one character.
 [Abc..]/[123..]: Toggle between digits and alphanumeric input, where. [Abc..] indicates current input
method is alphanumeric and [123..] indicates digits input.
 [Clear]: Clear all input.
 [Back]: Return
 【OK】: Start to search.
 【←】and【→】: move cursor one position in edit mode.
1.1.1 Find.【OK】
M i c h a e l
M i k e
N i c k
[10/129]
Call Del Back

Address Book List Format:
1. List all contacts in alphanumeric order.
2. Position at the best matched item.
3. User may use keypad to jump to the first contact prefixed with entered alphanumeric character.
4. Alternatively, user may use navigation key 【↑】and【↓】to scroll contact list.
5. Max entries: 500.
 [Call]: Dial to selected contact. Alternatively, You may lift the handset (offhook) or turn speaker on by
pressing 【SPK】 to make a call to selected contact as well.
 [Del]: Delete current contact
 [Back]: Return without any changes
 【OK】: Edit selected contact.
 【↑】and【↓】: Navigate through contacts list.
1.1.1.1 View Contact

D i s p l a y : M i c h a e l
P r o t o c o l : s i p
A d d r : 1 2 3 4 5 6 7 @ I S P . f o o .
Call Save Back
1. Display: it could consist of all ASCII letters except for left and right angle parentheses, ‘<’ and ‘>’. You
may enter the alias of your SIP AoR, phone number of this port, or the display string of this AoR.
2. Protocol: Currently, we support only sip and tel URI. Note: protocol is case-sensitive!
3. Addr: The Email-like address of your SIP address-of-record, AoR, with the user part and the domain part.
For example, if your SIP AoR is “Handsome Boy <sip:[email protected]>”, please enter it as
“[email protected]”.
 [Call]: Dial to this contact. Alternatively, You may lift the handset (offhook) or turn speaker on by
 [Save]: Save changes and return.
 [Back]: Return
 【OK】: Edit selected field.
 【↑】and【↓】: Navigate through itmes.
1.1.1.1.1 Edit Contact.Display

D i s p l a y :
M i c h a e l

Display consist of all ASCII letters except for left and right angle parentheses, ‘<’ and ‘>’. You may enter the
alias of your SIP AoR, phone number of this port, or the display string of this AoR.

 [Back]: Return without any changes.
 【OK】: Save changes and return.
 【↑】and【↓】: move cursor per line in edit mode.
1.1.1.1.2 Edit Contct.Protocol

A d d r : 1 2 3 4 5 @ I S P . f o o . n e
[11/129]
Call tel/sip Back

Currently YV2 supports only sip and tel URI. Note: protocol is case-sensitive!
 [tel]/[sip]: Toggle between available protocols, including “sip” and “tel”. [tel] indicates current protocol
is “tel”, and so on.
 [Back]: Return.
1.1.1.1.3 Edit Contact.Email-Like Address

E m a i l - L i k e A d d r e s s :
1 2 3 4 5 @ I S P . f o o . n e t

The Email-like address of your SIP address-of-record, AoR, with the user part and the domain part. For
example, if your SIP AoR is “Handsome Boy <sip:[email protected]>”, please enter it as

1.2 Add
If reached max allowed entries, show
A d d r e s s - B o o k . A d d :
M a x E n t r y : 5 0 0 !
Back
Otherwise, show
D i s p l a y :
A d d r :
Save Back
4. Display: it could consist of all ASCII letters except for left and right angle parentheses, ‘<’ and ‘>’. You
may enter the alias of your SIP AoR, phone number of this port, or the display string of this AoR.
5. Protocol: Currently, we support only sip and tel URI. Note: protocol is case-sensitive!
6. Addr: The Email-like address of your SIP address-of-record, AoR, with the user part and the domain part.
For example, if your SIP AoR is “Handsome Boy <sip:[email protected]>”, please enter it as
 [Save]: Add contact into address book and return.

 【↑】and【↓】: Navigate through itmes.
1.2.1 Add Contact.Display

D i s p l a y :
M i c h a e l
[12/129]

Display consist of all ASCII letters except for left and right angle parentheses, ‘<’ and ‘>’. You may enter the
alias of your SIP AoR, phone number of this port, or the display string of this AoR.

1.2.2 Add Contact.Protocol

A d d r : 1 2 3 4 5 6 7 @ I S P . f o o .
Save tel/sip Back
Currently YV2 supports only sip and tel URI. Note: protocol is case-sensitive!
 [Save]: Add contact into address book and return.

 [tel]/[sip]: Toggle between available protocols, including “sip” and “tel”. [tel] indicates current protocol
is “tel”, and so on.
1.2.3 Add Contact.Email-Like Address

E m a i l - L i k e A d d r e s s :
1 2 3 4 5 @ I S P . f o o . n e t

The Email-like address of your SIP address-of-record, AoR, with the user part and the domain part. For
example, if your SIP AoR is “Handsome Boy <sip:[email protected]>”, please enter it as

1.3 Del
M i c h a e l
M i k e
N i c k
Call Del Clear Back
List address book.
 [Del]: Delete current contact
 [Clear]: Delete all contacts from address book.
[13/129]
 【OK】: Edit selected contact.

 【↑】and【↓】: Navigate through contacts list.
1.3.1 Clear
C l e a r a l l c o n t a c t s ?
Yes No
 [Yes]: Clear all contacts from address book.
 [No]: Return without any changes.
1.4 Address Book.【OK】

1 . S p e e d D i a l
2 . C a l l S c r e e n
3 . F a v o r i t e
4 . I M P P
Back
 [Back]: Return.
 【OK】: Enter sub-menu
1.4.1 Speed Dials

0 . M i c h a e l
1 . M i k e
2 . N i c k
Call Del Clear Back
Speed Dials list format:
 Available entries: 0~19 (20 entries).
 User may use keypad to jump to specified entry, such as enter ‘9’ to jump to the 9th entry.
 [Call]: Dial to selected contact Alternatively, You may lift the handset (offhook) or turn speaker on by
 [Del]: Remove selected speed dial mapping.
 [Clear]: Remove all speed dial mappings
 [Back]: Return.
 【↑】and【↓】: Navigate through the list.
 【OK】: re-map selected entry
M i c h a e l
M i k e
N i c k
Back
List address book.
 [Back]: return without any changes.
 【OK】: re-map speed dial entry by selected contact.
 【↑】and【↓】: Navigate through the list.
1.4.2 Call Screen

D e v i l W e a r i n g P r a d a
N i c k
S u a n
Call Unblk Add Back
Call Screen List format:
[14/129]
1. List all contacts in black lsit in alphanumeric order.

2. User may use keypad to jump to the first contact prefixed with the entered alphanumeric character.
3. Alternatively, user may use navigation key 【↑】and【↓】to scroll contact list.
4. Max size: 100
 [Unblk]: Remove selected contact from black list and return (Unblock, revoke).
 [Add]: Add a contact from address book into black list.
B l a c k L i s t : :
M a x E n t r y : 1 0 0 !
Back
Otherwise, show address book for user to pick a contact to add black list.
M i c h a e l
M i k e
N i c k
Back
 【OK】: Add selected contact into black list.
 【↑】and【↓】: Navigate through address book.
 [Back]: Return.
 【↑】and【↓】: Navigate through black list.
 【OK】: Check selected record (Read only)
1 4 : 3 7 : 2 1 1 0 / 2 9 / 2 0 0 6
A t t e m p t s : 2 0 1
A o R : D e v i l W e a r i n g P r a
d a < s i p : b o s s @ I S P . f o o . c
o m >
Call Unblk Back
Show
 last time-of-call
 Accumulated call attempts originated from this contact.
 Address-of-record.
 [Call]: Dial to this contact. Alternatively, You may lift the handset (offhook) or turn
speaker on by pressing 【SPK】 to make a call to selected contact as well.
 [Unblk]: Remove this contact from black list (Unblock, revoke).
 [Back]: Return.
 【↑】and【↓】: Scroll Line.
1.4.3 Favorite
M i c h a e l
M i k e
N i c k
Call Add Del Back
Configure favorite contacts for easy access during call-state, where you can activate favorite list by pressing
[DSS] keys.
Favorite List Format:
1. Listed in alphanumeric order.
2. Max entries: 20.
 [Call]: Call to selected contact. Alternatively, You may lift the handset (offhook) or turn speaker on by
[15/129]
 [Add]: Add a new entry from address book into favoriate list.
F a v o r i t e L i s t . A d d :
M a x E n t r y : 2 0 !
Back
Otherwise, show address book for user to pick a contact to add into favoriate list.
M i c h a e l
M i k e
N i c k
Back
 【OK】: Add selected contact into favorite list.
 【↑】and【↓】: Navigate through address book.
 [Del]: Remove selected contact from favorite list.
 [Back]: return without any changes.
 【↑】and【↓】: Scroll list.
 【OK】: Review address-of-record
M i c h a e l < s i p : 1 2 3 4 5 @ I S P
. f o o . n e t >
Call Del Back

Show SIP address-of-record (Read only).
 [Call]: Dial to this contact. Alternatively, You may lift the handset (offhook) or turn
 [Del]: Remove this contact from favorite list.
 [Back]: Return.
1.4.4 IMPP
1 . S p e e d D i a l
2 . C a l l S c r e e n
3 . F a v o r i t e
4 . I M P P
State Back
 [State]: Set terminal’s state.
 [Back]: Return.
 【OK】: Enter sub-menu.
1.4.4.1State
1 . O n - L i n e
2 . A w a y
3 . B e R i g h t B a c k
4 . B u s y
5 . O n t h e P h o n e
6 . O u t t o L u n c h
7 . O f f - L i n e
Back
Configure the Presence state of this terminal as per RFC4480 (Rich Presence Extension to the Presence
Information Data Format).
[16/129]

 【OK】: Set and return.
i. On-Line: Presence state is open and user state is <other>idle</other>
ii. Away: Presence state is open and user state is away.
iii. Be Right Back: Presence state is open and user state is <other>Be Right Back</other> or
any of the following states:
a. dinner
b. meal
c. breakfast
d. worship
iv. Busy: Presence state is open and user state is busy or any of the following states:
a. meeting
b. performance
c. presentation
d. working
e. steering
f. appointment
v. On the Phone: Presence state is open and user state is on-the-phone
vi. Out to Lunch: Presence state is open and user state is lunch
vii. Off-Line: Presence state is closed.
1.4.4.2IMPP
R i c h a r d
P a t r i c k ( r i n g i n g )
M a r i a ( a c t i v e )
Call Add Del Back
IMPP List Format:
1 Show all subscribed Presence, Busy Lamp Filed (BLF), and Shared-Line Appearance (SLA)
status.
1. Show display or user-part of the email-like address whenever possible and listed in
alphanumeric order.
2. Max enteries: 20.
3. For Presence, show for “open” state (on-line) and show for “closed” state (off-line).
4. Busy Lamp Field (Dialog-Info)
 SUBSCRIBE failed:
 SUBSCRIBE ok but not received NOTIFY yet: No Symbol
 Idle (terminated/no-dialogs):
 Busy (confirmed/trying/preceeding):
 Ringing (early):
5. Shared Line Appearance (Call-Info)
 idle:
 alerting/held:
 seized/progressing/active/held-private:
 [Add]: Add a new contact into IMPP list.
If max allowed entries is reached, show
I M P P . A d d :
M a x E n t r y : 2 0 !
Back
 [Del]: Remove selected contact from IMPP list
[17/129]
 [Back]: Return.
 【OK】: review selected IMPP contact.
 【↑】and【↓】: Scroll IMPP list.
 【←】and【→】: Scroll columns on the selected line to view whole records. Scroll half of the line width on
each key press.
YV2 supports the following specifications and integrated them as Instant Message and Presence Protocol,
IMPP, for rich server features integration:
 Busy Lamp fileld (BLF):
This feature is to monitor the status of a contact or a list of contacts, including early
(ringing)/confirmed (conversation)/terminated (idle) states. Refer to RFC3865 and RFC4235 for
detail. Besides, please refer to RFC2387 for Muptipart-Related MIME data type and RFC4662 for
“Event Notify for Resource List”.
a. Failed to SUBSCRIBE to the dialog state of chosen contact!
b. Symbol:
 SUBSCRIBE ok but not received NOTIFY yet:
a. SUBSCRIBE to the dialog state of chosen contact succeeded but has not been NOTIFY yet.
b. Symbol: None
a. SUBSCRIBE to the dialog state of chosen contact succeeded and in idle mode.
b. Symbol:
a. the SUBSCRIBEd contact number is busy.
o One of the dialogs of any dialog-info entity is in confirmed | trying | proceeding state.
o User off-hook is in trying state.
b. Symbol:
a. the SUBSCRIBEd contact number is ringing.
o Not busy
o And one of the dialogs of dialog-info entity is in early state.
b. Symbol:
 Presence:
Monitor the presence state (availability or reachability to take calls) of a contact or a list of contacts.
Refer to RFC3265 for SIP Event Notification, RFC3856 for Presence Event Package and
RFC3863/RFC2387/RFC4662 for “application/pidf+xml”/ ”MIME Muptipart-Related” / “Event
Notify for Resource List” data format in NOTIFY.
a. Failed to SUBSCRIBE to the presence state of chosen contact!
b. Symbol:
a. SUBSCRIBE to the presence state of chosen contact succeeded but has not been NOTIFY
yet.
b. Symbol: None
 On-Line (“open”):
a. At least one of the entities of the SUBSCRIBED presence contact is available/on-line
(NOTIFY status is “open”).
b. Symbol:
 Off-Line (“Closed”):
a. All entities of the SUBSCRIBED presence contact number are unavailable/off-line
(NOTIFY status is “closed”).
b. Symbol:
Note, this terminal will be explicitly “unavailable” whenever:
[18/129]
i. “Do-Not-Disturb”, DND, feature is enabled

ii. “All-Calls-Forward” is enabled,
iii. user explicitly off line by “Logout”.
That is, this terminal will respond a “closed” presence state if ever got Presence SUBSCRIBE
from other terminals. Otherwise, this terminal is “available” with presence state “open”.
 Shared-Line Appearance (SLA):

This feature is for the endpoint and the service delivery platform to maintain consistent
“presentation” information, including
 The relative order of call appearances on a line and the current state of the call appearances.
 Enables call control clients, attendant consoles, and other applications to maintain a
synchronized view of call appearance information.
Please refer to “BROADWORKS SIP ACCESS SIDE EXTENSIONS INTERFACE
SPECIFICATIONS” release 13.0 version 1 from BroadSoft Inc. for detailed implementation.
a. SUBSCRIBE to the chosen shared line failed!
b. Symbol:
a. SUBSCRIBE to the dialog state of chosen contact succeeded but has not been NOTIFY yet.
b. Symbol: None
a. this appearance of chosen shared line is idle
b. Symbol:
a. this appearance of chosen shared line is:
o Seized
o Progressing: making outbound calls.
o Active
o Held-private: be Held and only can be retrieved by the endpoint which held it involved
in this call.
b. Symbol:
a. this appearance of chosen shared line is:
o Alerting: receiving an incoming call.
o Held: Holding the peer and can be retrieved other endpoints.
b. Symbol:
Call to this “Shared-Line Appearance” DSS key will:
2. Gain an empty channel 【A】/【B】 and perform line-seize SUBSCRIBE-NOTIFY transaction
before sending INVITE.
[19/129]
3. On receiving INVITE with Call-Info and “answer-after” parameter present, such as:
Call-Info: <sip:ProxyDNSorIP.com>;appearance-index=3;answer-after=0
 Auto Answer this call if
 “answer-after” parameter is present in Call-Info header
 From Header is a recognized Shared-Line AoR
 appearance-index is configured
 Mapping Rule
 answer-after=0: silent
 answer-after=1: imperious
 answer-after<3: urgent
 Others and present: normal
1.4.4.2.1 Add
1.4.4.2.1.1 Type
1 . B u s y L a m p F i e l d , B L F
2 . P r e s e n c e
3 . S h a r e d - L i n e , S L A
Back
 【OK】: Pick selected item.
1.4.4.2.1.2 Address-of-Record
M i c h a e l
[20/129]
M i k e
N i c k
Back
List address book.
 【OK】: Add selected contact into IMPP list.
 【↑】and【↓】: Navigate through the list. Type: BLF,
Presence or SLA
1.4.4.2.2 View Status
P r e s e n c e
o p e n : s o h o < s o h o @ F o o . n
c l o s e d : f r e d < 1 2 3 @ I S P .
Call IM Renew Back
IMPP entry format:
1. Show IMPP type: Presence, SLA or BLF (Fixed, not scrollable)
2. Show each status in the list as:
o Status
o Full SIP address-of-record
3. Show aggregrated status and subscribed AoR on the last line.
4. For Presence, show for “open” state (on-line) and show for “closed” state (off-line).
5. Busy Lamp Field (Dialog-Info)
6. Shared Line Appearance (Call-Info)
 idle:
 alerting/held:
 seized/progressing/active/held-private:
 [IM]: Send Instant Message to selected item.
 [Renew]: Refresh status by re-SUBSCRIBE immediately.
 [Back]: Return.
 【↑】and【↓】: Scroll lines.
 【OK】: review selected IMPP contact.
B L F : R i n g i n g
Status Line:
M i c h a e l < s i p : 1 2 3 4 5 @ I S P
Type+Status
f o o . n e t >
Call Back
Show (Read only)
1. Type (BLF, SLA, Presence): Status
2. SIP address-of-record
 [Call]: Dial to this contact. Alternatively, You may lift the handset (offhook) or turn speaker
on by pressing 【SPK】 to make a call to selected contact as well.
 [Back]: Return.
[21/129]
2 Call History
6 . N e t w o r k
8 . S e r v i c e
9 . A d v a n c e d
Missed Dialed Recvd Back
 [Missed]: Show the lastest 20 missed calls.
 [Dialed]: Shwo the latest 20 dialed numbers.
 [Recvd]: Show the lastest 20 received calls.
 [Back]: Return
2.1 Missed/Dialed/Received Calls

1 . R i c h a r d
2 . H o r a c e
3 . M a r r y
Call Del Clear Back
Missed Call list format:
 List all valid records with lastest call first.
By “latest”, it refers to time sequence rather than the recorded “time-of-call”; because YV2 has to
synchronize its time with network time server by SNTP, such that an actually “latest” call may recorded
as early as 2006 if YV2 fails to synchronize its time by either SNTP or user manually configuration.
 User may use keypad to jump to specified entry, such as enter ‘9’ to jump to 9th entry.
 [Call]: Dial to selected record. Alternatively, You may lift the handset (offhook) or turn speaker on by
 [Del]: Remove selected record.
 [Clear]: Remove all records in this list.
 [Back]: Return.
 【OK】: Review selected record
2.1.1 View Record

1 4 : 3 7 : 2 1 1 0 / 2 9 / 2 0 0 6
M i c h a e l < s i p : 1 2 3 4 5 @ I S P
. f o o . n e t >
Call Del Edit Back
Show (Read only)
1. time-of-call
2. SIP address-of-record
 [Del]: Delete this record.
 [Back]: Return.
 【↑】and【↓】: Scroll Line.
 【OK】/ [Edit]: Edit address-of-record to save to address book
[22/129]
A d d r : 1 2 3 4 5 @ I S P . f o o . n e
Call Save To AB Back
 [Call]: Dial to this contact.
 [Save]: Save changes and return.
 [To AB]: Save address-of-record into address book
 【OK】: Edit selected item.
 【↑】and【↓】: Navigate through items.
[23/129]
3 Terminal Settings
6 . N e t w o r k
8 . S e r v i c e
9 . A d v a n c e d
Back
 [Back]: Return.
 【OK】: Enter submenu.
1 . P a s s w o r d
2 . P r o g r a m m a b l e K e y s
3 . D a t e / T i m e
4 . R i n g e r
5 . A l e r t - I n f o
6 . L C D B a c k - L i g h t
7 . L a n g u a g e
8 . A l a r m
9 . P h o n e L o c k
Admin User Back
 [Back]: Return.
 【OK】: Enter submenu.
 【↑】and【↓】: Navigate through menu items.
3.1 Password
1 . P a s s w o r d
3 . D a t e / T i m e
4 . R i n g e r
5 . A l e r t - I n f o
7 . L a n g u a g e
8 . A l a r m
Admin User Back
Set the passwords to administrative and user-level account. These passwords will be needed while unlocking
the phone set (menu Lock), TELNET into the phone, logging in for web configuration, downloading or upgrading
image from network by TFTP/HTTP, and while network administrator tries to remotely configure your phone set
by flash update or rlogin.
 [Admin]: Modify password to administrator’s account
 [User]: Modify password to user’s account.
 [Back]: Return.
3.1.1 Admin/User
O l d : * * * * * * * * * *
N e w : * * * * * * * * * *
R e t y p e : * * * * * * * * * *
[24/129]
Save Back
Show 10 star signs, ‘*’, for security reasons.
 【OK】: Modified selected items
3.1.1.1 Passward.Admin.【OK】
O l d : * * *
N e w :
R e t y p e :
Del Clear Back
Password consists of [0-9*#] only, and disply star sign, ‘*’, for security reasons.
 【OK】: Finish editing.
 【←】and【→】: Move cursors.
Default password is “0000”.
3.2 Programable Keys

1 . P a s s w o r d
3 . D a t e / T i m e
4 . R i n g e r
5 . A l e r t - I n f o
. L C D B a c k - L i g h t
7 . L a n g u a g e
8 . A l a r m
Back
 [Back]: Return.
3.2.1 Programmable Keys.【OK】

1 . I M P P
2 . R e g i s t e r
3 . D T M F
4 . F a v o r i t e L i s t
5 .
6 .
7 .
8 .
DTMF Back
 [DTMF]: Configure predefined DTMF list for easy access to IVR system
 [Back]: Return.
 【OK】: Re-map selected programmable keys.
Configurable 8 programmable keys. These 8 user-defined keys, excluding those NO-Function keys, will
be shown during call state whenever [DSS] is pressed.
3 √ D T M F R e l a y b y I N F O
4 √ A u t o - A n s w e r
[25/129]
5 . D N D
6 . S I P D o m a i n S t a t u s
7 . F o r w a r d
8 √ R e g i s t e r
Back
 [Back]: return.
 【OK】: Activate selected feature
 【↑】and【↓】: navigrate through list.
a. Show ‘√’ to indicate activation status of “Auto-Answer”, “DND”, “DTMF Relay by INFO”;
and unmap of these features will turn off selected feature as well .
b. “Register”: Shoe ‘√’ to indicate successfully registered to all active SIP service domains and a
‘X’ symbol to indicate none of the active domains are registered. No symbols shown if register
to any of the active SIP service domains succeeded.
c. Otherwise promp to selected menu for further interaction.
3.2.1.1DTMF
C i t y B a n k A n t : 0 0 1 0 0 1 0
M a s t e r C a r d : 1 2 3 4 5 6 7 8 9
P I N : 8 8 8 0 0 0 8 8 8
Dial Add Del Back
Specify a sequence of DTMF keys to dial in pre-dial phase or to transmit after call setup. This is shortcut to
transmit a sequence of preconfigured DTMF keys, which aims to facilitate IVR system interaction. For example,
you may configure your personal ID or banking account, and just activate corresponding DTMF entry to generate
those pre-configured DTMF tones whenever appropriate (such as inquired by tele-banking system). If a DTMF
entry is activated during digit-collecting phase, then those DTMF keys will be collected in a manner identical to
those manually pressed by user.
Acceptable keys are [0-9 | * | # | p | ,], where ‘p’ and ‘,’ (comma) stands for one-second pause. Depending on
the configuration, such DTMF sequence may be transmitted to the peer either inbandly or outbandly (rfc2833 or
INFO). You may also configure it to use as server feature access code during conversation, such as configuring it
as “&76”, and it will send DTMF “&76” “as is” during conversation.
DTMF List Format:

1. Show all DTMF entries in alphanumeric order by tags.
2. Max entries count: 10
 [Dial]: Dial selected DTMF string.
 [Add]: Add new DTMF entry.
D T M F . A d d :
M a x E n t r y : 1 0 !
Back
 [Del]: Remove selected entry list
 [Back]: Return.
 【OK】: Edit selected entry.
 【↑】and【↓】: Scroll list.
3.2.1.1.1 View DTMF

1 . T a g : C i t y B a n k A n t
2 . D T M F : 0 0 1 0 0 1 0 0 1 0 0 1 0 0
Dial Save Back
[26/129]
1. Tag: User-friendily display associated with this DTMF sequence numbers.

2. DTMF: A sequence of DTMF keys to dial in pre-dial phase or to transmit after call setup. Acceptable
keys are [0-9 | * | # | p | ,], where ‘p’ and ‘,’ (comma) stands for one-second pause. Depend ing on the
configuration, such DTMF sequence may be transmitted to the peer either inbandly or outbandly
(rfc2833 or INFO).
 [Dial]: Dial selected DTMF string.
 [Save]: Save changes and return
 【OK】: Edit selected entry
3.2.1.1.1.1 Edit DTMF

1 . T a g : C i t y B a n k A n t
2 . D T M F : 0 0 1 0 0 1 0 0 1 0 0 1 0 0

1. Tag: Specify user-friendily display associated with this DTMF sequence numbers.
2. DTMF: Specify a sequence of DTMF keys to dial in pre-dial phase or to transmit after call setup.
Acceptable keys are [0-9 | * | # | p | ,], where ‘p’ and ‘,’ (comma) stands for one-second pause. Depending
on the configuration, such DTMF sequence may be transmitted to the peer either inbandly or outbandly
(rfc2833 or INFO). You may also configure it to use as server feature access code during conversation,
such as configuring it as “&76”, and it will send DTMF “&76” “as is” during conversation.

3.2.1.2 Set Programmable Keys

A d d r e s s B o o k
A u t o - A n s w e r
C a l l D e t a i l
C a l l H i s t o r y
C a l l R e t u r n
C h a n n e l I n f o
C O D E C
D o N o t D i s t u r b ( D N D )
D T M F L i s t
D T M F R e l a y b y I N F O
F a v o r i t e
F o r w a r d
I M P P
L o g o u t
M e s s a g e
N e t w o r k S t a t u s
N o F u n c t i o n
P a c k e t i z a t i o n
P h o n e L o c k
R e g i s t e r
R e j e c t
S I P D o m a i n S t a t u s
Back
[27/129]
3.2.1.2.1 Address Book

Shortcut to Address Book menu.
3.2.1.2.2 Auto-Answer
Auto-answering incoming calls on idle mode system-wide.
 Press this DSS to toggle auto-answer mode.
 Once enabled, it will Show check symbol, ‘√’, along with this feature to indicate its current status,
and the phone will auto-answer all incoming calls arrive on idle mode.
 Unmap this DSS feature will NOT turn auto-answer off.
 This is a shortcut key to configure 『Preferences』=>”Auto-Answer”.
1 √ A u t o A n s w e r
3 . D N D
Back
3.2.1.2.3 Call History

Enter “2.Call History” / “Missed Call” menu directly if there are any unread missed call; otherwise enter
“Call History” menu directly.
3.2.1.2.4 Call Detail

1 . M i c h a e l
2 . J a s o n
3 . F o x
Call Back
Show 10 call detail records of the latest connected and finished calls (either incoming or outgoing). Those
records (with their caller IDs, AoR, shown) are sorted by their finished time with latest comes first. Besides, they
are volatile in memory such that they will be clean up every time the system reboots.
 [Call]: Dial to selected item. Alternatively, You may lift the handset (offhook) or turn speaker on by
 [Back]: Return
 【↑】and【↓】: Navigate through items.
 【OK】: View record.
1 . F r o m
2 . T i m e o f C a l l
3 . D u r a t i o n : 2 3 : 1 0
4 . C O D E C : G . 7 2 3 . 1 - 5 . 3 k
5 . C o n t a c t
6 . U s e r A g e n t
Call RTP Media Back
o [Call]: Dial to selected item. Alternatively, You may lift the handset (offhook) or turn speaker
on by pressing 【SPK】 to make a call to selected contact as well.
o 【↑】and【↓】: Navigate through items.
o [RTP]: show local and remote RTP session, with port in rtp/rtcp UDP port pair.
L o c a l R T P :
I P : 1 9 2 . 1 6 8 . 1 9 2 . 1 6 8
P o r t : 1 2 3 4 4 / 1 2 3 4 6
R e m o t e R T P :
I P : 1 0 . 0 . 2 4 . 3 3
P o r t : 8 0 0 0 / 8 0 0 1
[28/129]
Back
 【↑】and【↓】: Scroll Lines
 [Back]: return
o [Media]: Show media traffic information. The media-related information will be available only
when the call lasted for more than 20 seconds.
T x P a c k e t : 2 9
R x P a c k e t : 1 8 9 6
T x B y t e : 5 6 0 K B
R x B y t e : 3 4 4 5 K B
Back
 [Back]: return
o From/To: SIP address-of-record of the peer.

M i c h a e l < s i p : 1 2 3 4 @ I S P .
c o m >
Call Back
 [Call]: Call to this address. Alternatively, You may lift the handset (offhook) or turn
 [Back]: return
o Time of Call
1 4 : 1 2 : 3 4 1 2 / 1 6 / 2 0 0 6
Back
 [Back]: return
o Duration: call active time (show mm:ss in-line).
o CODEC: CODEC employed for the call (show in-line).
o Contact: IP of the peer’s CPE.
M i c h a e l < s i p : 1 2 3 4 @ 1 9 2 .
1 6 8 . 1 . 3 3 >
Call Back
[Call]: Call to this address. Alternatively, You may lift the handset (offhook) or turn
 [Back]: return
o User Agent: The phone tool used by the peer for this call.
I P S I P P h o n e / 2 . 2 3 . 8
Back
 [Back]: return
3.2.1.2.5 Call Return

Make a call back to the last incoming numbers (missed or received call). It differs from server supported call
return, such as “*69”, in that it will find the latest incoming calls from “Missed calls” and “Received calls” then
dial out the number.
3.2.1.2.6 Channel Info

1 . F r o m
2 . C o n t a c t
3 . C O D E C : G . 7 2 3 . 1 - 5 . 3 k
[29/129]
4 . U s e r A g e n t
Call RTP A/B Back
Show information of the current active calls on each channel. Those records (with their caller IDs, AoR,
shown) are volatile in memory such that they will be clean up every time the system reboots.
 [A]/[B]: Toggle between channel A and B, with [A]/[B] indicate current channel.
3.2.1.2.7 Do Not Disturb (DND)

Toggle switch to enable or disable Do-Not-Disturb, DND, feature. Once enabled all incoming calls will be
forwarded unconditionally to the forwarded target number if applicable; otherwise, incoming calls will be turned
down as “486 Busy Here”. Unmap of this feature will cancel DND as well.
Show check symbol, ‘√’, along with this feature to indicate its current status once enabled.
3 . D N D
Back
3.2.1.2.8 DTMF List

Shortcut to invoke 『3.Terminal Settings』/ 『DTMF』 menu.
3.2.1.2.9 DTMF Relay by INFO

This is a shortcut key to configure 『Advanced』/『CODEC』=>”DTMF Relay by INFO” directly. Once
enabled, all pressed DTMF keys (0-9*#&) during a call will be sent by SIP INFO method; otherwise those DTMF
keys will be transmitted by DTMF over RTP (either by RFC2833 or DTMF tone mixed with voice stream).
Unmap of this key will not disable this feature. Show check symbol, ‘√’, along with this feature to indicate
its current status once enabled.
3 . D N D
Back
3.2.1.2.10 Favorite
Shortcut to invoke 『1.Address Book』/ 『Favor』 menu.
3.2.1.2.11 Forward
Forward incoming ringing calls on call arrival to preconfigured forwarding target number; otherwise enter
“4.Call Forward” menu. Although [Forward] will be one of the context-sensitive soft-buttons on call-waiting, it
may disappear if user does not process this incoming call before this context changed, this programmable key can
always be pressed at any time.
3.2.1.2.12 IMPP
Shortcut to invoke 『1.Address Book』/ 『IMPP』 menu.
3.2.1.2.13 Logout
Unregister all activated SIP address-of-records and cease regular auto-registration scheduling to keep user off-
line until 【 Register 】 feature is activated to be on-line again. Once logged out, it appears to be in “closed”
Presence state.
After reboot the phone-set will clear this status and register to all activated SIP address-of-records afte startup.
3.2.1.2.14 Message
Access to out-of-dialog instant messaging: on pressing, it will enter 『8.Service』/ 『Message』/『Inbox』if
there are any unread new messages; otherwise enter 『8.Service』/ 『Message』menu.
[30/129]
3.2.1.2.15 Network Status

Shortcut to invoke 『9.Advanced』/ 『Network Status』 menu.
3.2.1.2.16 No Function
No specific function for this key.
3.2.1.2.17 Register
2 . D N D
3 . R e g i s t e r
Back
Register to all activated domains on-demand, and appear to be in “open” Presence State unless it is in
“DND”.
3.2.1.2.18 Reject
Reject incoming ringing calls on arrival as “486 Busy”. Although [Busy] will be one of the context-sensitive
soft-buttons on call-waiting, it may disappear if user does not process this incoming call before this context
changed, this programmable key can always be pressed at any time.
3.2.1.2.19 SIP Domain Status

3 √ S I P D o m a i n S t a t u s
Back
Show a symbol preceding each active SIP service domain:
a. ‘√’ to indicate successfully registered to all active SIP service domains.
b. ‘x’ to indicate none of the active domains are registered.
c. No symbol to indicate registered to any of the active SIP service domain succeeded.
If pressed (activated), it will show registration status of each active SIP service domain.
1 √ I S P . f o o . n e t
2 x S i p . I S P . c o m
3 . I S P . p r i v a t e . b i z
Back
Show a symbol preceding each active SIP service domain:
a. ‘√’ to indicate a register success state
b. ‘x’ to indicate a register failed state
c. ‘.’ To indicate a registering state.
YV2 supports up to 3 SIP service domains, which this terminal may register to; and you may circle active
service domains by pressing [Realm] soft-button while making outbound calls.
foo.biz SIP.isp.comfoo.bizAuto
3200 -locate
3.3 Date/Time
1 . P a s s w o r d
3 . D a t e / T i m e
4 . R i n g e r
5 . A l e r t - I n f o
[31/129]
7 . L a n g u a g e
8 . A l a r m
Zone FMT DST Back
YV2 will synchronize its time by Simple Network Time Protocol, SNTP, with network time server regularly if
SNTP is enabled. To keep the time you manually set here, you must disable SNTP.
 [Zone]: Set time-zone

 [FMT]: Set date/time display format
 [DST]: Daylight Saving Time (DST) adjustment.
 [Back]: Return.
 【OK】: Adjust system date and time.
3.3.1 Zone
T i m e Z o n e
G M T : ± 0 0 : 0 0
± Back
This terminal will synchronize its time by Simple Network Time Protocol, SNTP, with network time server
regularly if SNTP is enabled. Before setting the GMT time inquired from NTP server, it will convert it into local
time based on the time zone as well as daylight saving adjustment (see below).
To leave the current system date and time set from menu-“Date/Time” intact, you could either disable SNTP
from menu-”8.Service” / “SNTP”.
System default is GMT+00:00.
 [±]: Toggle between ‘+’ and ‘-‘.

 [Back]: Return without changes.
 【←】and【→】: Navigate through fields.
 [0-9]: enter values.
 ±: Press any key [0-9*#] to toggle between ‘+’ and ‘-‘
3.3.2 FMT
1 . M M / D D / Y Y Y Y
2 . D D / M M / Y Y Y Y
3 . Y Y Y Y / M M / D D
4 . Y Y Y Y / D D / M M
5 . Y Y Y Y - M M - D D
Back
Specify the date format for display purpose (such as time-of-call in records of call history).
MM: 2-digit month
DD: 2-digit day of month
YYYY: 4-digit year

 【↑】and【↓】: Navigate through items
3.3.3 DST
1 . O f f s e t : + 2 : 0 0
2 . S t a r t D a t e
3 . S t o p D a t e
[32/129]
Back
Daylight Saving Time (DST) adjustment.
 [Back]: Return
 【OK】: Edit selected items
3.3.3.1 DST Offset

D S T O f f s e t
O f f s e t : ± 0 : 0 0
± Back
Offset from this terminal’s time when DST is in effect. When DST is over, the specified offset is no longer
applied to this terminal’s time. Valid values are H:MM, -H:MM or +H:MM.
System default is 0:00 to disable DST adjustment.
 [±]: Toggle between ‘+’ and ‘-‘.

 [0-9]: enter values.
3.3.3.2 DST Start Date

M o n t h : A p r i l
D a y : S u n , L a s t W e e k
T i m e : 0 2 : 0 0
Back
 【OK】: Edit selected tiem.
3.3.3.2.1 DST Start Date.Month

J a n u a r y
F e b u r a r y
M a r c h
Back
3.3.3.2.2 DST Start Date.Day

1 . M o d e : D a y - o f - M o n t h
2 . D a y - o f - M o n t h
3 . W e e k - o f - M o n t h
WoM/DoM Back
 [WoM]/[DoM]: Toggle between [Week-of-Moth] and [Day-of-Month].
[33/129]
3.3.3.2.3 DST Start Date.Day.Day-Of-Month

D a y - o f - M o n t h
[ 1 - 3 1 ] 1
Del Back
 [Del]: delete an input digit.
 [0-9]: Enter values.
 【OK】: Save changes and return..
3.3.3.2.4 DST Start Date.Day.Week-Of-Month

W e e k : L a s t W e e k
D a y - o f - W e e k : S u n
Back
 【OK】: Edit selected item.
 Week
0 . L a s t W e e k
1 . 1 s t W e e k
2 . 2 n d W e e k
3 . 3 r d W e e k
4 . 4 t h W e e k
5 . 5 t h W e e k
6 . 6 t h W e e k
Back
 【OK】:Set and return
 Day-of-Week
0 . S u n
1 . M o n
2 . T u e
3 . W e d
4 . T h u
5 . F r i
6 . S a t
Back
 【OK】: Set and return
3.3.3.2.5 DST Start Date.Time
[34/129]
Press 【OK】 to edit selected item

M o n t h : A p r i l
D a y : S u n , L a s t W e e k
T i m e : 0 2 : 0 0
Save Back
Time of day on which DST begins in HH:MM 24 hour’s format.

 【←】and【→】: Navigate through fields in a item.
3.3.4 Set Date/Time

T o d a y
2 1 : 3 7 : 2 5 1 1 / 2 3 / 2 0 0 6
Save Back
Specify current date and time on this terminal. Enter the current date, where the time is in 24-hour format
and the date format depends on the “3.Date/Time” / “FMT” configuraiton.
Note: this terminal will synchronize its time by Simple Network Time Protocol, SNTP, with network time
server regularly if SNTP is enabled. To keep the time you manually set here, please disable SNTP from
menu-”SNTP mode” first.

3.4 Ringer
1 . P a s s w o r d
3 . D a t e / T i m e
4 . R i n g e r
5 . A l e r t - I n f o
7 . L a n g u a g e
8 . A l a r m
Type Vol Back
Specify the ring to play on call arrival.
 [Type]: Configure ringer type.

 [Vol]: Configure ringer volume
 [Back]: Return.
3.4.1 Type
1 . R i n g e r - 0
2 . R i n g e r - 1
3 . R i n g e r - 2
4 . R i n g e r - 3
5 . R i n g e r - 4
6 . R i n g e r - 5
7 . R i n g e r - 6
8 . R i n g e r - 7
[35/129]
9 . R i n g e r - 8
1 0 . R i n g e r - 9
Back
 【OK】: Save and return
 【↑】and【↓】: Navigate (and play) through ringers.
3.4.2 Vol
R i n g e r V o l u m e :
< < < < < <
Back
Adjust volume real time by playing selected ring.
 [Back] /【OK】: Save changes and return.

 Use Volume Keys:【Up】and【Down】, and navigation keys:【↑】【
, ↓】【
, ←】and【→】to adjust volume.
3.5 Alert-Info
1 . i n - H i : R i n g e r - 1
2 . o u t - l o : R i n g e r - 4
3 . i n t r a - h i : R i n g e r - 3
Tag Ring Back
YV2 supports “Alert-Info” header in the first SIP INVITE message as per RFC3261, “Alert-Info”
header dictates the phone to use an alternative ringing tone, which is specific for that call. The header should be in
a format similar to “Alert-Info: <http://MediaServer.ISP.com/Announce.pcmu>;AnyParameter=xxxx” or “Alert-
Info: xxxx”, where “xxxxx” (case-insensitive) is the tone tag for one of the available ringing tones (0~11). This is
useful to distinguish calls, for example local calls from calls coming from PSTN. Besides, this header is ignored
for re-INVITE. If the specified tone is out of range, the current ringer is used.
From this page, user may re-map which tone should be played when “xxxx” tone tag is specified. For
example, if “Alert-Info: ringtone-0” is mapped to “Ringer7”, then “Ringer7” will be played whenever “Alert-
Info: ringtone-0” or “Alert-Info: <http://SIP.ISP.com/file.pcmu>;anyParameter=ringtone-0” is specified in the
initial INVITE.
Default is to map “ringtone-0” to Ringer0, “ringtone-1” to Ringer1, and so on.
 [Tag]: Configure the tag of selected alert-info

 [Ring]: Configure the ringer type of selected alert-info
 [Back]: Return.
3.5.1 Tag
T a g :
i n - H i

Specif the tag to be compared to in “Alert-Info” header. This tag is case-insensitive and can appear as header
value or in any header parameter value.

[36/129]
3.5.2 Ring
1 . R i n g e r - 0
2 . R i n g e r - 1
3 . R i n g e r - 2
4 . R i n g e r - 3
5 . R i n g e r - 4
6 . R i n g e r - 5
7 . R i n g e r - 6
8 . R i n g e r - 7
9 . R i n g e r - 8
1 0 . R i n g e r - 9
Back
Specify which ring to play when a match is found in “Alert-Info” header.

 Play selected ring real time.
3.6 LCD Back-Light

1 . O f f w h e n e v e r I d l e
2 . O f f a f t e r 1 0 S e c o n d s
3 . O f f a f t e r 3 0 S e c o n d s
4 . O f f a f t e r 1 M i n u t e
5 . O f f a f t e r 2 M i n u t e s
9 . A l w a y s O n
Back
Specify how long the LCD backlite stays on after going into idle state.

3.7 Language
1 . C h i n e s e - S i m p l i f i e d
2 . C h i n e s e - T r a d i t i o n a l
3 . E n g l i h
4 . J a p a n e s e
5 . S p a i n - M e x i c o
Back
Specify the language for UI localization, such as configuration menu. This is reserved for future use and
currently only English is available.

3.8 Alarm
1 . O n e - t i m e A l a r m
[37/129]
2 . R e g u l a r A l a r m
3 . R i n g e r
Back
Set up temporary alarm and / or regular alarm.
On timer expiration (either alarm type), the phone will start ringing for 60 seconds with increasing volume
(stepping from the current speaker volume). Press any key after timer expiration will stop its activation (ringing);
otherwise, it will repeat the ringing process after 5 minutes ‘till any key is pressed or it has been 30 minutes after
its expiration.
 [Back]: Return.
 【OK】: Enter sub-menu.
3.8.1 Regular Alarm

1 . A c t i v a t e
2 . T i m e : 0 6 : 3 0
3 . R e g u l a r i t y
On/Off Back
Configure a timer which triggers regularly.
 [On]/[Off]: Toggle switch to enable or disable alarm. Show check symbol ‘√’ in-line if enabled!
 [Back]: Return.
 【OK】: Enter sub-menu
3.8.1.1 Time
Press 【OK】to edit the time of this regular alarm of when it triggers.
1 . A c t i v a t e
2 . T i m e : 0 6 : 3 0
3 . R e g u l a r i t y
Save Back
 [Save]/【OK】: Save changes and return.
 【←】and【→】: Navigate through fields in a item.
3.8.1.2 Regularity
0 . S u n
1 √ M o n
2 . T u e
3 . W e d
4 √ T h u
5 . F r i
6 . S a t
Save On/Off Back
Configure the regularity when the timer triggers.
 [Save]/【OK】: Save changes and return.

 [On]/[Off]: Toggle switch to enable or disable alarm on selected day. Only “On” will have indication.
Show check symbol ‘√’ in-line if enabled!
[38/129]
3.8.2 Ringer
1 . R i n g e r - 0
2 . R i n g e r - 1
3 . R i n g e r - 2
4 . R i n g e r - 3
5 . R i n g e r - 4
6 . R i n g e r - 5
7 . R i n g e r - 6
8 . R i n g e r - 7
9 . R i n g e r - 8
1 0 . R i n g e r - 9
Back
Position on current values on entering.
 Play selected ring real time.
3.9 Lock
1 . O f f
2 . F u n c t i o n K e y s
3 . T a k e C a l l O n l y
Back
 Off: The terminal is not in security mode.
 Function Keys: This terminal is locked on security level. In security mode, the following keys will be
protected (key press will be ignored):
 Menu Key. 【Menu】
 Redial key: 【Redial】
 Message Waiting Indication: 【MWI】
 Context-sensitive soft-keys: [Call History], [Address Book] and [DSS] keys.
 Take Call Only: Lock the phone set such that no outbound calls could be made and the phone could
not be configured either. That is, the phone would leave in “receive only” mode (No dial tone could be
heard).
Once locked in either Function Keys (security) mode or take call only mode, you need either user-level or
privileged password to unlock it.
Un l o c k p a s s wo r d :

[39/129]
4 Call Forward
1 . T a r g e t N u m b e r
2 . A l l C a l l s F o r w a r d
3 √ B u s y F o r w a r d
4 √ N o A n s w e r F o r w ar d
Back
Configure the target number to forward to while this phone is busy or not answered within a predefined
guarding interval. This forwarding number is also employed while the phone is engaged in Do Not Disturb, DND,
mode or while the user presses 【Forward】 key on an incoming waiting call.
The system forwarding rules are checking Do Not Disturb mode first, then All Calls Forward, Busy Forward,
finally going to No Answer Forward while no-answer timer expires.
 [Back]: Return
 【OK】: Enter submenu
4.1 Target Number

M y A l t e r n a t i v e N o < s i p : 1
2 3 @ I S P . f o o . c o m >
Call Reset Del Back

This forwarding number is to forward incoming calls when the phone is busy or not answered within a
predefined guarding interval. It is also employed while the phone is engaged in Do Not Disturb, DND, mode or
while the user activates 【Forward】 feature on an incoming waiting call.
 [Call]: Dial to target number if available. Alternatively, You may lift the handset (offhook) or turn
 [Reset]: Reset target number from address book.
M i c h a e l
M i k e
N i c k
Back
 [Back]: Return without any changes.
 【OK】: Pick selected contact.
 [Del]: Remove target number (no target number for call-forward).
 [Back]: Save changes and return.
 【OK】: Return.
4.2 Call Forward.All Calls Forward

4 √ N o A n s w e r F o r w a r d
On/Off Back
You can configure to unconditionally forward all incoming calls by enable the All Calls Forward feature.
Forwarded calls are logged in the Missed Calls. If this feature is enabled, LCD will show “FWD:TargetAoR” to
remind user that all incoming calls will be forwarded unconditionally thereafter.
Default is disabled.
 [On] / [Off]: Toggle between enable and disable this feature, where [On] and [Off] indicates current
setting. Show check symbol ‘√’ in-line if enabled!
[40/129]
 [Back]: Return
4.3 Call Forward.Busy Forward

On/Off Back
Configure to forward incoming waiting calls when the system is busy, on which time all lines are occupied.
Forwarded calls are logged in the Missed Calls..
Default is enabled
 [Back]: Return
4.4 Call Forward.No Answer Forward

On/Off Timer Back
Configure to forward incoming waiting calls after ringing for a predefined interval. Forwarded calls are
logged in the Missed Calls. When No Answer Forward feature is enabled, the incoming calls will be forwarded to
the Target Number if the call has not been answered for a predefined time (see next section); otherwise the phone
will keep ringing indefinitely.
Default is enabled.
 [Timer]: ]: Configure the time to wait (ringing) before forwarding incoming calls as No Answer. Default
is 30 seconds
N o A n s w e r T i m e r
[ 1 0 - 6 0 ] ( s e c o n d )
3 0
Del Back
o [Del]: Delete one character
o [Back]: Return without any changes
o 【OK】: Save changes and return.
o 【←】and【→】: move cursor one position in edit mode.
[41/129]
5 Preference
1 √ C a l l W a i t i n g
2 . D i a l T i m e o u t
3 . H o l d R e c a l l T i m e r
4 . A u t o - R e d i a l
5 . D i a l P l a n
6 √ M e s s a g e A l e r t
7 . A u t o - A n s w e r
8 . U n h o l d o n T r a n s f e r
9 √ C o n f e r e n c e A l e r t
1 0 . X f e r o n E x i t C o n f
Back
Set up personal call preferences.
 [Back]: Return
5.1 Call Waiting

5 . D i a l P l a n
On/Off Back
Call Waiting notifies the user, who is on an established call, that an additional external call has been present
and is “waiting to be answered.” The waiting call receives normal ringing until it is answered, the incoming
calling party abandons the call, or the ringing cycle timer expires, and the call is given No-Answer-Forward
treatment (if applicable).
When the user has engaged in a call, and some new incoming calls are waiting for answer, the phone will
play an auditable tone twice in 7 seconds to alert the user on active channel. You could change this setting such
that the phone will reply an incoming call a “486 Busy Here” while you are busy. Default is enabled.
 [Back]: Return
5.2 Dial Timeout

D i a l T i m e o u t
[ 1 0 - 6 0 0 ] ( s e c o n d )
3 0
Del Back
Configure the dial timeout if no responses (any provisional responses such as “100 Trying” or “180
Ringing”) from the peer. If the timer expires before the peer starts ringing back, it will play disconnect tone and
prompt the user “Dial timeout”.
System default is 30 seconds.
 [Del]: Delete one character
[42/129]

5.3 Hold Recall Timer

H o l d R e c a l l I n t e r v a l
[ 1 0 - 6 0 0 ] ( s e c o n d )
6 0
Del Back
When some calls are put on hold, such as during a transfer procedure, the phone will play an auditable tone
regularly to alert the user on active channel. Default alerting interval is 60 seconds. Besides, the phone will start
ringing if the handset is placed on-hook and there is a call currently on hold.
The alerting process is as following:
(A) User are in on-hook state:
i. The phone will ring.
ii. The recall state continues ‘till user picks up the call. However, if user does not answer the call
within 1 minute, the call will be disconnected.
(B) User are in off-hook state, either engaged in other calls or about to making calls:
i. Play hold recall tone once to remind the user that some calls are in hold.
ii. Reset the hold recall timer for next time alarm.

5.4 Auto-Redial
1 . S t o p C r i t e r i o n
2 . R e t r y I n t e r v a l
3 . T o t a l D u r a t i o n
Back
Once auto-redial is in progress, it will be automatically canceled if either the stop criterion is met or the total
activation duration is expired.
 [Back]: Return
 【OK】: Edit selected item
 【↑】and【↓】: Navigate through menu items
5.4.1 Stop Criterion

1 . S t o p o n R i n g i n g
2 . S t o p o n C o n n e c t e d
Back
Specify the circumstances to stop auto-redialing.
1. Stop on Ringing: only when the peer starts ringing back will auto-redialing stop (Default).
2. Stop on Connected: Only when the peer picks up will auto-redialing stop.

[43/129]
5.4.2 Retry Interval

R e d i a l I n t e r v a l
[ 1 - 2 5 5 ] ( s e c o n d )
1 5
Del Back
Specify the gap (measured in seconds) between two successive re-dials to avoid overflowing the networks
with fast retries.
Default is 15 seconds.

5.4.3 Totoal Duration

A c t i v e I n t e r v a l
[ 3 0 - 8 6 4 0 0 ] ( s e c o n d )
1 8 0 0
Del Back
Specify the activation duration of this auto-redial feature once starts, measured in seconds. Once the timer
expires, the auto-redial feature will be canceled.
Default is 1800 seconds (30 minutes).

5.5 Dial Plan

1 . I n t e r - D i g i t T i m e o u t
2 √ L A N D i a l
3 . C a l l C o m m a n d
4 . D i a l P l a n
5 . H o t L i n e
Back
 [Back]: Return
5.5.1 Inter-digit Timeout

I n t e r - D i g i t T i m e o u t
[ 3 - 9 ] ( s e c o n d )
4
Del Back
Configure the inter-digit timeout while dialing numbers in digit-collecting phase. Default is 4 seconds. If
you dial the alphabets very often, such as URL, then you may increase the inter-digit timeout to, say, 6 seconds.
While making calls, the first digit timeout is 15 seconds, and the inter-digit timeout after the first digit is defined
here. The phone will dial out the collected digits on timer expired. To speed up the dial process, press [Call] soft-
button whenever finishing dial.

[44/129]
5.5.2 LAN Dial

2 √ L A N D i a l
4 . D i a l P l a n
5 . H o t L i n e
On/Off Back
Configure whether a dial string starting with a star, ‘*’, key signals a LAN dial. For example, if your IP is
“192.168.3.10” and you want to make a direct IP call to another host (either a phone-set or a PC) which IP is
“192.168.3.20”, you may dial “*20” to make a call to it. To specify the UDP port, you may dial “*20**9999” to
call to “192.168.3.20:9999”.
Generally, to make LAN dial, you should use a star sign, ‘*’, as a dot, ‘.’; and double stars, “**”, as a colon,
‘:’, to specify the UDP port the callee listens on. Besides, you may specify the last 1, 2 or 3 fields of the peer’s IP,
plus additional port.
For example, suppose the terminal’s IP is 192.168.3.10.
 Example 1: to call “192.168.3.20:5060”, you may dial:
 ***192*168*3*20**5060 (where “**” as ‘:’ to specify UDP port)
 ***192*168*3*20 (default is call to UDP port 5060)
 ****20# (since the peer locates on the same subnet as you do, you could specify only the last
portion of its IP as shorthand to “192.168.3.20:5060”.
 Example 2: to call “[email protected]:5060
 ***888*192*168*4*90**5060 (where “**” as ‘:’ to specify UDP port)
 ***888*192*168*4*90 (default is call to UDP port 5060)
By default, this feature is enabled. You may want to disable LAN dial if it conflicts with some dial numbers
and/or you want to send those LAN dial strings “as is” (for example, “*123” may be reserved by ISP for server
feature access).
 [On]/[Off]: Toggle between enable and disable this feature, where [On] and [Off] indicates current
 [Back]: Return
 【OK】: Return
5.5.3 Call Command

1 . M S A C :
2 . A n o n y m o u s C a l l : * 6 7
3 . C L I P : * 8 2
4 . M W I :
5 . S e r v e r H o l d :
6 . C a l l P i c k u p : * *
7 . F e a t u r e C o d e 0 :
8 . F e a t u r e C o d e 1 :
9 . F e a t u r e C o d e 2 :
Back
Configure various call commands, such as Calling Line Identification Restriction (CLIR), or Calling Line
Identification Presentation (CLIP), on a per call basis. Alternative, you may change server feature access code to
fit your local regulation, such as change the “Call Return” access call from “*69” to “#69”.
 [Back]: Return
1 . M S A C :
2 . A n o n y m o u s C a l l : * 6 7
3 . C L I P : * 8 2
[45/129]
4 . M W I :
5 . S e r v e r H o l d :
6 . C a l l P i c k u p : * *
7 . F e a t u r e C o d e 0 :
8 . F e a t u r e C o d e 1 :
9 . F e a t u r e C o d e 2 :
 [Del]: Delete one character.
 [Abc..]/[123..]: Toggle between digits and alphanumeric input, where. [Abc..] indicates current
input method is alphanumeric and [123..] indicates digits input.
 [Clear]: Clear all input.
 【OK】: Save changes and return.
 【←】and【→】: move cursor one position in edit mode.
 Conference Server Access Code:

Whenever the [Conf] soft-key is pressed, YV2 will try to initiate a 3-way local conferencing. However, some
ISP provide media server for server-side conference support as well. Thus, if you want to send such access
code to server, such as “*#”, instead of launch a local conference after pressing [Conf] soft-key, you may
define the access code here. If defined, this terminal will send specified MSAC code as DTMF sequence by
either inband or outband (rfc2833 or SIP INFO) during conversation. Leave it blank for 3-way local
conference on pressing [Conf] soft-key.
 CLIR: Calling Line Identification Restriction.
System default is to enable Calling Line Identification Presentation, CLIP, all the time. You may enable the
Call Line Identification Restriction, CLIR on a per-call basis by using the call command dial string. To
enable CLIR on a per-call basis requires that the dial string sequence (typically *67) that users enter on
their dial-pad prior to dialing the phone number match the specified CLIR string defined in the call
command. Whenever the prefix is detected in dial string, it will be stripped off from the original dial string
and send to proxy with CLIR enabled. The SIP INVITE message sent by YV2 would look like:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.3.253:5060;branch=abc7801
Via: SIP/2.0/UDP 192.168.3.51:5060
From: "Anonymous" <sip:[email protected]>;tag=22516
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: "Anonymous" <sip:[email protected]:5060>
User-Agent: IP-Phone/2.0
Content-Length: 171
Content-Type: application/sdp
v=0
o=ipr000dc300051 0 0 IN IP4 192.168.3.5
System default activation code is “*67”.
 CLIP: Calling Line Identification Presentation.

System default is to enable Calling Line Identification Presentation, CLIP, all the time. You may cancel CLIR
by using the CLIP call command dial string.
To enable the Calling Line Identification Presentation, CLIR, on a per-call basis (to counter CLIR) requires
that the dial string sequence (typically *82) that users enter on their dial-pad prior to dialing the phone number
match the specified CLIP string defined in the call command. Whenever the prefix is detected in dial string, it
will be stripped off from the original dial string and send to proxy with CLIP enabled.
System default activation code is “*82”.
 MWI: Message Waiting Indication.

Whenever the【MWI】 is pressed, YV2 will try to make a call to the SUBSCRIBEd MWI account. However,
some ISPs provide voice mailbox access by specific server feature access code. Thus, if you want to send
such server feature access code to server, such as “#”, you may define the access code here. If defined, this
[46/129]
terminal will make an outbound call to the defined code; for example, if “#” is configured, it will call to
“sip:#@ISP.com”. Leave it blank for terminal to call to the SUBSCRIBEd MWI account.
 Server Hold
Whenever the【HOLD】 is pressed, YV2 will try to put the peer on hold by SIP re-INVITE, and retrieve the
peer by SIP re-INVITE if the 【 HOLD 】 is pressed again. Besides, this terminal will play music locally
whenever it is put on hold by the peer. However, some ISP provide hold operation as well as server generated
music-on-hold stream through their servers. Thus, to integrate such server/IP-PBX, you may re-define the
operation of 【HOLD】. Once defined, such as “&”, the terminal will generate the defined server hold access
code as DTMF sequence by either inband or outband (rfc2833 or SIP INFO) during conversation to invoke
server hold features. Leave it blank for terminal to perform HOLD/unHOLD operations by SIP re-INVITE
locally.
 Call Pickup: Default is “**”. YV2 supports directed call pickup as well as group call pickup.
Specify the group pickup code to pick up a ringing call of the same group (RFC4235-Dialog Event Page,
RFC4462-Event Notify for Resource List, RFC2387-Muptipart-Related MIME type, and draft-ietf-sipping-
service-examples-10.txt.
For example, if Bob and Peter are part of a work group at example.com that can pick up each others calls.
Alice calls Bob who does not answer. Peter wishes to pick up the call then he could dial this specified “Call
Pickup” code followed by Bob’s number, such as “**Bob”, where “**” is the assumed “Call Pickup” code.
Then Peter’s terminal will send a SUBSCRIBE to Bob to retrieve the dialog information. Bill then generates
an INVITE with a Replaces to Alice. Alice answers the INVITE and sends a CANCEL to stop Bob's phone
ringing. Note that the order of the CANCEL/ACK sequence in message-9 through message 14 is not
significant.
# Message Content
1 INVITE sip:[email protected] SIP/2.0
Content-Length: 223
[47/129]
Via: SIP/2.0/UDP 192.168.3.57:5060;branch=z9hG4bK469312dfb
From: <sip:[email protected]:5060>;tag=489c13a17
Supported: replaces, timer
CSeq: 66526 INVITE
User-Agent: YV2/1.0
Max-Forwards: 70
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
v=0
o=iprD0E9015189 4553807 4553807 IN IP4 192.168.3.57
s=-
c=IN IP4 192.168.3.57
t=0 0
m=audio 8004 RTP/AVP 0 8 4 18 101
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-16
a=fmtp:4 ptime=30;bitrate=6.3
SIP/2.0 180 Ringing

Content-Length: 0
Server: IPR-720E/1.0.1
To: <sip:[email protected]>;IPDIAL=1;tag=46f66d0e
2
CSeq: 66526 INVITE
SUBSCRIBE sip:[email protected] SIP/2.0

Content-Length: 0
Date: Fri, 23 Jun 2006 16:00:49 GMT
Via: SIP/2.0/UDP 192.168.3.61:5060;branch=z9hG4bK41d57a19
From: <sip:[email protected]>;tag=41c6a9c
Event: dialog
3
Supported: eventlist
CSeq: 40397 SUBSCRIBE
Expires: 30
User-Agent: ATA-740S/1.1.3
Max-Forwards: 70
Accept: multipart/related, application/rlmi+xml, application/dialog-info+xml
4 SIP/2.0 200 OK
Content-Length: 0
Server: ATA-740S/1.1.3
To: <sip:[email protected]>;tag=47251617
[48/129]
NOTIFY sip:[email protected]:5060 SIP/2.0
Content-Length: 396
Content-Type: application/dialog-info+xml
Via: SIP/2.0/UDP 192.168.3.101:25320;branch=z9hG4bK4dae3c77
Subscription-State: terminated
To: <sip:[email protected]>;tag=41c6a9c
From: <sip:[email protected]>;tag=47251617
Event: dialog
CSeq: 2998 NOTIFY
User-Agent: IPR-720E/1.0.1
Max-Forwards: 70
5 <?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info"
version="0 state="full"
entity="[email protected]">
<dialog id="17f9388"
call-id="[email protected]"
local-tag="46f66d0e"
remote-tag="489c13a17"
direction="recipient" // [recipient | initiator]
remote-uri="[email protected]"
remote-target="[email protected]:5060">
<state>early</state> // [trying | early | confirmed | terminated]
</dialog>
</dialog-info>
SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/UDP 192.168.3.101:25320;branch=z9hG4bKdb0e6d
To: <sip:[email protected]>;tag=41c6a9c
From: <sip:[email protected]>;tag=47251617
6
CSeq: 2998 NOTIFY
Server: ATA-740S/1.1.3
Contact: <sip:192.168.3.101:25320>

Content-Length: 0
Date: Fri, 23 Jun 2006 16:00:49 GMT
7 Event: dialog
Expires: 0
Max-Forwards: 70
Accept: application/dialog-info+xml
9 INVITE sip:[email protected] SIP/2.0

Content-Length: 286
[49/129]
Via: SIP/2.0/UDP 192.168.3.61:5060;rport;branch=z9hG4bKb815e3f
From: <sip:[email protected]>;tag=dc8cec1
Supported: replaces,timer
CSeq: 40372 INVITE
Max-Forwards: 70
Session-Expires: 1800
Replaces: [email protected];
to-tag=489c13a17;from-tag=46f66d0e;early-only
Require: replaces
v=0
o=iprDC3000361 11173553 11173553 IN IP4 192.168.3.61
s=-
c=IN IP4 192.168.3.61
t=0 0
m=audio 8000 RTP/AVP 0 8 18 4 101
a=ptime:20
a=fmtp:101 0-16
SIP/2.0 200 OK
Server: YV2/1.0
Content-Length: 245
Via: SIP/2.0/UDP 192.168.3.61:5060;branch=z9hG4bKb815e3f
To: <sip:[email protected]>;tag=50e14646
Supported: replaces,timer
From: <sip:[email protected]>;tag=dc8cec1
CSeq: 40372 INVITE
Session-Expires: 1800;refresher=uas
9 Contact: <sip:[email protected]:5060>
v=0
o=iprD0E9015189 4553700 4553700 IN IP4 192.168.3.57
s=-
c=IN IP4 192.168.3.57
t=0 0
m=audio 8002 RTP/AVP 0 8 18 4 101
a=fmtp:101 0-16
a=ptime:20
11 CANCEL sip:[email protected] SIP/2.0
Content-Length: 223
[50/129]
CSeq: 66526 INVITE

User-Agent: YV2/1.0
Max-Forwards: 70
 Server Feature Access Codes.

Configure various feature access codes to activate server features, such as call return, voice mailbox, etc. On
making outbound calls, those feature access codes are sequentially checked against. If the dial string matches
any of the feature access code, the dial string will be dialed as “is” to facilitate server feature access. For
example, if you configure Server Feature Access Code-0 to “*69”, then whenever you dial “*69”, YV2 will
dial it as sip:*[email protected] to server instead of translating it as LAN dials.
Besides, you may configure a prefix match by append a ‘v’ suffix to those feature access codes. For example,
if you configure “*#v”, then when you dial “*#1234”, the resultant dial string will be rendered as
“sip:*#[email protected]”.
Note: such server features replies on the SIP proxy server support.
5.5.4 Dial Plan

. r 4

The programmable dial plan is designed for the service provider to customize the behavior of YV2 for
collecting and sending dialed digits. The dial plan allows the terminal user to specify the events that trigger the
sending of dialed digits. These events include the following:
 The termination character has been entered.
 The specified dial string pattern has been accumulated.
 The specified number of dialed digits has been accumulated.
Dial Plan consists of alphanumeric string, and the maximum number of characters is 255.
System default dial plan is empty.
For example, "911 | [1_9]>#.r5" denotes that:
 911 => dial out immediately
 [1_9]>#.r5
 [1_9]: Any digit from 1 to 9.
 (>#) press '#' to end dialing at any time.
 (.r5): 6 digits from [0-9,*#]
It will dial out immediately whenever a dial string starts with any digit form 1 to 9, and at most 6 digits more
has been collected (7 digits in total) or user press '#' to end dial.
Please refer to Appendix A – Dial Plan for syntax.
5.5.5 Hot Line

2 √ L A N D i a l
4 . D i a l P l a n
[51/129]
5 . H o t L i n e
On/Off AoR Back
Specify the number to dial out whenever the user hooks off.
For example, if you enable this feature, and fill in the number: “sip:[email protected]” or “[email protected]” or
“888”, then the terminal will dial to the specified number whenever user turns on the speaker phone or hooks off
without waiting for user input.
 [On]/[Off]: Toggle between enable and disable this feature, where [On] and [Off] indicates current
 [Back]: Return
 [AoR]: Specify the number to dial out whenever the user hooks off. For example, if you enable this
feature, and fill in the number: “sip:[email protected]” or “[email protected]” or “888”, then the terminal will
dial to the specified number whenever user turns on the speaker phone or hooks off without waiting for
user input.
A d d r e s s - o f - R e c o r d :

o [Del]: Delete one character.
o [Abc..]/[123..]: Toggle between digits and alphanumeric input, where. [Abc..] indicates current
o [Clear]: Clear all input.
o [Back]: Return without any changes.
o 【↑】and【↓】: move cursor per line in edit mode.
5.6 Message Alert

5 . D i a l P l a n
On/Off Back
Configure whether the phone should play alerting tone on receiving and on sending messages. The alerting
tone serves to remind the user that new messages have been received or messages have been sent successfully
(failed).
Default is enabled.
 [Back]: Return
[52/129]
5.7 Auto-Answer
1 . A u t o - A n s w e r A l l
2 . C o n t r o l L i s t
On/Off Back
By “auto-answer”, the phone-set will play a distinguished auditable tone shortly (unless “Silent” is specified),
then answer the call by turning on speaker-phone (or headset if applicable) during idle period. By idle, it either
takes no call or is putting a call on hold. Otherwise, the incoming call will proceed as normal incoming calls (play
call-waiting tone to remind user of a call is waiting for answer).
 [Back]: Return
 【OK】: Edit control list if selected otherwise return.
Once “Auto-Answer All” is enabled, it will auto-answer all incoming calls on idle mode. By idle, it means
the phone-set is either engaging in no calls or putting a call on hold. It will play distinguished auditable ring
then turn on speaker-phone (or head-set if applicable) whenever an incoming call arrives.
Default is disabled
Note 1: if this featured is mapped as a programmable DSS function key, pressing the mapped DSS key will
toggle the auto-answer mode. Besides, unmap of this DSS key will NOT turn off auto-answering.
Note 2: if 【DND】is on, Do-Not-Disturb feature wins; otherwise, for All-Calls-Forward and Auto-Answer
features, All-Calls-Forward takes precedence.
Note 3, YV2 supports server-side invoked auto-answer feature, please refer to “Auto Answer” on “YV2
User’s Guide” for detail.
Note 4, YV2 supports auto-answer based on called number as well. That is, if you registered several SIP
address-of-records, you could configure to auto-answer calls arrived on specific SIP AoR(s). Please refer to
“7.SIP Settings” / “Service Domain” / “Auto-Answer” for how to auto-answer incoming calls arrived on specific
accounts.
Note 5, On idle, all incoming calls will check against the server-side invoked auto-answer feature first, and
then the global auto-answer feature, finally fall to the account-specific auto-answer feature.
5.7.1 Auto-Answer Control List

1 . E n t r y 1
2 . E n t r y 2
3 . E n t r y 3
4 . E n t r y 4
5 . E n t r y 5
Level AoR Back
User controlled auto-answer list. Whenever an incoming call arrives, this terminal will check the SIP address-
of-record specified in the <From> header in the received INVITE message against user-defined auto-answer
control list. If a match exists, this incoming call will be auto-answered based on their configured priority.
 [Level]: Modify anto-answer level.

 [AoR]: Set address-of-record.
 [Del]: Remove this entry.
 [Back]: Return
[53/129]
5.7.1.1Level
1 . N o r m a l
2 . U r g e n t
3 . I m p e r i o u s
4 . S i l e n t
Back
Auto-Answer Priority:
 Normal
 Respond a “486 Busy Here” if all lines are occupied.
 Otherwise, auto-answer this incoming call.
 Urgent
 Dropped an inactive call if all lines are occupied.
 Put any on-going calls to hold and auto-switch to an available line if not idle.
 auto-answer the newly arrived call.
 Imperious
Identical to “Urgent” except that the phone-set will auto-answer this call even the do-not-disturb (DND) or all-calls-
forward feature has been activated.
Note, only products based on Agere VoIP1000/T8300 ARM platform support this “imperious” auto-answer level;
other products, including those based on AudioCode AC49x MIPS platform, will interpret “imperious” as “urgent”.
 silent
Identical to “Imperious” except that no distinguished auditable tone will ever be played. This is for
supervised monitor/announce feature.
Note, only products based on Agere VoIP1000/T8300 ARM platform support this “slent” auto-answer level; other
products, including those based on AudioCode AC49x MIPS platform, will interpret “silent” as “urgent”.
Incoming call processing rules (by precedence):

i. If “silent” or “Imperious” is specified, then auto-answer it. Besides, if “silent” is specified, then no
“distinguished auditable tone” will be played.
ii. If the phone is engaged in do-not-disturb mode, then DND wins
iii. Otherwise, if the phone has turned on unconditionally forward feature, then all incoming calls are
forwarded.
iv. Check for auto-answer feature:
a. Check for server-side auto-answer feature.
b. Check for the global switch for auto-answer feature.
c. Match this call against all enabled service domains for auto-answering feature, if a match is
located, auto-answer it.
v. Otherwise, proceed as normal incoming calls.
Note, only products based on Agere VoIP1000/T8300 ARM platform support both “imperious” and “silent” auto-
answer level; other products, including those based on AudioCode AC49x MIPS platform, will interpret both levels as
“urgent”.
5.7.1.2AoR
A d d r e s s - o f - R e c o r d :

Specify which caller (<From> header in the received INVITE SIP message) should be auto-answered.
[54/129]
5.8 Unhold on Transfer

5 . D i a l P l a n
On/Off Back
Specify whether to unhold the transferee (as well as the transfer target in the case of attended transfer) before
issuing REFER to finish transfer.
Default is unchecked (recommended).
 [Back]: Return
5.9 Conference Alert

5 . D i a l P l a n
On/Off Back
Specify whether to play alerting tone during an ad hoc 3-way conference.
Once enabled, terminal will regularly (every 60 seconds) play alerting tone during an ad hoc 3-way local
conference to all attendees to reminde them a conference is undergoing.
Default is enabled.
 [Back]: Return
5.10 Xfer on Exit Conf

[55/129]
5 . D i a l P l a n
On/Off Back
YV2 supports ad hoc 3-way local conference. After setting up a conference, your terminal will serve as a
media mixer such that the other two parties can hear each other. However, when you hang up, all parties in this
conference will be disconnected because there is no media mixer.
Under some circumstances, you may want to keep the other two parties connected even after you exit this
conference. By enabling this feature, this terminal will carry out an attended transfer on exiting ad hoc 3-way
local conference. As a result, the other two parties may keep conversation without your involvement.
 [Back]: Return
[56/129]
6 Network
1 . A c t i v e S t a t u s
2 . G e n e r a l
3 . R T P S e t t i n g s
4 . N A T & F i r e w a l l
5 . S N M P
6 . V L A N
Ping Back
 【OK】: Enter selected submenu.
 [Ping]: ping the remote host for its availability or to verify the network configuration of the phone. If no
replies are received within a 5-second timeout period, it will show “Host unreachable.”
P i n g
l a r r y . g l o o . n e t

Enter the remote host to ping by sending ICMP echo request packets. You could use either a dotted
IP or a DNS name.
o 【OK】: Start to ping.
P i n g
I C M P r e p l y f r o m l a r r y
. g l o o . n e t
Back
 【OK】: Return
 【↑】and【↓】: Scroll lines if there are too many lines to fit into one screen.
6.1 Active Status

I P : 1 9 2 . 1 6 8 . 3 . 2 0 0
M A C : 0 0 0 E C 3 1 2 3 4 5 6
D N S : 1 9 2 . 1 6 8 . 3 . 1
D N S : 1 9 2 . 1 6 8 . 3 . 2 5 4
L A N : 1 0 0 M , F u l l - D u p l e x
P C : : D o w n
Back
Display currently active host IP, Ethernet MAC address and the active DNS IP (secondary DNS will be
shown if available), LAN and PC Ethernet link status in order. The phone will set MAC as it host name in 12
hexadecimal characters.
 [Back]: Return.
 【↑】and【↓】: Scroll screen per line if there are too many lines to fit into one screen.
LAN and PC Ethernet link status are one of the following:

 Down: not connected
 Up: Ethernet jack connected
 100M, Full: 100 MB, Full-Duplex (Usually connects to a 100-MB Ether-Switch).
[57/129]
 100M, Half: 100 MB, Half-Duplex (Usually connects to a 100-MB Ethernet hub).
6.2 General
1 . M o d e
2 . S t a t i c S e t t i n g s
3 . P P P o E S e t t i n g s
4 . D N S S e r v e r
5 . H o s t N a m e
6 . T i m e - t o - L i v e
Back
 [Back]: Return.
6.2.1 Mode
1 . D H C P
2 . S t a t i c A s s i g n
3 . P P P o E
Back
Choose the protocol to configure host IP, gateway, network mask and / or domain name server.
Default is “DHCP”.
 [Back]: Return.
 【↑】and【↓】: Scroll menu items.
Supported DHCP options are:

 Client PC address
 DHCP option 1—Client Subnet Mask
 DHCP option 2—Date/Time offset
 DHCP option 3—Gateway IP on the client’s subnet
 DHCP option 6—One or two Domain Name servers
 DHCP option 15—Domain name
 DHCP option 42—Network Time Protocol servers
 DHCP option 66 (TFTP server name).
 DHCP option 120 (SIP Servers as per RFC3361)
Note: To make DHCP option 6 take effect, you must disable 【 MENU 】 =>”6.Network” / ”General” /
“DNS Server” / “Static DNS” by picking “DHCP”.
Note: If DHCP option 2 is present, it will overwrite the time-zone setting in menu-“Time-zone”
Note: If DHCP option 42 is present, it will overwrite the SNTP server in menu-“8.Service” / “SNTP” /
“Server IP”
Note: DHCP option 66 will overwrite the Auto-provision server in “8.Service” / “Auto-Provision” /
“TFTP server”.
Note: DHCP option 120 is not present in initial DHCP OFFER response, it will send a second
INFORM request for DHCP option code 120 to automatically discover SIP servers (at most 5
servers).
Note: Under DHCP mode, whenever the LAN connection is broke (such as you accidentally unplug the
LAN Ethernet jack), YV2 will retry to acquire IP from DHCP server when the LAN port is
connected again.
[58/129]
6.2.2 Static Settings

1 . H o s t I P
2 . N e t w o r k M a s k
3 . G a t e w a y I P
Back
Static assign host IP, network mask and gateway IP.
 [Back]: Return.
H o s t I P :
1 9 2 . 1 6 8 . 0 0 3 . 2 0 0
Back
Note: You should enter "0.0.0.0" to disable gateway routing. That is, if you reside on a LAN without
gateway, you must set this to "0.0.0.0" rather than a non-existent IP; otherwise the network packets
may not be routed correctly (which may result in no voice packets could be sent from this phone)!
This constrain applies to DHCP and PPPoE as well: DHCP and PPPoE server should not designate a
non-existent or invalid gateway.
Default gateway IP is “000.000.000.000”
6.2.3 PPPoE Settings

1 . U s e r N a m e
2 . P a s s w o r d
3 . S e r v i c e N a m e
Back
Point-to-Point Protocol over Ethernet (PPPoE) for xDSL dialup.
 [Back]: Return.
6.2.3.1User Name
L o g i n U s e r N a m e :

Name used by the phone for PPPoE authentication during dialup.

6.2.3.2Password
P a s s w o r d :
1 2 3 4
[59/129]

Password used by the phone for PPPoE authentication during dialup.
Initially show 10 ‘*’ signs but show clear text on user input because alphanumeric characters may be needed.

6.2.3.3Service Name
S e r v i c e N a m e :

Some ISP require the use of service name.

6.2.4 DNS Server

1 √ S t a t i c D N S
2 . P r i m a r y D N S
3 . S e c o n d a r y D N S
On/Off Back
Configure to obtain domain name server by DHCP, or use the one statically assigned. The configured DNS
server will be activated if “Static DNS” is enabled.
Default is disable to utilize the DNS servers acquired by DHCP.
 [Back]: Return.
D N S S e r v e r :
1 9 2 . 1 6 8 . 0 0 3 . 2 5 4
Back
Statically assign the primary domain name server IP.
Please enter "000.000.000.000" to disable this setting.
Generally, those IP(s) are invalid:
1. xxx.xxx.xxx.000 (this is LAN ID if network mask is 255.255.255.0)
2. xxx.xxx.xxx.255 (this is LAN broadcast IP if network mask is 255.255.255.0)
3. 255.255.255.255 (Broadcast IP)
4. 127.0.0.1 (localhost IP)
[60/129]

6.2.5 Host Name

H o s t N a m e :
c p e 1

Set host name of this terminal which will be used on DHCP request for better network administration as in
6.2.6 Time-to-Live
I P P a c k e t T T L
[ 7 5 - 2 5 5 ]
1 4 4
Del Back
Configure the TTL value of IP packet. The default value is 144. For RTP packets the TTL will use a value
higher than 127, and a value higher or equal to 75 for SIP signaling.

 【OK】: Save changes and return
[61/129]
6.3 RTP Settings

1 . P o r t B a s e : 8 0 0 0
2 . P o r t R a n g e : 6
3 . R T P I P T o S : 0 x B 8
4 . R T C P I P T o S : 0 x 7 4
5 . S y m m e t r i c F l o w
Back
 [Back]: Return
6.3.1 Port Base

R T P P o r t B a s e
[ 2 - 6 5 5 3 4 ]
8 0 0 0
Del Back
Configure the starting Real-time Transport Protocol (RTP/RTCP) port for media transmission.
Conventionally, the RTP will use an even UDP port, and the RTCP will use a UDP port one higher than that of
RTP; such that the port base should be an even number and larger than 3000 to avoid conflict. We suggest a base
higher than 3000; however, it accepts a value between 2 and 65534 to accommodate single line configuration.
Default is 8000.

6.3.2 Port Range

R T P P o r t R a n g e
[ 2 - 1 0 0 ]
1 6
Del Back
Configure the number of ports which RTP will be used for media transport. It is an offset from the pre-
configured base port. Each channel takes 2 ports, one for RTP and the other for RTCP session. Conventionally,
the RTP will use an even UDP port, and the RTCP will use a UDP port one higher than that of RTP. As a result,
the range specified here should be an even number and at least doubles the lines equipped on the phone. For
example, if the phone has two lines, then the least port range requirement is 4. We suggest that a value that triples
the lines should be more suitable as a minimum requirement; however, it accepts a value between 2 and 100 to
accommodate single line configuration.
Default is 16.

6.3.3 PTP IP ToS

I P T y p e - o f - S e r v i c e
[ 0 - 2 5 5 ]
1 8 4
Del Back
[62/129]

Specify the IP type-of-service (ToS) bits of RTP packets transmitting voice streams.
The IP ToS bits consist eight bits:
7 (MSB) 6 5 4 3 2 1 0(LSB)
Precedence Delay Throughput Reliability Reserved (00)
Where MSB denotes “Most Significant Bit” and LSB denotes “Least Significant Bit”.
 Bits 5-7 specify the precedence:
 111 - Network Control
 110 - Internetwork Control
 101 - CRITIC/ECP (RTP default)
 100 - Flash Override
 011 - Flash (RTCP default)
 010 – Immediate
 001 – Priority
 000 - Routine (normal)
 Bit 4: 0 = Normal Delay, 1 = Low Delay (RTP/RTCP default).
 Bit 3: 0 = Normal Throughput, 1 = High Throughput (RTP default).
 Bit 2: 0 = Normal Reliability, 1 = High Reliability (RTCP default).
 Bit 0-1: Reserved for Future Use.
System default is 0xb8 (184 in decimal) for RTP and 0x74 (116 in decimal) for RTCP packets. Please enter it
in decimal.
6.3.4 Symmetric Flow

1 . P o r t B a s e : 8 0 0 0
2 . P o r t R a n g e : 6
3 . R T P I P T o S : 0 x B 8
4 . R T C P I P T o S : 0 x 7 4
5 . S y m m e t r i c F l o w
On/Off Back
 [Back]: Return
 【OK】: Return
YV2 supports symmetric RTP flow for the cases where only one endpoint is behind a NAT, and RTP packet
flow will be possible in at least one direction. Since a client behind a NAT can usually successfully send RTP
packets to another client in the public Internet, in a symmetric mode, RTP sent in the other direction could be sent
to the address and port that RTP was received from. As a result, even if you are behind a NAT and choose “Full
Access” as your way to traverse NAT, you could still reach another YV2 on the public internet (since the peer
would adjust RTP flow accordingly).
Generally, if your terminal is behind a symmetric NAT, which cannot be traversed by STUN, or your
terminal is NATed, you should activate this feature. And the other party on the WAN (or capable of
traversing the NAT it currently behind without problem) should NOT enable this feature.
Once “Symmetric RTP flow” is enabled, YV2 will send its INVITE message stipulating its desire for
“Symmetric RTP flow” in SDP during call setup phase if the “UDP traversal” mode configured in 『 Main
Menu』=>”6.Network” / ”NAT & firewall” / ”UDP Traversal” is either “Full access” or “Enable STUN”. That is
if you have configured “Static NAT map” as your way to traverse NAT, it will not request the peer to enable
[63/129]
symmetric RTP flow). The SDP sent by this UAC would be:
o=SIP-Phone 28908445311 28908445311 IN IP4 192.168.3.51
s=-
t=0 0
c=IN IP4 192.168.3.51
m=audio 35000 RTP/AVP 4 18 0 8 101
a=direction:active IN IP4
If the UAS supports this extension (which YV2 does), it will wait for RTP packets to be received from the
client behind the NAT before sending, the answer SDP will be:
v=0
o=SIP-Phone 28908445405 28908445405 IN IP4 192.168.3.101
s=-
t=0 0
c=IN IP4 192.168.3.101
m=audio 35000 RTP/AVP 18
a=direction:passive IN IP4
Note: for backward compatibility with other SIP phones, system default is to disable “Symmetric RTP
flow”, thus the “a=direction” attribute will not appear in the exchanged SDP.
To make symmetric RTP flow works, the UAS (which sends passive RTP flow) must locate on the WAN.
For YV2, it must comply with one of the following conditions:
 It resides on the public Internet and has public IP (Full-Access).
 It has configured to employ “2.Static NAT map” as its way to traverse its NAT from 『 Main
Menu』=>”6.Network” / ”NAT & firewall” / ”UDP traversal”, and you have configured your NAT to
statically map the corresponding UDP ports (both SIP signaling and RTP ports) to your phone.
 It is behind a NAT which can be traverse by STUN and you have configured your YV2 to employ
STUN to traverse you NAT.
 It is behind a UPnP-capable NAT and you have configured your terminal to employ UPnP to
dynamically MAP your UPnP-aware NAT.
6.4 Network & Firewall

1 . S T U N
2 . U D P T r a v e r s a l
3 . S t a t i c N A T I P
4 . D y n a m i c N A T I P
5 . D i a g n o s e U P n P
Back
 [Back]: Return
To properly configure your NAT traversal scheme, refer to “NAT Traversal” on “YV2 User’s Guide”.
6.4.1 STUN
1 . S T U N S e r v e r
2 . D i a g n o s e N A T
Back
Simple Traversal of UDP through NATs (STUN) is a lightweight protocol that allows applications to
discover the presence and types of Network Address Translators (NATs) and firewalls between them and the
public Internet. It also provides the ability for applications to determine the public IP addresses allocated to them
[64/129]
by the NAT.
 [Back]: Return
6.4.1.1STUN Server
S T U N S e r v e r
l a r r y . g l o o . n e t

Configure the Simple Traversal of UDP through NAT protocol server. You may use a dotted IP or a DNS
name. If you fill in the DNS name, such as “isp.com”, YV2 will try to locate the STUN server of “isp.com”
domain via DNS SRV query (_stun._udp.isp.com). If the STUN SRV record query failed, YV2 will use the IP of
“isp.com” as the default STUN server with UDP port 3478. Therefore, if the DNS returns more than one STUN
SRV records, YV2 will switch to the secondary STUN server whenever it fails to reach the primary STUN server
(STUN server redundancy). Besides, YV2 will switch between STUN servers only if it fails to reach the primary
STUN server for more than 30 seconds to avoid instability in switching back and forth.
Default is “larry.gloo.net”.
To use a different STUN server port other than the default UDP port 3478, please specify it in the format like
“STUN.isp.com:8888” or “210.159.100.17:8888”.
6.4.1.2Diagnose NAT
N A T T y p e :
S y m m e t r i c N A T ( 2 2 2 . 6 8 .
8 7 . 1 3 5 )
Back
 [Back]: Return.
 【↑】and【↓】: move cursor per line if there are too many lines to fit into one screen.
YV2 will communicate with the STUN server (running outside the NAT, assigned in “6.Network” /
“Firewall & NAT” / “STUN Server”) to determine the type of NAT it is behind and the diagnosis may take a
quite while (roughly 10 seconds or so in a worse case). If the phone fails to contact a STUN server, it will report it
as “No STUN server is available”; otherwise, it will show the IP of the NAT / firewall.
 Public internet (Neither NAT nor Firewall is involved)
 Firewall blocks UDP (Definitely, you cannot communicate with parties outside of your LAN!).
Note: this may also be a false alarm if the specified STUN server is either down or unreachable.
 Symmetric UDP Firewall (STUN may work!)
Firewall that allows UDP out and responses have to come back to the source of the request (like a
symmetric NAT, but no translation. We call this a symmetric UDP Firewall)
[65/129]
 Full cone NAT (Enable STUN should suffice).

A full cone NAT is one where all requests from the same internal IP address and port are mapped
to the same external IP address and port. Furthermore, any external host can send a packet to the
internal host, by sending a packet to the mapped external address.
 Restricted cone NAT (STUN may work!)
A restricted cone NAT is one where all requests from the same internal IP address and port are
mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP
address X) can send a packet to the internal host only if the internal host had previously sent a
packet to IP address X.
 Port restricted cone NAT (STUN may work!)
A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers.
Specifically, an external host can send a packet, with source IP address X and source port P, to the
internal host only if the internal host had previously sent a packet to IP address X and port P.
 Symmetric NAT
A symmetric NAT is one where all requests from the same internal IP address and port, to a
specific destination IP address and port, are mapped to the same external IP address and port. If the
same host sends a packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a packet can send a
UDP packet back to the internal host.
 Non full cone NAT
If failed to identify the NAT type, it will show “Non Full cone NAT” (such as the one bundled in
Win2000).
 If the diagnosis is success, it will show the mapping of SIP signaling port on NAT / firewall as well
 If failed to identify the NAT type, it will show “Non Full cone NAT” (such as the one bundled in
Win2000).
The red font indicates that the phone has problem traversing the NAT / Firewall of that type, and the user should
consider manually configure the port mappings on NAT / Firewall.
6.4.2 UDP Traversal

1 . F u l l A c c e s
2 . S t a t i c N A T M a p
3 . E n a b l e S T U N
4 . E n a b l e U P n P
Back
 [Back]: Return.
Choose “Full access” if the phone is not behind a NAT / Firewall or you are behind a SIP-aware NAT.
Otherwise, if you are behind a NAT / Firewall, choose “Static NAT map” whenever possible. Besides, if the
NAT / Firewall is either a Symmetric NAT or a Firewall that blocks UDP you must choose “Static NAT map” as
well. If the NAT/Firewall/Gateway/Router in your LAN is UPnP-aware (Universal Plug and Play), you may turn
on UPnP to dynamically open and close TCP/UDP ports on these devices for SIP signaling and RTP media
streaming; otherwise choose STUN, Simple Traversal of UDP through NAT, as your last resort.
Generally, if your IP falls within the ranges: 10.0.0.0 to 10.255.255.255, 172.16.0.0 to 172.31.255.255, and
192.168.0.0 to 192.168.255.255, then you may have trouble making and taking calls from the public WAN.
Default is Full access.
Note: YV2 supports symmetric RTP flow for the cases where only one endpoint is behind a NAT, and RTP
packet flow will be possible in at least one direction. Since a client behind a NAT can usually successfully send
RTP packets to another client in the public Internet, in a symmetric mode, RTP sent in the other direction could
be sent to the address and port that RTP was received from. As a result, even if you are behind a NAT and choose
“1.Full Access” as your way to traverse NAT, you could still reach another YV2 on the public internet (since the
peer would adjust RTP flow accordingly). Generally, if your terminal is behind a symmetric NAT, which cannot
[66/129]
be traversed by STUN, or your terminal is NATed, you should activate this feature. And the other party on the
WAN (or capable of traversing the NAT it currently behind without problem) should NOT enable this feature.
6.4.3 Static NAT IP

S t a t i c N A T I P
1 4 0 . 1 1 3 . 2 3 . 3

Assign the static NAT IP to use by SIP signaling and RTP media stream. You could use either a dotted IP or a
DNS name. This field is valid if and only if you choose “Static NAT map” from the “UDP traversal”
configuration. You may detect your NAT IP by STUN from menu-“Diagnose NAT” (please fill in only the IP,
excluding the diagnosed SIP port).
Note: you should also map the SIP signaling port, RTP port base and RTP port range used for media stream
from NAT to the phone to enable UDP traverse through NAT / Firewall (if there are several phones reside under
the same NAT, their SIP and RTP ports must not be overlapped since they all share the same NAT resource).
Note: if your NAT uses dial-up modem or connects to the WAN via ADSL by PPPoE, then the IP of your NAT
may change whenever it re-dials up. Please synchronize it if the NAT does not come with a fixed IP. Besides, you
may detect your NAT IP by STUN from menu- “Diagnose NAT” and re-configure the new NAT IP into
menu-“Static NAT IP”.

6.4.4 Dynamic NAT IP

1 . N o t i f y o n I P C h a n g e
2 . A u t o - U p d a t e b y S T U N
Back
If your NAT uses dial-up modem or connect to the WAN via ADSL by PPPoE, then the IP of your NAT may
change whenever it re-dials up. If you choose “1.Notify on IP Change”, the phone will notify you by sending an
instant message into your inbox whenever it detects a mismatch between the NAT you configured into
menu-“Static NAT IP” and the one it auto-discovered by STUN. On the contrary, if you choose “2.Auto-Update
by STUN”, it would auto-refresh the NAT IP into menu-“Static NAT IP” whenever it detects an inconsistency.
Note: either option requires a viable STUN server has been configured into menu-“STUN server”.
Default is “1.Notify on IP Change” and the detection interval is 20 minutes.
 [Back]: Return.
6.4.5 Diagnose UPnP

U P n P :
Back
Detect the external IP of UPnP-capable NAT/Gateway/Firewall devices (support either WANIPConnection
and WANPPPConnection). It will search for UPnP devices and show its external IP. If such devices existed, this
terminal will open UDP/TCP mapping on NAT device for media streaming and SIP signaling by UPnP as
necessary; Otherwise if there is no such devices co-exist (or turn off UPnP explicitly), it may show “Not found
[67/129]
UPnP device”. Consult to your LAN administrator for further information.

Note: the UPnP mode must have been enabled before you can diagnose UPnP device on your LAN.
 [Back]: Return.
 【↑】and【↓】: move cursor per line if there are too many lines to fit into one screen
6.5 SNMP
1 . R e a d C o m m u n i t y
2 . W r i t e C o m m u n i t y
3 . T r a p C o m m u n i t y
4 . T r a p M a n a g e r
Back
YV2 supports Simple Network Management Protocol (SNMP) v2 and Management Information Base (MIB)
v2 for SNMP Get/Set/Trap operation. It supports the following (RFC1213)
i. “system” group: “sysObjectID” represents its model.
ii. “interface” group: Only Ethernet ports are included in the table. “ifAdminStatus” is effective.
For LAN port, “ifIndex is 1, and 255 for Ethernet WAN port.
iii. “IP” group
iv. “ICMP” group
v. “TCP” group
vi. “UDP” group
vii. “SNMP” group.
Besides, it supports proprietary enterprise IAD MIB to Get /Set current configuration by SNMP as well,
including:
i. Hardware type & version, software version
ii. MAC address of WAN/LAN
iii. Number of port (phone/Ethernet), Call-Channel 【A / B】status
iv. Real-time parameters, including gateway, DNS, SNMP, RTP, CODEC, etc.
v. SIP settings, Registrar server, and Proxy server.
vi. Image upgrade and restart.
 [Back]: Return.
1. Read Community: Set the “read community” for SNMP Get operation.
Default is “public” (case-sensitive).
2. Write Community: Set the “write community” for SNMP Set operation.
Default is “private” (case-sensitive).
Note: Values set by SNMP Set Operation will not take effect immediately. In order to facilitate
multiple values set in a succession, YV2 will not apply new values set by SNMP ‘till it returns
to idle mode for more than 5 seconds. Besides, a second new value application will not execute
if the previous one was done within 5 seconds ago.
3. Trap Community: Specify the community name used by the SNMP manager to verify traps.
Default is “public” (case-sensitive).
4. Trap Manager: Specify the SNMP manager to which the SNMP trap sends.
You could use either a dotted IP or a DNS name to specify the SNMP agent. YV2 will issue a trap to
SNMP manager when:
i. System startup
ii. System restart (by user/SNMP command/image upgrade)
[68/129]
iii. Call-Channel 【A / B】 Disabled

iv. Registrar Server availability.
Note: YV2 is capable of reporting traps to two different SNMP managers. The second trap manager
(not a backup manager) could only be set via auto-provision, or by SNMP browser to set it. If both
trap managers are available, YV2 will report each trap to both of them.
R E a d C o m m u n i t y
p U b l i c

6.6 VLAN
1 . A c t i v a t i o n
2 . V L A N I D : 4 0 9 4
3 . V L A N C o S : 6
On/Off Back
YV2 supports IEEE 802.1Q/802.1P/802.1D VLAN and IP class of service (differentiated service in IP ToS
bits). VLAN support will insert a 4-byte tag immediately after source address field in the 802.3 Ethernet frame.
Destination VLAN LAN
Source MAC
MAC tag Type
Address Data FCS
Address (4 bytes) 0x0800
(6-byte)
(6-byte) (2-byte)
.
VLAN Protocol ID:0x8100 (16 bits)Priority (3 bits)CFIVLAN ID (12 bits)
Once IEEE 802.1Q VLAN tagging is enabled, it will also map IP differentiated service set in ToS bits to
user priority field.
 [Back]: Return
6.6.1 VLAN ID
V L A N I D
[ 1 - 4 0 9 4 ]
1
Del Back
IEEE 802.1Q VLAN ID is 12-bit.
The following VLAN ID values are reserved:
VLAN ID Meaning / Usage
The null VLAN ID, indicating that the tag header contains only user_priority information, no
0 VLAN identifier is present in the frame. This VID value shall not be configured as a PVID,
configured in any Filtering Database entry, or used in any Management operation.
The default PVID value used for classifying frames on ingress through a Bridge Port. The
1
PVID value can be changed by management on a per-Port basis
[69/129]
Reserved for implementation use. This VID value shall not be configured as a PVID,
0xFFF configured in any Filtering Database entry, used in any Management operation, or
transmitted in a tag header.
If you disable VLAN tagging (disable menu-Activation), the VLAN ID will not take effect and be displayed
as “N/A” (not applicable).

6.6.2 VLAN CoS

V L A N C l a s s - o f - S e r v i c e
[ 0 - 7 ]
6
Del Back
IEEE 802.1P VLAN User priority field, default is 6; where 0 is normal and 7 is the highest priority.

[70/129]
7 SIP Settings
1 . S I P S e r v i c e
2 . S e r v i c e D o m a i n
3 . E N U M & E . 1 6 4
Back
 [Back]: Return
7.1 SIP Service

1 . T r a n s p o r t : 3
2 . S I P P o r t : 5 0 6 0
3 . E x p i r e s : 1 8 0 0
4 √ r p o r t
5 . H o l d b y R F C 3 2 6 1
6 . S i g n a l I P T o S : 0 x B 4
7 . S e s s i o n T i m e r : 1 8 0 0
8 . K e e p A l i v e T i m e r
9 √ E x p o s e S W V e r
1 0 . S I P T i m e r T 1
Back
Configure the SIP stack on the phone, including signaling port, transport protocol (TCP or UDP) to use and
the default registration expiration timeout (measured in seconds).
 [Back]: Return
7.1.1 Transport
1 . U D P
2 . T C P
3 . U D P a n d T C P
Back
By default, YV2 supports both UDP and TCP for SIP signalling, and this setting prioritize transportation
protocols on making requests. UDP bears a higher priority. If you specify TCP only, then YV2 will try to make
requests to the peers/servers by TCP, and fall back to UDP if the peers/servers do not support TCP.
The default is UDP and TCP. Currently there is no TLS support.
 [Back]: Return.
7.1.2 SIP Port

S I P L i s t e n P o r t
[ 1 - 6 5 5 3 5 ]
5 0 6 0
Del Back
Specify both the TCP and UDP port for SIP signaling. Default is 5060.
Note, to facilitate “Contact Dialing”, “IP Dialing” and “LAN Dialing” (please refer to “Dial Scheme” on “ YV2
User’s Guide”), where most users forget to dial the SIP signaling port of the peer, and end in no responses if the
peer doesn’t listen on the standard UDP port 5060 for SIP signaling, YV2 will always listens on UDP-5060 for SIP
signaling in addition to the user configured SIP service port. That is, if you configure it to listen on port 8888, it
[71/129]
will listen on three ports (TCP-8888, UDP-8888 and UDP-5060). However, if UDP-5060 is overlapped with RTP
ports for media session (see below menu-RTP settings); it will not listen on UDP-5060 for SIP signaling to avoid
conflict.

7.1.3 Expires
S I P R e g i s t e r E x p i r e s :
[ 2 0 - 8 6 4 0 0 ] ( S e c o n d )
3 6 0 0
Del Back
Set the default expiration time of registration; measured in seconds. We suggest a value not less than 1200
seconds to avoid heavy network traffic due to registration. This expiration may modified (shorten) by SIP
registrar server, and YV2 will re-register to server when 50% of the modified expiration time has been passed. It,
however, will re-register to SIP registrar after 90 seconds on registration failure.
Default is 3600 seconds.

7.1.4 rport
1 . T r a n s p o r t : 3
2 . S I P P o r t : 5 0 6 0
3 . E x p i r e s : 1 8 0 0
4 √ r p o r t
5 . H o l d b y R F C 3 2 6 1
6 . S i g n a l I P T o S : 0 x B 4
7 . S e s s i o n T i m e r : 1 8 0 0
1 0 . S I P T i m e r T 1
On/Off Back
Enable the support of RFC3581, an Extension to the Session Initiation Protocol (SIP) for Symmetric
Response Routing.
When SIP operates on UDP, responses to requests are returned to the source address the request came from,
and to the port written into the topmost Via header field value of the request. This behavior is not desirable in
many cases, most notably, when the client is behind a Network Address Translator (NAT). This extension defines
a new parameter for the Via header field, called "rport", that allows a client to request that the server send the
response back to the source IP address and port from which the request originated.
A client sends an INVITE to a proxy server which looks like, in part:
Via: SIP/2.0/UDP 10.1.1.1:4540;rport;branch=z9hG4bKkjshdyff
If the proxy does not recognize this extension, you SHOULD disable this feature such that YV2 will not
insert this “rport” parameter into Via header while sending SIP requests.
Default is enabled to help NAT traversal if your SIP proxy/soft-switch supports it (for backward
compatibility.you may disable it because some server think the rport parameter bears wrong format and
silently drop this reques).
[72/129]
 [Back]: Return
7.1.5 Hold by RFC3261

1 . T r a n s p o r t : 3
2 . S I P P o r t : 5 0 6 0
3 . E x p i r e s : 1 8 0 0
4 √ r p o r t
5 . H o l d b y R F C 3 2 6 1
6 . S i g n a l I P T o S : 0 x B 4
7 . S e s s i o n T i m e r : 1 8 0 0
1 0 . S I P T i m e r T 1
On/Off Back
By RFC2543, it uses “c=0.0.0.0” in SDP to perform a hold operation; but this has been depreciated in
RFC3261. Instead, it recommends the use of “a=sendonly” attribute in SDP to hold the peer.
Disable this feature to be backward compatible with RFC2543 implementation.
 [Back]: Return
7.1.6 Signal IP ToS

I P T y p e - o f - S e r v i c e
[ 0 - 2 5 5 ]
1 8 0
Del Back
Specify the IP type-of-service (ToS) bits of SIP packets transmitting signals. This setting affects both TCP
and UDP signaling packets.
The IP ToS bits consist eight bits:
7 (MSB) 6 5 4 3 2 1 0(LSB)
Precedence Delay Throughput Reliability Reserved (00)
Where MSB denotes “Most Significant Bit” and LSB denotes “Least Significant Bit”.
 Bits 5-7 specify the precedence:
 111 - Network Control
 110 - Internetwork Control
 101 - CRITIC/ECP (RTP default)
 100 - Flash Override
 011 - Flash (RTCP default)
 010 – Immediate
 001 – Priority
 000 - Routine (normal)
[73/129]
 Bit 4: 0 = Normal Delay, 1 = Low Delay (RTP/RTCP default).

 Bit 3: 0 = Normal Throughput, 1 = High Throughput (RTP default).
 Bit 2: 0 = Normal Reliability, 1 = High Reliability (RTCP default).
 Bit 0-1: Reserved for Future Use.
System default is 0xA8 (168 in decimal) for SIP signaling packets. Please enter it in decimal.
7.1.7 Session Timer

R F C 4 0 2 8 S e s s i o n T i m e r
[ 9 0 - 1 4 4 0 ] ( S e c o n d )
1 8 0 0
Del Back
Set the session refresh time, measured in seconds, as per RFC4028. Default is 1800 seconds. The minimum
interval is 90 seconds. On timer expiration, it will refresh the session either by re-INVITE or UPDATE the
session.

7.1.8 Keep Alive Timer

S I P N A T K e e p A l i v e
[ 0 - 8 6 4 0 0 ] ( S e c o n d s )
0
Del Back
The interval to send empty UDP packets to SIP registrar server to keep NAT port mapping open after
finishing registration. Set to 0 or a value smaller than Register Expire time effectively disable this feature. This
option sets only the frequency to send dummy UDP packets to registrar server to keep SIP signaling channel
through NAT; to turn on this feature for each service realm, go to “7. SIP Settings” / “Service Domain” / “n-th
Realm” / “Keep Alive”.

7.1.9 Expose SW Ver

1 . T r a n s p o r t : 3
2 . S I P P o r t : 5 0 6 0
3 . E x p i r e s : 1 8 0 0
4 √ r p o r t
5 . H o l d b y R F C 3 2 6 1
6 . S i g n a l I P T o S : 0 x B 4
7 . S e s s i o n T i m e r : 1 8 0 0
1 0 . S I P T i m e r T 1
On/Off Back
Specify whether it should include “User-Agent” or “Server” SIP header in each SIP message. These headers
consist of product name and version. Default is checked (expose).
[74/129]

 [Back]: Return
7.1.10SIP Timer T1
S I P T i m e r T 1
[ 2 0 0 - 4 0 0 0 ] ( m s )
5 0 0
Del Back
By RFC3261, T1 timer is an estimated measure of network packet round-trip time, and default to be 500-
ms. However, if the network link is extremely slow, such as in a satellite link, and its round trip time is more than
500-ms, you may increase T1 timer to reduce SIP messages retransmission and increase the transaction timeout.
However, change of T1 will affects 7 out of 11 SIP timers (see below), which are Timer-(A, B, E, F, G, H and J).
The foreseeable result of lengthen T1, such as change it from 500-ms to 1-second, will double Timer-(B, F, H and
J) as well, which in turn will make the transaction timeout more time-consuming. For example, by default, at
most 32 seconds a user will wait for transaction timeout (such as dial a non-existed #), now it takes as long as 64
seconds for such timeout.
And below summarizes the meaning and defaults of the various timers used by RFC3261.
Timer Value Meaning
T1 500ms default Round-Trip Time, RTT, Estimate
The maximum retransmit interval for non-INVITE requests
T2 4s
and INVITE responses
T4 5s Maximum duration of a message will remain in the network
Initially T1, and
Timer A INVITE request retransmit interval, for UDP only.
doubles for each retransmit
Timer B 64*T1 INVITE transaction timeout timer
Timer C > 3min Proxy INVITE transaction timeout
> 32s for UDP and
Timer D Wait time for response retransmits
0s for TCP/SCTP
Initially T1, and
Doubles for each retransmit
Timer E Non-INVITE request retransmit interval, UDP only.
‘till hitting T2, then T2 is used
instead.
Timer F 64*T1 Non-INVITE transaction timeout timer
Initially T1, and
Doubles for each retransmit
Timer G INVITE response retransmit interval
‘till hitting T2, then T2 is used
instead.
Timer H 64*T1 Wait time for ACK receipt
T4 for UDP and
Timer I Wait time for ACK retransmits
0s for TCP/SCTP
64*T1 for UDP and
Timer J Non-INVITE request retransmits
0s for TCP/SCTP
T4 for UDP and
Timer K Wait time for response retransmits
0s for TCP/SCTP
7.2 Service Domain

1 √ 1 s t D o m a i n
2 . 2 n d D o m a i n
3 . 3 r d D o m a i n
[75/129]
Back
 [Back]: Return
YV2 supports up to 3 different service domains simultaneously. The default service domain of each line-N
will be active service-domain-N; otherwise, it will use the 1st active service domain if not specified.
SER.iptel.org
ISP.net
WAN
foo.biz
WAN
Company
Company
LAN
LAN
sip:[email protected]
foo.com
7.2.1 N-th Domain

1 √ A c t i v a t i o n
2 . A u t h e n t i c a t i o n
3 . A d d r e s s - o f - R e c o r d
4 . P r o x y S e r v e r
5 . R e g i s t r a r
7 . K e e p N A T A l i v e
Back
 [Back]: Return
[76/129]
7.2.1.1Activation
5 .R e g i s t r a r
6 .A u t o - A n s w e r
7 .K e e p N A T A l i v e
On/Off Back
Configure to activate this service domain or not.
Default is disabled and it will not take effect (neither try to register nor show it as active domains) if any of
the following conditions is met:
a. Address-of-record is invalid or unavailable.
b. Domain part in address-of-record is missing.
c. No FQDN of registrar server is specified on manual mode.
d. Registrar server is configured as “Same as proxy” while proxy server is on manual mode but no FQDN
of proxy server is specified.
 [Back]: Return
7.2.1.2Authentication
1 . S e v e r L o g i n
2 . S e v e r P a s s w o r d
3 √ q o p S u p p o r t
Back
 [Back]: Return
7.2.1.2.1 Server Login

L o g i n U s e r I D :
M i c h a e l

Name used by the phone for authentication if a registration is challenged y the proxy server or registrar
server during initialization.

7.2.1.2.2 Server Password
[77/129]
L o g i n P a s s w o r d
1 2 3 4

Password used by the phone for authentication if a registration is challenged by the proxy server or registrar
server during initialization.
Note: the password will be shown on the LCD without encryption to facilitate the alphabetic input.
Besides, please remember to check for ending blanks and spaces; if you erroneously input the ending spaces,
the whole string (including ending spaces) will be treated as your password without trim.

7.2.1.2.3 qop Support

1 . I n c l u d e q o p
2 . I g n o r e q o p
Back
Some SIP servers does not support qop parameter during HTTP digest authentication (keep responding with
a “401 Unauthorized” on registration), then you should consider to ignore the qop parameter in the HTTP digest
authentication calculation. Note, however, in most cases, qop parameter should be included.
Once disabled, the qop parameter will be ignored and thus excluded in HTTP digest authentication
calculation.
Default is to include qop calculation.
 [Back]: Return.
7.2.1.3Address-of-Record
A d d r : 1 2 3 4 5 6 7 @ I S P . f o o .
Save Back
Specify the number or SIP URI to register, such as “Michael Wu <sip:[email protected]>”.
Note: the domain part will be recognized as the service domain associated with the registered SIP address of
record.
 [Save]: Save changes and return

 【↑】and【↓】: Navigate through menu itmes.
7.2.1.4Proxy Server
1 . M o d e : A u t o - L o c a t e
2 . D e f a u l t P r o x y F Q D N
[78/129]
3 . T r a n s p o r t : 1
4 . U D P P o r t : 5 0 6 0
5 . T C P P o r t : 5 0 6 0
Back
When default outbound proxy is used, all SIP requests are sent to this default outbound proxy server instead
of locating the next hop SIP proxy server by RFC 3263. All responses continue to reconcile the normal Via
processing rules. The media stream is not routed through the outbound proxy. Default is auto-locating.
YV2 implements SIP server redundancy when the DNS server returns more than one IP for the specified
domain name record. It will try the additional server IP when the first server IP has no response.
 [Back]: Return.
7.2.1.4.1 Mode
1 . A u t o - L o c a t e
2 . U s e D e f a u l t P r o x y
3 . R F C 3 3 6 1
Back
 [Back]: Return.
Configure to use default proxy server or not. If Auto-locating is enabled, it will acquire the SIP proxy server
responsible for the target URI by enquiring DNS NAPTR records of both “SIP+D2U” and “SIP+D2T”, then SRV
records of the target SIP domain. For example, if the target AoR is “someone <sip:[email protected]>”, then it
would query for “_sip._udp. dummy.com” or “_sip._tcp. dummy.com” (RFC 3263). By auto-locating SIP servers,
you could achieve SIP proxy redundancy on SRV records level. That is if the query for DNS SRV records returns
more than one SRV record, YV2 will use the SRV record of higher priority as primary SIP server, whereas the one
of lower priority as a backup server whenever it fails to reach the primary one. For example, you may configure
your DNS server to include the following SIP “Service Record” for UDP transport “ (_sip._udp); please change
those host and IP in red font accordingly:
$ORIGIN YourISP.com
;; Pref Weight Port Target
_sip_udp.YourISP.com. IN SRV 0 0 1000 proxy.sip.YourISP.com.
_sip._udp.YourISP.com IN SRV 10 0 1000 backup.sip.YourISP.com.
proxy.sip.YourISP.com IN A 192.168.192.99.
backup.sip.YourISP.com IN A 192.168.192.11.
This terminal supports DHCP option code 120 to automatically discover SIP servers as per RFC 3361. If
you enable DHCP, Dynamic Host Configuration Protocol, for network configuration, this terminal supports
RFC3361 for DHCP option code 120 in addition to acquire host IP, gateway, network mask, and DNS server.
Besides, if DHCP option code 120 is not included in the same DHCP offer for host IP/gateway/network mask, it
will send a second DHCP INFORM request for such information to automatically discover SIP server. The
purpose here is to ensure it works with existing DHCP servers which do not support RFC3361 for normal host
IP/gateway/netmask settings, but acquire DHCP option 120 for SIP server from another co-exist DHCP servers.
Note, once operate in this mode, both it will use the discovered SIP servers included in DHCP option code
120 (5 SIP servers at most) as both SIP registrar and proxy server, and treat those discovered SIP servers
listening on UDP port 5060. That is Registrar server will be same as proxy and the settings including Proxy
FQDN, transport, and UDP port of SIP outbound proxy server will be ignored for this service domain.
7.2.1.4.2 Default Proxy FQDN

D e f a u l t P r o x y F Q D N
s i p . I S P . f o o . n e t

Specify the IP address of the outbound proxy server. You can use either a dotted IP address or a DNS name.
If you specify a DNS name and the DNS server returns more than one IP for the specified hostname, YV2 will try
[79/129]
them in sequence if the first server IP contacted has no response (Proxy server redundancy).

When default outbound proxy is enabled, all SIP requests are sent to this default outbound proxy server
instead of locating the next hop SIP proxy server by RFC 3263. All responses continue to reconcile the normal
Via processing rules. The media stream is not routed through the outbound proxy.
NAT and outbound proxy modes can be independently enabled or disabled. The received= tag is added to
the Via header of all responses if there is no received= tag in the uppermost Via header and if the source IP
address is different from the IP address in the uppermost Via header. Responses are sent back to the source under
the following conditions:
i. If a received= tag is in the uppermost Via header, the response is sent back to the IP address contained in
the received= tag (and so is rport, as specified by RFC 3581 if you turn on the rport support from menu-
rport).
ii. If there is no received= tag and the IP address in the uppermost Via header is different from the source IP
address, the response is sent back to the source IP. Otherwise, the response is sent back to the IP address
in the uppermost “Via” header.
7.2.1.4.3 Transport
1 . U D P
2 . T C P
Back
The SIP transport protocol on which the default outbound proxy listens. Default is UDP. Currently there is
no TLS support.
 [Back]: Return.
7.2.1.4.4 UDP Port

P r o x y U D P P o r t :
[ 0 - 6 5 5 3 5 ]
5 0 6 0
Del Back
Specify the UDP port number on which the outbound proxy server listens. Default is 5060. Use 0 to disable
UDP signaling.

7.2.1.4.5 TCP Port

P r o x y T C P P o r t :
[ 0 - 6 5 5 3 5 ]
5 0 6 0
Del Back
Specify the TCP port number on which the outbound proxy server listens. Default is 5060. Use 0 to disable
[80/129]
TCP signaling.

7.2.1.5Registrar
1 . M o d e : S a m e a s P r o x y
2 . R e g i s t r a r F Q D N
3 . T r a n s p o r t : 1
4 . U D P P o r t : 5 0 6 0
5 . T C P P o r t : 5 0 6 0
Back
 [Back]: Return.
7.2.1.5.1 Mode
1 . S a m e a s P r o x y
2 . M a n u a l l y A s s i g n
3 . A u t o - D i s c o v e r
4 . M u l t i c a s t
Back
Configure how to locate the registrar server to register SIP AoR. Default is 『Same as Proxy』.
 [Back]: Return.
1. Same as Proxy: Choose 『Same as proxy』 if you want to register on default outbound proxy server. If
you do not configure default outbound proxy, then it will behave like Auto-discover (see below).
Note, if “RFC3361 DHCP Option for SIP Servers” is enabled for “SIP Proxy Server”, Registrar
Server will be adjusted to “Same as Proxy”, ignoring what has been configured for Registrar Server.
2. Manually Set: Assign a registrar server below.
3. Auto-discover by querying DNS NAPTR records of both “SIP+D2U” and “SIP+D2T”, SRV records of
SIP domain. For example, if the registered AoR is “someone <sip:[email protected]>”, then it would
query for NAPTR record of “_sip._udp. dummy.com” or “_sip._tcp. dummy.com” (RFC 3263). By
auto-discover registrar server, you could achieve Registrar server redundancy in SRV record level.
4. Multicast (224.0.1.75) Registration: Register by sending register information to multicast address
224.0.1.75
YV2 implements SIP registrar redundancy when the DNS server returns more than one IP for the specified
domain name record. It will try the additional server IP when the first server IP has no response.
7.2.1.5.2 Registrar FQDN

R e g i s t r a r S e r v e r F Q D N
s i p . I S P . f o o . n e t

Specify the IP address of the registrar server. You can use either a dotted IP address or a DNS name. If you
specify a DNS name and the DNS server returns more than one IP, YV2 will try them in sequence if the first
server IP contacted has no response (Registrar server redundancy).
[81/129]
7.2.1.5.3 Transport
1 . U D P
2 . T C P
Back
The SIP transport protocol on which the registrar server listens. Default is UDP. Currently there is no TLS
support.
 [Back]: Return.
7.2.1.5.4 UDP Port

R e g i s t r a r U D P P o r t
[ 0 - 6 5 5 3 5 ]
5 0 6 0
Del Back
Specify the UDP port number of the registrar server. Default is 5060. Use 0 to disable UDP signaling.

7.2.1.5.5 TCP Port

R e g i s t r a r T C P P o r t :
[ 0 - 6 5 5 3 5 ]
5 0 6 0
Del Back
Specify the TCP port number of the registrar server. Default is 5060. Use 0 to disable TCP signaling.

7.2.1.6Auto-Answer
On/Off Back
[82/129]
Auto-answer incoming calls arrive on this account (<FROM> header in the initial INVITE message) in idle
mode. By idle, it means the phone-set is either engaging in no calls or putting a call on hold.
If this feature is on, it will play distinguished auditable ring then turn on speaker-phone (or head-set if applicable)
whenever an incoming call arrives on this registered SIP address-of-record account during idle mode.
 [Back]: Return
Note 1, if [DND] is on, Do-Not-Disturb feature wins; otherwise, for All-Calls-Forward and Auto-Answer
features, All-Calls-Forward takes precedence.
Note 2, YV2 supports server-side invoked auto-answer feature (Section-“Auto Answer” on “YV2 User’s
Guide”) and auto-answer all incoming calls on idle mode system-wide (section-“Auto-Answer” on this
document).
Note 3, all incoming calls will check against the server-side invoked auto-answer feature first, and then the
global auto-answer feature, finally fall to the account-specific auto-answer feature.
7.2.1.7Keep NAT Alive

On/Off Back
Check to turn on the feature to regularly send empty UDP packets to SIP registrar server to keep NAT port
mapping between this terminal and the SIP registrar server. If the SIP registrar server does NOT treat constant
dummy UDP packets as denial-of-service, DOS, attack, and your terminal is behind NAT, you may turn on this
feature to keep SIP signal channel passing through NAT after finishing registration.
To set the frequency of sending dummy UDP packets, please go to “7. SIP Settings” / “SIP Service” / “Keep Alive
Timer”.
 [Back]: Return
7.3 ENUM & E.164

1 . E N U M D N S S u f f i x
2 . M i n E N U M L e n g t h
3 . U R I F o r m a t
4 . I n t ‘ l A c c e s s C
d e o
Back
Configure the appended DNS suffix for ENUM query and the minimum digit-length of a valid ENUM dial
string.
 [Back]: Return
[83/129]
7.3.1 ENUM DNS Suffix

E N U M D N S S u f f i x :
e 1 6 4 . a r p a

Set the ENUM suffix for E.164 query on DNS. Leaving it blank or setting it as “e164.arpa” to comply with
RFC2916 standard ENUM suffix, otherwise you may set it to a proprietary suffix, such as “e164.net”.
Default is “e164.arpa”.
E.g., if the dial string is "+886-3-1234567", the ENUM query string send to DNS server to resolve will be:
 Strip off all non-digits.
 Reverse the string.
 Insert a dot, '.', between each digit.
 Append the ENUM suffix, such as "e164.arpa".
 Send to DNS to resolve.
Thus, the resultant DNS query string would be:
"7.6.5.4.3.2.1.3.6.8.8.e164.arpa".

7.3.2 Min ENUM Length

M i n V a l i d E N U M L e n g t h
[ 1 - 1 6 ] ( D i g i t )
6
Del Back
Configure the minimum digit length of a valid ENUM dial string.
A valid ENUM dial string must starts with a '#', which will be translated to as a '+', and follows at least "Min
length" digits. If the rules does not apply, the dial string will be dial "as is" without any attempt to carry out an
ENUM resolution.
Default is 6 digits.

7.3.3 URI Format

1 . S I P U R I
2 . T E L U R I
Back
Configure what request URI should be sent whenever YV2 failed to resolve ENUM to SIP URI.
1. SIP URI: YV2 will send the request to the specified SIP service domain proxy in the following form:
INVITE sip: +PhoneNumber@domain:port;user=phone
For example, if you dial “+88631234567” and YV2 fails in ENUM resolution, it will send:
INVITE sip:[email protected];user=phone
[84/129]
2. TEL URI: YV2 will send request to the specified SIP service domain proxy in the form:
INVTTE tel: +PhoneNumber
For example, if you dial “+88631234567” and YV2 fails in ENUM resolution, it will send:
INVITE tel:+88631234567
 [Back]: Return.
7.3.4 Int’l Access Code

R e p l a c e t e l : ‘ + ‘ a s :
+

While making calls to PSTN, you may dial “+886-3-1234567”, and this setting configures whether the
leading ‘+’ should replace with an international access code, which is “00” in most countries, before actually
sending it to proxy server.
The default is leave the leading ‘+’ intact for proxy (or PSTN gateway) to replace it with the international
access code based on their geographic locations.
Thus, the leading ‘+’ will be replaced with the string specified here in either format, SIP URI or TEL URI, when
YV2 fails to resolve an ENUM number. The format could be specified in menu-“URI format”.

[85/129]
8 Service
1 . M W I
2 . M e s s a g e
3 . S N T P
4 . A u t o - P r o v i s i o n
Back
Configure the external services available to the phone.
 [Back]: Return
8.1 MWI
1 . V o i c e M a i l b o x A o R
2 √ U n s o l i c i t e d N O T I F Y
Back
 [Back]: Return
Voice mail allows you to access messages left by callers when you are unavailable to take their call. Voice
mail is an optional feature configured by your system administrator. Your particular phone setup might not
support accessing voice mail in this way.
To configure the voice mailbox number to dial when the message waiting indication button, 【MWI】, is
pressed, click the “Contacts” on the right panel to pick an entry from your address book; and delete the number
in input text area to remove it.
The corresponding MWI LED will be flashing whenever it receives NOTIFY messages stipulating some
new unread (new) voice messages in your voice mailbox. To access voice mail, press the 【MWI】 button and
follow the voice instructions.
YV2 supports messages waiting indication as per RFC3842. It accepts NOTIFY messages with or without
prior SUBSCRIBE (note, by http://pentest.tele-consulting.com/advisories/05_07_06_voip-phones.txt, there is a
weakness of unsolicited NOTIFY for MWI. You may turn off unsolicited MWI NOTIFY, thus effectively disable
“Out-of-Dialog MWI Notification”).
 MWI Subscription
If you have configured the voice mailbox link, YV2 will SUBSCRIBE to this link by sending the
SUBSCRIBE message to SIP server. Please refer to the following message flow (suppose your SIP AoR
is sip:[email protected] and the voice mailbox you configured is sip:[email protected]):
Via: SIP/2.0/UDP 192.168.3.50
From: John <sip:[email protected]>;tag= 17542c1
To: John <sip:[email protected]>
CSeq: 281 SUBSCRIBE
Contact: sip:[email protected]
Event: message-summary
Accept: application/simple-message-summary
Expires: 3600
User-Agent: SIP-Phone /1.1
Content-Length: 0
[86/129]
SIP/2.0 202 Accepted

Via: SIP/2.0/UDP 192.168.3.50
Record-Route: <sip:192.168.3.1;lr=1>
Contact: sip:192.168.3.1:6060
Expires: 3600
User-Agent: ABS GW v5.1.0
To: John <sip:[email protected]>;tag=980765
From: John <sip:[email protected]>;tag= 17542c1
CSeq: 281 SUBSCRIBE
Content-Length: 0
 Out-of-Dialog MWI Notification

YV2 accepts unsolicited MWI NOTIFIY messages as well. The sample NOTIFY message is as follows
(suppose “sip.isp.com” is one of your active service domains and your SIP AoR is
sip:[email protected]):
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/udp 192.168.0.1;branch=z9hG4bKcfb4
Content-Type: application/simple-message-summary
Contact: sip:192.168.0.1:6060
User-Agent: ISP Soft-Switch
Event: message-summary
Subscription-State: active
From: <sip:[email protected]>;tag=d8370cb
Call-ID: d07b59da8e
CSeq: 224493566 NOTIFY
Content-Length: 39
Max-Forwards: 70
Messages-Waiting: yes
Message-Account: sip: [email protected]
Voice-Message: 4/8 (1/2)
Note, when the MWI LED is off while pressing【MWI】, it will dial to the voice mailbox you configured.
However, when the MWI LED is flashing and you press 【MWI】, it will make a call to the Message-Account
stipulated on the latest NOTIFY message it received (if this field is absent or is not a SIP AoR, the AoR in
request is used instead). If there are unsolicited out-of-dialog NOTIFY messages received from different service
domains, those voice mailboxes will be called in turn (in circular fashion) each time 【MWI】is pressed.
8.1.1 Voice Mailbox AoR

V o i c e M a i l b o x A o R
s i p : v m s @ I S P . f o o . c o m
Call Reset Del Back

 [Call]: Dial to voice mailbox address-of-record if available. Alternatively, You may lift the handset
(offhook) or turn speaker on by pressing 【SPK】 to make a call to selected contact as well.
 [Reset]: Reset voice mailbox address-of-record from address book.
M i c h a e l
M i k e
N i c k
Back
[87/129]
 [Del]: Remove currently configured voice mailbox address-of-record (no MWI subscription).
 [Back]: Save changes and return.
8.1.2 Unsolicited NOTIFY

1 . A c c e p t
2 . R e j e c t
Back
By http://pentest.tele-consulting.com/advisories/05_07_06_voip-phones.txt, there is a weakness of
unsolicited NOTIFY for MWI. You may turn off unsolicited MWI NOTIFY, thus effectively disable “Out-of-
Dialog MWI Notification”.
 [Back]: Return.
8.2 Message
1 . W r i t e M e s s a g e
2 . I n b o x
3 . O u t b o x
4 . D r a f t
Back
Instant messaging service.
Note, YV2 supports “flashing short message” to display received out-of-dialog message on LCD without user
interaction (or any alert) for 2~3 seconds. Such flashing messages are for server-side notification, and they will
not be saved (thus “flashing”). To activate such feature, the received out-of-dialog instant message must carry a
proprietary header “P-Flash-SMS: on” (case-sensitive) in received MESSAGE messages.
 [Back]: Return.
8.2.1 Write Message

Compose a new message.
8.2.1.1 Compose
H e l l o w o r l d ~

 【OK】: Finish composing message.
[88/129]
8.2.1.2 Finish Writing

H e l l o w o r l d
Send Save Abort Write
 [Send]: deliver
 [Save]: save to draft
 [Abort]: Discard message and return.
 [Write]: go back to previous step to modify message.
 【↑】and【↓】: scroll lines if there are too many lines to fit into a screen.
 【OK】: Send
Service Domain
8.2.1.3Send
S e n d T o : I S P . f o o . n e t
Del Realm AB Back

 [Realm]: Circle service domain. (always enable [Abc..] input mode).
 [AB]: Pick a contact from Address Book. (Note, no [Clear] !)
M i c h a e l
M i k e
N i c k
Back
8.2.2 Inbox
1 H e l l o W o r l d !
Unread
2 . A d v e r t i s e
Message
Indicator 3 . D i n n e r t o g e t h e r
Del Clear Back
Store 50 most recently received messages.
List messages in the received order with the latest listed first. Besides, unread messages will have
indicators preceeding them.
 [Del]: Delete selected message.

 [Clear]: Clear all messages
D e l e t e a l l m e s s a g e s ?
Yes No
o [Yes]: Clear all messages from inbox
o [No]: Return to list mode
 [Back]: Return
 【OK】: Read selected message.
 【↑】and【↓】: Navigate through messages.
[89/129]
8.2.2.1Read
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
1 . F r o m : M i c h a e l
2 . C o n t e n t
3 . R e p l y
4 . F o r w a r d
Call Edit Del Back
Show receivd time on the first line. Besides, it also show display or user-part of the email-like address on
“From” item in-line.
 [Call]: Call to message sender.
 [Edit]: Edit sender’s number to save into address book.
 [Del]: Delete this message and return.
 [Back]: Return
 【OK】: enter submenu.
8.2.2.1.1 From/Edit
A d d r : 1 2 3 4 5 6 7 @ I S P . f o o .
 [Call]: Call to this contact.
 [To AB]: Add modified contact into address book and return.
8.2.2.1.2 Content
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
2 . C o n t e n t
3 . R e p l y
4 . F o r w a r d
Call Del Back
 [Back]: Return
 【OK】: Edit content
H e l l o W o r l d !

 [Abc..]/[123..]: Toggle between digits and alphanumeric input, where. [Abc..] indicates current
[90/129]
8.2.2.1.3 Reply
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
2 . C o n t e n t
3 . R e p l y
4 . F o r w a r d
Call Del Back
 [Back]: Return
 【OK】: Compose a new message and reply to sender.
The process is similiart to write a brand new message, except for
1. It will return to “Reply” item on finishing.
2. The default target number on sending is the original message sender.
M i c h a e l < s i p : 1 2 3 4 5 6 7 @ I
S P . f o o . c o m >
Del Realm AB Back
8.2.2.1.4 Forward
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
2 . C o n t e n t
3 . R e p l y
4 . F o r w a r d
Call Del Back
 [Back]: Return
 【OK】: Edit the original message and forward to another contact.
The process is similiart to write a new message, except for
1. It will return to “Forward” item on finishing.
2. The default message on composing is the original message.
H e l L o w o r l d
8.2.3 Outbox
1 . H e l l o W o r l d !
Del Clear Back

Keeps 20 most recently sent messages.
List all sent messages by time with the latest listed first.

[91/129]
Yes No
o [Yes]: Clear all messages from outbox
 [Back]: Return
8.2.3.1Read
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
1 . T o : M i c h a e l
2 . C o n t e n t
3 . R e s e n d
4 . F o r w a r d
Call Edit Del Back
Show sent time on the first line. Besides show display or user-part of the email-like address on “To” item in-
line.
 [Call]: Call to target number.
 [Edit]: Edit target number to save into address book.
 [Back]: Return
 【OK】: enter submenu.
8.2.3.1.1 To/Edit
A d d r : 1 2 3 4 5 6 7 @ I S P . f o o .
 [To AB]: Add modified contact into address book and return.
8.2.3.1.2 Content
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
1 . T o : M i c h a e l
2 . C o n t e n t
3 . R e s e n d
4 . F o r w a r d
Call Del Back
 [Back]: Return
 【OK】: Edit content
H e l l o W o r l d !

[92/129]

8.2.3.1.3 Resend
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
1 . T o : M i c h a e l
2 . C o n t e n t
3 . R e s e n d
4 . F o r w a r d
Call Del Back
 [Back]: Return
 【OK】: Edit original message and resend to target number.
The process is similiart to write a brand new message, except for
1. It will return to “Resend” item on finishing.
2. The default message on composing is the original message.
H e l L o w o r l d

3. The default target number on sending is the original message sender.
M i c h a e l < s i p : 1 2 3 4 5 6 7 @ I
S P . f o o . c o m >
Del Realm AB Back
8.2.3.1.4 Forward
2 1 : 3 2 : 1 5 1 0 / 2 4 / 2 0 0 6
1 . T o : M i c h a e l
2 . C o n t e n t
3 . R e s e n d
4 . F o r w a r d
Call Del Back
 [Call]: Call to this contact
 [Back]: Return
 【OK】: Edit the original message and forward to another contact.
The process is similiart to write a new message, except for
a. It will return to “Forward” item on finishing.
b. The default message on composing is the original message.
H e l l o w o r l d
[93/129]
8.2.4 Draft
1 . H e l l o W o r l d !
Del Clear Back

Store 20 most recently unsent messages.
List all draft messages by time order with the latest listed first. Besides, show the first line on listing.

Yes No
o [Yes]: Clear all draft messages.
 [Back]: Return
H e l l o w o r l d
Send Edit Del Back

 [Send]: deliver => Refer to “Send” on “Write Message”.
 Draft message will be removed once sent.
 [Edit]: Edit messages => Refer to “Compose” on “Write Message”.
 Modified message will overwrite the original draft message.
 [Del]: Delete this draft message and return.
 [Back]: Return.
 【↑】and【↓】: scroll lines if there are too many lines to fit into a screen.
8.3 SNTP
1 . M o d e
2 . S e r v e r
Back
Simple Network Time Protocol, SNTP, is used to acquire current date-time so as to synchronize the date-
time on the phone. Once assigned, the phone will try to synchronize the date-time with the specified NTP server
regularly and convert the returned GMT time to local time based on time zone and daylight saving adjustment.
 [Back]: Return.
8.3.1 Mode
1 . U n i c a s t
2 . M u l t i c a s t
3 . A n y c a s t
4 . D i s a b l e
5 . S I P R e g i s t e r
Back
[94/129]
 [Back]: Return.
The Simple Network Time Protocol is used to synchronize time with YV2. If you set SNTP server to Anycast
mode, the phone will send SNTP query to LAN broadcast address. Otherwise, it sends a request to the specified
SNTP / NTP server, extracting the reported time from the reply, and overwrites the phone’s time. Typically,
SNTP / NTP servers operating in broadcast mode send update messages every 64 to 1024 seconds. The default
time on system starting up is 00:00, January 1, 2007, GMT. If you choose to disable SNTP, it will not
synchronize current system time with NTP server, nor will it adjust daylight saving time. Default is
Unicast.
Unicast Multicast Anycast

SNTP request to the specified Nothing SNTP packet to the local
SNTP server if available. When in multicast mode, network broadcast address,
SNTP requests are not sent. 224.0.1.1.
After the first SNTP response
Sends
is received, the phone
switches to unicast mode with
the server being set as the one
who first responded.
SNTP response from the SNTP data via the SNTP / Unicast SNTP data from the
SNTP server and ignores NTP multicast address from SNTP server that first
responses from other SNTP the local network broadcast responded to the network
Receives servers. address from any server on broadcast request.
the network.
The default multicast address
is 224.0.1.1.
By “SIP Register”, this terminal will synchronize its local time based on the “Date” header present in
successful 2xx SIP responses to REGISTER method. Per rfc3261, it restricts the time zone in SIP-date to "GMT",
in a format similar to (case-sensitive)
Date: Sat, 13 Nov 2010 23:29:00 GMT
As a result, the parsed time will be converted to local time based on time-zone and daylight saving time
(DST) adjustment as well (refer to “3.Phone Settings” / “Date/Time”).
8.3.2 Server
S N T P S e r v e r :
c l o c k . p s u . e d u

Assign the SNTP server. You can use either a dotted IP address or a DNS name. For available SNTP / NTP
servers near your location, please refer to Appendix C – “Available NTP Servers”.
Note: If you enable DHCP, the NTP server may be acquired by DHCP option 42.

[95/129]
8.4 Auto-Provision
1 . P r o t o c o l
2 . S e r v e r
3 . B a t c h D e f a u l t
4 . T e r m i n a l - S p e c i f i c
5 . S e c u r i t y
Back
 [Back]: Return
Configure the provision server to download provisioning files. It is used by LAN administrator for batch
phones configuration. There are two configuration files that you can use to define the SIP parameters: the default
configuration file and the phone-specific configuration file.
We suggest that you use the default configuration file to define values for SIP parameters that are common to
all phones. Doing so will make controlling and maintaining your network easier. You can then define only those
parameters that are specific to a phone in the phone-specific configuration file. Phone-specific parameters should
be defined only in a phone-specific configuration file, or they should be manually configured.
Please refer the following flow chart for the detail of auto-provision, and here summarize the flow:
 If YV2 fails to get each configuration file for 3 times (it tries only once if HTTP is in use), it would
start with the previous configurations.
 After successfully get the configuration file from provision server, YV2 will retrieve parameters from
the file and save the configurations in NV memory for later use.
 In the configuration file, the latest image version is given. YV2 will compare this version with its
current image version stored last time (use empty string for the first time). If the latest version is
different (no mater newer or older) from its current version, YV2 will get the new image from
provision server. YV2 will store this latest version in NV memory, which will be compared next time.
The real version for the image may be different from this version tag.
 Another provision server for downloading image is specified in configuration file. If it does not
specified (the IMAGE_SERVER tag is not present), it will use the provision server where this
configuration file is retrieved from. However, these two provision servers may probably have the same
address. The file name for the new image is also specified in configuration file (the IMAGE tag). See
“Available configurations” for details. The format of the image (compressed or not) is transparent to
the network manager.
 YV2 will store the latest version tag in NV memory after download and store the image file
successfully.
 After finishes the auto-provision flow, it enters business registration flow.
[96/129]
 Auto-provision Flow
System Startup
Read local configuration

Initialize netowrk To terminal service
Auto-provision? No Auto-Provision Done
No Yes
Set target file to be batch-

default configuration file
Get target configuration file Yes

No Set target file to
from provision server
terminal-specific
(TFTP retry 3 times,
configuration file
HTTP tries once)
No
Save factory default,
update new ringing Has processed with
tones and MoH Success? No terminal-specific
configuration file?
Yes Yes
Save new
Success?
configuration
Download new factory Image Version is

default file, user-defined different (even No
ringing tone, and music- older)?
on-hold files from
provision server Yes No
(IMAGE_SERVER)
(Trials: TFTP: 3, HTTP: 1) User confirm
download in one
Yes minute?
Yes No
Has new factory
default file? Download new image from
provision server
(IMAGE_SERVER)
(Trials: TFTP: 3, HTTP: 1)
Save image and

update image Yes Success?
version tag
[97/129]
 Available configurations:
Configuration tag in red font demands privileged account and password to modify them on web pages.
Category Name Type Sample
General
TERMINAL_SPEAKER_VOLUME Integer [1-15]
TERMINAL_HANDSET_VOLUME Integer [1-15]
TERMINAL_HEADSET_VOLUME integer [1-15]
Any string less than 32 characters.
Note, this is just a tag for comparison only,
and it may be different from the actual image
version. This is for rolling back to previous
image version. Auto-provision will try to
download the image file from provision server
IMAGE_VERSION String if the tag is different from its current one no
matter it is newer or older!
It could be a multiple-value separated by any
of the following characters, “;,[Space][Tab]”,
such as “backup.isp.com; web.ISP.com”
(without the quotes). And the system will pick
one of them randomly for load balance.
Specify the image filename, including any path, to
download from privision server.
Note: It will wait for user’s confirmation for 60
IMAGE String seconds before actually proceeding to download
new IMAGE. If user does not respond or veto the
download, it will continue without actually
download the new IMAGE.
Provision server IP to download the new image. If
this tag does not present, IP SIP Phone uses the
provision server from which it downloaded this
configuration file.
IMAGE_SERVER String It could be a multiple-value separated by any of the
following characters, “;,[Space][Tab]”, such as
“backup.isp.com, web.ISP.com” (without the
quotes). And the system will try them in sequence
‘till download successfully.
Specify the factory default file to download
from IMAGE _SERVER and saved into phone-
DEFAULTVALUE_FILE String
set. This tag is valid only after a successful
IMAGE download.
Specify the music-on-hold to download from
IMAGE _SERVER and saved into phone-set.
This tag is valid only after a successful
IMAGE download.
The filename extension must be either
HOLD_TONE String
“.pcmu”( for 8k sample rate, 8-bit, mono,
PCM μ-law header-less raw data); or
“.pcma” (for 8000 sample rate, 8-bit, mono
PCM A-law header-less raw data). The
extension is case-insensitive.
Address Book
Address book AB[0-499] AoR “Display”<sip:[email protected]>
Note, the specified AoR will be added into
address book (the numbering of index has
nothing to do with where the records will be
stored on address book.
Note: This AoR will overwrite the old record
if found on address book.
Note: If you don’t specify a protocol, the
default is “sip”.
Note: the protocol filed for “sip” or “tel”
MUST be in lower case.
E.g.:
AB[0] = sip:[email protected]
AB[1] = HR Manager <sip:[email protected]>
AB[2] = dad<sip:[email protected]>
AB[3]=”girl friend” <[email protected]>
// more contacts follows
The available index is from 0 to 499, which
[98/129]
translates to 500 entries in address book.

Besides, the index in are irrelevant; that is to
say, on auto-provision, the terminal will check
its address book to see whether the imported
contact has been existed or not (only protocol
and email-like address are checked against,
and display-name are ignored). If it has been
existed, then nothing will be done; otherwise
find an empty entry and add it.
Voice Mail box <sip:[email protected]>
Note: If the specified number does not exist on
address book, it will be added first.
Speed Dials SPD[0-19] AoR Note 2: If you don’t specify a protocol, the
Note 3: the protocol filed for “sip” or “tel”
E.g.: sip:[email protected]
If the address does not exist on your address
book yet, this AoR will be added into address
book first, and then the specified AoR will be
Call Screen CALL_SCREEN[0-99] AoR added into blocking list.
E.g.: sip:[email protected]
If the address does not exist on your address
book yet, this AoR will be added into address
book first, and then the specified AoR will be
Favorite FAVORITE[0-19] AoR added into blocking list.
[Presence | BLF | SLA];AoR
you should append the full SIP AoR, use a

semicolon, ‘;’, as separator. If the designated
hot line address does not exist on current
address book, it will be added. E.g.:
IMPP[1] = Presence;Boy
IMPP IMPP[0-19] AoR Friend<sip:[email protected]>

Note 2: If you don’t specify a protocol, the
[ Idle | Away | ByRightBack | Busy |
IMPP-State CFG_ACTIVITY_STATUS String
OnThePhone | OutToLunch | OffLine]
Terminal Settings
TERMINAL_RING_VOLUME Integer [0-12]
Ringer CFG_RINGER_TYPE Integer Starting from ‘0’ and must be non-negative.
[10 | 30 | 1:00 | 2:00 | 3:00 | 5:00 | 10:00 | On]
LCD Back-Light CFG_LCD_BACK_LIGHT String
mm:ss or “On”
Phone Lock TERMINAL_LOCK integer [Off | Security | Phone]
TERMINAL_LOCALE string [en | zh_cn | zh_tw | jp | es_mx]
Date/Time
SNTP SNTP_MODE String [unicast | multicast | anycast | disabled |
sip_register]
[99/129]

such as “backup.isp.com, web.ISP.com”
(without the quotes). And the system will try
them in sequence ‘till download successfully.
SNTP_SERVER String Factory default is any of the following:
clock.psu.edu, tick.utoronto.ca,
ntp.shim.org, tick.keso.fi, ntp.ucsd.edu,
timex.cs.columbia.edu, ntp.obspm.fr,
ntp0.uk.uu.net, ntp.cyber-fleet.net
CURRENT_DATE_TIME Date YYYY / MM / DD HH:MM:SS
TERMINAL_DATE_FORMAT String [M/D/Y | D/M/Y | Y/M/D | Y/D/M | Y-M-D]
TIMEDEF_TIME_ZONE_OFFSET time [GMT][+ | -]00:00
TIMEDEF_DST_OFFSET time [+ | -]0:00
HH:mm MM/DD |
DST_START time HH:mm MM-[0-6]-[Sun|Mon|Tue|Wed|Thu|
DST Fri|Sat]
HH:mm MM/DD |
DST_END time HH:mm MM-[0-6]-[Sun|Mon|Tue|Wed|Thu|
Fri|Sat]
DSS Features
[ CFG_DSS_NO_FUNCTION |
CFG_DSS_ADDRESS_BOOK |
CFG_DSS_AUTO_ANSWER |
CFG_DSS_CALL_DETAIL |
CFG_DSS_CALL_HISTORY |
CFG_DSS_CALL_RETURN |
CFG_DSS_CHANNEL_INFO |
CFG_DSS_CODEC |
CFG_DSS_DND |
CFG_DSS_DTMF_LIST |
CFG_DSS_DTMF_RELAY |
CFG_DSS_FAVORITES |
CFG_DSS_FORWARD |
CFG_DSS_IMPP |
DSS features DSS[0-15] String CFG_DSS_INSTANT_MESSAGE |
CFG_DSS_LOGOUT |
CFG_DSS_NETWORK_INFO |
CFG_DSS_PACKETIZATION |
CFG_DSS_PHONE_LOCK |
CFG_DSS_REJECT |
CFG_DSS_REGISTRATION |
CFG_DSS_SERVICE_REALM]

Note 2: If you don’t specify a protocol, the
Alarm
Alarm ringer CFG_ALARM_RINGER_TYPE integer Starting from ‘0’ and must be non-negative.
Account
PRIVILEGED_ACCOUNT String N/A
PRIVILEGED_PASSWORD String [* | # | 0-9] , max length is 27
Account TERMINAL_ACCOUNT String N/A
TERMINAL_PASSWORD String [* | # | 0-9] , max length is 27
Tones
CFG_DIAL_TONE String 350,0,0,440,0,0,0
CFG_BUSY_TONE String 480,500,500,620,500,500,0
CFG_RINGBACK_TONE String 440,1200,3000,480,1200,3000,0
CFG_CALLWAITING_TONE String 350,100,100,440,100,100,3
CFG_ALERT_TONE String 440,200,0,440,200,0,1
CFG_REORDER_TONE String 480,250,250,620,250,250,0
CFG_SPECIAL_INFO_TONE String 350,100,100,440,100,100,10
[100/129]
Alert-Info
Eg.: ringtone-0
CFG_ALERT_INFO_TAG[i] String
Where i ranges from 0 to 11
Starting from ‘0’ and must be non-negative ,
CFG_ALERT_INFO[i] integer
Where i ranges from 0 to 11
Call Forward
Voice Mail box <sip:[email protected]>
Target number FORWARD_TARGET AoR Note 2: If you don’t specify a protocol, the
Unconditional [0 | 1 | true | false | on | off | enable | disable]
FORWARD_ALL_CALL_FWD boolean
Forward
Busy Forward FORWARD_BUSY_FWD boolean [0 | 1 | true | false | on | off | enable | disable]
No Answer [0 | 1 | true | false | on | off | enable | disable]
FORWARD_NO_ANSWER_FWD boolean
Forward
No answer timer FORWARD_NO_ANSWER_TIMEOUT integer [10-600]
Preferences
Auto Answer CFG_AUTO_ANSWER boolean [0 | 1 | true | false | on | off | enable | disable]
Call Waiting CFG_CALL_WAITING_ALARM boolean [0 | 1 | true | false | on | off | enable | disable]
Auto Hold on [0 | 1 | true | false | on | off | enable | disable]
AUTO_HOLD_ON_CALL_SWITCH boolean
Call Switch
Follow Redirect FOLLOW_REDIRECT boolean [0 | 1 | true | false | on | off | enable | disable]
Xfer after Conf CONFERENCE_JOIN boolean [0 | 1 | true | false | on | off | enable | disable]
Message Alert CFG_MESSAGE_ALERT boolean [0 | 1 | true | false | on | off | enable | disable]
LAN Dial CFG_LAN_DIAL boolean [0 | 1 | true | false | on | off | enable | disable]
Show User-Part
CFG_SHOW_USERPART_AS_CID boolean [0 | 1 | true | false | on | off | enable | disable]
as Caller-ID
Play Alerting Tone
CFG_PLAY_OOB_TONE_ON_CONFERENCE boolean [0 | 1 | true | false | on | off | enable | disable]
during Conference
Dial Timeout DIAL_TIMEOUT integer [30-600]
Hold Recall [10-600]
HOLD_RECALL_TIMER integer
Timer
Inter-digit [3 | 9]
CFG_INTER_DIGIT_TIMEOUT integer
Timeout
Unhold On [0 | 1 | true | false | on | off | enable |
CFG_UNHOLD_ON_TRANSFER boolean
Transfer disable], default is false
Dial key CFG_DIAL_KEY String POUND | # | STAR | * | FLASH | & | DSS
[0 | 1 | true | false | on | off | enable |
CFG_HOT_LINE_MODE boolean
disable], default is false
Hot Line
E.g.: “888” or “sip:[email protected]” or
CFG_HOT_LINE_ADDR String
“[email protected]”
Dial Plan CFG_DIAL_PLAN String 911 | [1-9]r3 | .>#-
AUTOREDIAL_STOP_CRITERION String [ringing | connected]
Auto-redial CFG_AUTOREDIAL_RETRY_INTERVAL Integer 1-255 seconds
CFG_AUTOREDIAL_TOTAL_DURATION integer 30-86400 seconds
Voice Mailbox
Voice Mailbox <sip:[email protected]>
Voice Mail URI CFG_SERVICE_MESSAGE_URI AoR Note 2: If you don’t specify a protocol, the
CFG_ACCEPT_UNSOLICITED_MWI_NOTIF [0 | 1 | true | false | on | off | enable | disable]
Unsolicited Notify Y
boolean
Call Command
Conference Server *#
IPRCFG_CALL_COMMAND_MSAC string Max input is 8 characters.
Access Code
[101/129]
*67
Anonymous call IPRCFG_CALL_COMMAND_CLIR string
Max input is 8 characters.
*82
CLIP IPRCFG_CALL_COMMAND_CLIP string
#37
MWI IPRCFG_CALL_COMMAND_MWI string
&
Server Hold IPRCFG_CALL_COMMAND_SERVER_HOLD string
**
Call Pickup IPRCFG_CALL_COMMAND_CALL_PICKUP string
IPRCFG_CALL_COMMAND_CUSTOMIZED[ i ranges from 0 to 9.
Feature Access-i i]
String
Network
NETWORK_MODE integer [static | DHCP | PPPoE]
NETWORK_HOST_IP IP 192.168.0.10
NETWORK_NET_MASK IP 255.255.255.0
NETWORK_GATEWAY_IP IP 192.168.0.254
NETWORK_PPPOE_ACCOUNT String 55 character at most
NETWORK_PPPOE_PASSWORD String 55 character at most
NETWORK_STATIC_DNS boolean [0 | 1 | true | false | on | off | enable | disable]
NETWORK_PRIMARY_DNS_IP IP: 192.168.0.1
NETWORK_SECONDARY_DNS_IP IP: 192.168.0.1
NETWORK_TTL integer [75-255]
CFG_NETWORK_DOMAIN String SIP.isp.com
CFG_NETWORK_DHCP_HOST_NAME String
RTP_PORT_BASE integer [2-65534];
[2-100]. The value 2 is for single line
RTP_PORT_RANGE integer
configuration
[0-255]
RTP_TOS integer
e.g.:0xb8 or 184
RTP [0-255]
RTP_TOS_4_VIDEO integer
e.g.:0x98
[0-255]
RTCP_TOS integer
e.g.:0x74 or 116
RTP_SYMMETRIC_FLOW boolean [0 | 1 | true | false | on | off | enable | disable]
SIP.isp.com
NAT_STUN_DOMAIN String stun.SIP.isp.com (61.222.26.124)
_stup._udp.SIP.isp.com
NAT & Firewall NAT_MODE String [public | NAT | STUN | UPnP | VPN]
211.19.200.254
NAT_IP String
gateway.SIP.isp.com
CFG_DYNAMIC_NAT_IP_MODE String [notify | auto-refresh]
private
CFG_SNMP_READ_COMMUNITY String
Note, it is case-sensitive
private
CFG_SNMP_WRITE_COMMUNITY String
SNMP public
CFG_SNMP_TRAP_COMMUNITY String
CFG_SNMP_MANAGER[0] String
CFG_SNMP_MANAGER[1] String
NETWORK_VLAN_MODE boolean [0 | 1 | true | false | on | off | enable | disable]
VLAN NETWORK_VLAN_ID integer 0-4095
NETWORK_VLAN_COS integer 0-7, default is 6
SIP Settings
SIPSERVICE_TRANSPORT string [TCP | UDP | TLS]
SIPSERVICE_SIP_PORT Integer [1-65535]
CFG_REGISTER_EXPIRE integer [20-86400] measured in seconds
SIPSERVICE_KEEP_ALIVE_TIMER integer [0-86400] measured in seconds
500
CFG_SIP_TIMER_T1 integer
SIP Service [200-4000] measured in milliseconds.
[0-255]
SIPSERVICE_IP_TOS integer
e.g.:0xA8 or 168
CFG_SIP_SERVICE_RPORT
[0 | 1 | true | false | on | off | enable | disable]
SIPSERVICE_HOLD_BY_3261 boolean
EXPOSE_SOFTWARE_VERSION [0 | 1 | true | false | on | off | enable | disable]
SIPTGATEWAY_MANUAL String [SIP | TEL] URI Format
[102/129]
CFG_INTERNATIONAL_ACCESS_CODE String
ENUM & E.164 ENUM_SUFFIX String e164.arpa
ENUM_MIN_VALID_LEN integer 1-16
Nth Domain
SIPSERVICE_DOMAIN[0-4]_ENABLE boolean [0 | 1 | true | false | on | off | enable | disable]
SIPSERVICE_DOMAIN[0-4]_LOGIN String 31 character at most
SIPSERVICE_DOMAIN[0-4]_PASSWORD String 31 character at most
Authentication SIPSERVICE_DOMAIN[0- [0 | 1 | true | false | on | off | enable | disable]
boolean
4]_NO_QOP_DIGEST_AUTH
“display”
SIP AoR SIPSERVICE_DOMAIN[0-4]_ADDRAOR String
<sip:[email protected]>;user=phone
SIPSERVICE_DOMAIN[0- [0 | 1 | true | false | on | off | enable | disable]
Auto-Answer 4]_AUTO_ANSWER
boolean
SIPSERVICE_DOMAIN[0- [auto-locate | default | rfc3361]
Integer
4]_SIPPROXYSVR_MANUAL
SIPSERVICE_DOMAIN[0-
String
4]_SIPPROXYSVR_FQDN
SIPSERVICE_DOMAIN[0- [TCP | UDP | TLS]
String
4]_SIPPROXYSVR_TRANSPORT
Proxy Server SIPSERVICE_DOMAIN[0- [1-65535]
Integer
4]_SIPPROXYSVR_UDPPORT
SIPSERVICE_DOMAIN[0- [1-65535]
Integer
4]_SIPPROXYSVR_TCPPORT
Integer
4]_SIPPROXYSVR_TLSPORT
SIPSERVICE_DOMAIN[0- [proxy | auto-locate | assigned | multicast ]
Integer
4]_SIPREGSVR_MANUAL
SIPSERVICE_DOMAIN[0- 85 characters at most
String
4]_SIPREGSVR_FQDN
SIPSERVICE_DOMAIN[0- [TCP | UDP | TLS]
String
4]_SIPREGSVR_TRANSPORT
Registrar SIPSERVICE_DOMAIN[0- [1-65535]
Integer
4]_SIPREGSVR_UDPPORT
Integer
4]_SIPREGSVR_TCPPORT
Integer
4]_SIPREGSVR_TLSPORT
Keep NAT Alive SIPSERVICE_DOMAIN[i]_KEEP_ALIVE[0-4] boolean [0 | 1 | true | false | on | off | enable | disable]
Auto Provision
CFG_AUTO_PROVISION_MODE String [NONE | TFTP | HTTP]
such as “backup.isp.com, web.ISP.com”
CFG_AUTO_PROVISION_SERVER String
(without the quotes). And the system will pick
one of them randomly for load balance. 80
characters at most
256 characters at most
CFG_AUTO_PROVISION_BATCH String
Default is “sipDefault.cnf”
256 characters at most
e.g. “8855.cfg”. Default is “sip$MAC$.cnf”
CFG_AUTO_PROVISION_CUSTOMIZED String where the pattern “$MAC$” will be replaced
by the MAC address represented in 12-digit
hexadecimals.
Advanced
Ping CFG_PING_REMOTE_HOST String sip.SIP.isp.com
CODEC
CNG CODEC_SILENCE_SUPPRESSION boolean [0 | 1 | true | false | on | off | enable | disable]
CODEC_PREF_711U
CODEC_PREF_711A
Preference CODEC_PREF_729 integer [0-8]:0: disabled, 1 highest, 8 lowest
CODEC_PREF_723_53
CODEC_PREF_723_64
Voice [10/30 | 20/30 | 30 | 30/30 |
RTP_FRAME_PER_PACKET integer 40/30 | 40/60]
Packetization
[96-128], default is 101, and 128 to disable
RFC2833 CODEC_RFC2833_PAYLOAD_TYPE integer
RFC2833.
[103/129]
DTMF Relay CODEC_DTMF_RELAY boolean [0 | 1 | true | false | on | off | enable | disable]

Security
Key to AES encrypts data while
importing configuration data from your
CFG_CRYPTO_KEY_IMPORT String
PC or downloading auto-provision file
from provision server.
Key to AES encrypts data while exporting
configuration data to your PC or
CFG_CRYPTO_KEY_EXPORT String
uploading configuration data to provision
server.
Specify whether data should be AES
Security
encrypted by
CFG_CRYPTO_ENCRYPT_EXPORT Boolean CFG_CRYPTO_KEY_EXPORT key on
exporting configuration data to your PC
or uploading it to provision server.
Specify whether the configuration data
should be AES decrypted by
CFG_CRYPTO_DECRYPT_IMPORT Boolean CFG_CRYPTO_KEY_IMPORT key on
importing it from your PC or
downloading from provision server.
Log
Integer 3,3,3,3,3
Log CFG_LOGGER_LEVEL
[32] {[0-9],}32
CFG_LOGGER_TYPE String [File | TCP | UDP]
CFG_LOGGER_ADDRESS String logger.SIP.isp.com
 Batch configuration sample

 IP addresses are represented as dotted decimal
 Port number use decimal value
 No quotation mark will be used even for string
 Double slash, “//”, is used for comment
// ************************************************************************************
// * syntax:
// * 1. anything after double slash, " // ", are ignored 'till end of line, '\n'.
// * 2. each line forms a property
// * 3. key and value separator is any of the following characters encountered:
// * [':' | '\t' | '=' | or ' ']
// * 4. anything after the 1st key-value separator encountered is the property value on this line
// * e.g.:
// * / / this is a comment
// * YourISP = YourISP Inc. // this is inline comment
// * MS Microsoft
// * www.YourISP.com 210.59.188.1
// *********************************************************************************** /
// Terminal settings
TERMINAL_PASSWORD = 0000 // YourISP
TERMINAL_RING_VOLUME = 8
TERMINAL_HEADSET_VOLUME = 8
TERMINAL_HANDSET_VOLUME = 8
TERMINAL_SPEAKER_VOLUME = 8
TERMINAL_DATE_FORMAT = Y-M-D
// forward
FORWARD_NO_ANSWER_TIMEOUT = 30
FORWARD_BUSY_FWD = 1
FORWARD_NO_ANSWER_FWD = 1
// codec
CODEC_PREF_723_53 = 1
CODEC_PREF_723_64 = 2
CODEC_PREF_729 = 3
CODEC_PREF_711A = 4
[104/129]
CODEC_PREF_711U = 5
CODEC_SILENCE_SUPPRESSION = 1 / / CNG
SIPSERVICE_TRANSPORT = TCP | UDP

SIPSERVICE_SIP_PORT = 5060
SIPSERVICE_TLS_PORT = 5061
// System date and time,
TIMEDEF_TIME_ZONE_OFFSET = +8:00 // !! you should change this
TIMEDEF_DST_OFFSET = 0:00
DST_START = 00:00 04/01
DST_END = 00:00 10-01-Mon
// TERMINAL_RING_VOLUME= 10
SIPSERVICE_DOMAIN[0]_SIPPROXYSVR_MANUAL = default // !! you should change this
SIPSERVICE_DOMAIN[0]_SIPPROXYSVR_FQDN= fudge.SIP.isp.com // !! you should change this
SIPSERVICE_DOMAIN[0]_SIPREGSVR_MANUAL = proxy // !! you should change this
SIPSERVICE_DOMAIN[0]_SIPREGSVR_FQDN=fugde.SIP.isp.com //!! you should change this
SIPSERVICE_DOMAIN[0]_SIPREGSVR_TRANSPORT = UDP // !! you may change this
SIPSERVICE_DOMAIN[0]_SIPREGSVR_UDPPORT = 5070 // !! you should change this
SIPSERVICE_DOMAIN[0]_SIPREGSVR_TCPPORT = 5070 // !! you should change this
ENUM_SUFFIX = e164.arpa
// RTP
RTP_PORT_BASE = 35000
RTP_PORT_RANGE = 20
RTP_FRAME_PER_PACKET = 20/30
AUTOREDIAL_RETRY_INTERVAL = 30
AUTOREDIAL_MAX_CONCURRENCY = 6
Hold_RECALL_TIMER = 60
CALL_WAITING_ALARM_INTERVAL = 7
AUTO_Hold_ON_CALL_SWITCH = 1
CFG_BLOCK_ANONYMOUS_CALL = 0
FOLLOW_REDIRECT = 1
DIAL_TIMEOUT = 30
CFG_CALL_WAITING_ALARM = 1
CFG_INTERNATIONAL_ACCESS_CODE = +
CFG_INTER_DIGIT_TIMEOUT = 4
CFG_AUTO_PROVISION_BATCH = sipDefault.cnf
CFG_AUTO_PROVISION_CUSTOMIZED = = sip$MAC$.cnf
// where the pattern $MAC$ will be replaced by the MAC address represented by 12 hexadecimals.
// NAT
NAT_STUN_DOMAIN = stun.SIP.isp.com
// SNTP
// we write “clock.psu.edu” twice to make it have higher chance to hit.
SNTP_SERVER = clock.psu.edu, tick.utoronto.ca, clock.psu.edu // multiple-value, randomly pick one
// Auto-provision
CFG_AUTO_PROVISION_SERVER = 192.168.3.1
SNTP_SERVER
CFG_LOGGER_LEVEL = 3,3,3,3,3,3,3,3,3,3,3,3,3,3 ,3,3,3,3,3,3,3,3,3,3
CFG_LOGGER_TYPE = TCP
CFG_LOGGER_PORT = 56447
8.4.1 Protocol
1 . D i s a b L e d
2 . T F T P
3 . H T T P
Back
[105/129]
Select which protocol to download files for auto-provision. If selected, the phone will try to download two
configuration files from the provision server via either TFTP or HTTP protocol on system startup; otherwise, it
will use the current settings stored in flash ROM.
 [Back]: Return.
8.4.2 Server
S e v e r
1 9 2 . 1 6 8 . 3 . 1

Specify the IP address of provision server. You can use either a dotted IP address or a DNS name. You may
include an alternative server port, such as “10.0.0.1:8000”. Default TFTP server port is UDP 69 and default HTTP
server port is TCP 80. If you did not configure the SIP parameters from a provision server, you can configure
them directly by keypad, a web browser or a telnet client.
You may enter multiple servers separated by any of the following characters, “;,[Space][Tab]”, such as
“backup.isp.com, web.ISP.com” (without the quotes). And the terminal will try them in sequence ‘till download
successfully.
Note: If you enable DHCP, the TFTP server may be acquired by DHCP option 66 (TFTP server IP address).

8.4.3 Batch Default

B a t c h D e f a u l t :
s i p D e f a u l t . c n f

Enter the default configuration file to download from provision server. The file generally is used for LAN
administrator batch configures all the phones, such as proxy server and network environment. We suggest that
you maintain the SIP parameters that are common to all your phones. By maintaining these parameters in the
default configuration file, you can perform global changes without having to modify the terminal-specific
configuration file for each terminal.
If the file could be fetched from provision server, the setting read from the configuration will overwrite the
settings read from flash ROM.
System default is “sipDefault.cnf”.

[106/129]
8.4.4 Terminal-Specific
T e r m i n a l - S p e c i f i c :
s i p $ M A C $ . c n f

Enter the terminal-specific configuration file to download from provision server. If the file could be fetched
from provision server, the setting read from the configuration will overwrite the settings from 『Batch Default』
and those read from flash ROM.
Default is sip$MAC$.cnf, where $MAC$ is the MAC address in hexadecimal representation.

8.4.5 Security
1 . P r o t o c o l
2 . S e r v e r
3 . B a t c h D e f a u l t
4 . T e r m i n a l - S p e c i f i c
5 . S e c u r i t y
On/Off Back
Set personal secret AES key to decipher the downloaded configuration files by AES of 192-bit key length. If
you enabled the security option but left the decryption key blank, the terminal will use built-in AES default key to
decipher them.
 [Back]: Return
D e c r y p t i o K e y :
* * * *

Set personal secret AES key to decipher the downloaded configuration files by AES of 192-bit key
length. If you enabled the security option but left the decryption key blank, the terminal will use built-in
AES default key to decipher them.
Initially show 10 ‘*’ signs but show clear text on user input because alphanumeric characters may be
needed.
[107/129]
[108/129]
9 Advanced
1 . C O D E C
2 . S y s t e m A d m i n
3 . S y s t e m S t a t u s
4 . L o g
Back
 [Back]: Return.
9.1 CODEC
2 . P a c k e t i z a t i o n
3 √ C o m f o r t N o i s e
4 . R F C 2 8 3 3 P T : 1 0 1
5 . D T M F R e l a y b y I N F O
Back
The CODECs available on the terminal includes G.711A (64kbps), G.711μ (64kbps), G.729A (8kbps),
G.729AB (8kbps with CNG), G.723.1 (both 5.3 and 6.4 kbps) and G.723.1A (both 5.3 and 6.4 kbps with CNG).
The default preference is to prioritize them based on their compressed voice quality: the higher quality it is, the
higher priority it will be. That is the default CODEC preference is G.711μ, G.711A, G.726, G.729A then
G.723.1A.
High quality comes in a price: it consumes more precious bandwidth. Take G.711μ for example, it takes as much
as 64 kbps per call, as opposed to 5.3 kbps if G.723.1A is employed. Thus, if you have limited bandwidth for
VoIP, you should consider prioritize those CODECs come with lower bit-rates such that it will use your most
preferred CODEC on making / taking calls.
 [Back]: Return.
9.1.1 Preferences
1 . G . 7 1 1 A - 6 4 k : 1
2 . G . 7 1 1 μ - 6 4 k : 2
3 . G . 7 2 3 . 1 - 6 . 4 k : 3
4 . G . 7 2 9 - 8 k : 6
5 . G . 7 2 3 . 1 - 5 . 3 k : 
Up Down  Back
CODEC Priority List Format: list by priority with most preferred first.
Prioritize between the available CODECs. CODECs will be listed based on their priorities. The smaller the
value is, the higher the priority would be. Those disabled voice CODECs, which preference is zero, will be listed
last. Note, however, if you disable all CODECs, it will auto-arrange the available CODECs and those come with
lower bit-rates take precedence. Besides, you should just lower the priorities of G.711 in most circumstances
rather than overly disabling them such that the phone could gracefully fall back to G.711A or G.711μ when the
peer does not support those low bit-rate CODECs for voice compression.
Default is “Increasing voice quality”: Those come with higher compressed voice quality take precedence.
Such that it will prioritize those CODECs with the highest compressed voice quality first while making calls.
However, it will respect the peer’s preferences while taking calls.
 [Up]: Increase the priority of selected CODEC

 [Down]: Decrease the priority of selected CODEC
 []: Disable selected CODEC
[109/129]

9.1.2 Packetization
1 . 3 0 ( G 7 2 3 . 1 ) / 1 0 m s
2 . 3 0 ( G 7 2 3 . 1 ) / 2 0 m s
3 . 3 0 m s
4 . 3 0 ( G 7 2 3 . 1 ) / 4 0 m s
5 . 6 0 ( G 7 2 3 . 1 ) / 4 0 m s
Back
Configure the default voice packetization transmitted to the peer. The terminal will adjust the audio frames
carried per RTP packet (voice packetization) based on this setting:
Voice Packetization G.723.1 G.729 G.711
30(G.723.1)/10(Others)-ms LAN 1 Frames (10 ms)
30(G.723.1)/20(Others)-ms xDSL, ISDN 2 Frames (20 ms)
1 Frame (30 ms)
30(G723.1)/30(Others)-ms xDSL, ISDN, Cable 3 Frames (30 ms)
30(G.723.1)/40(Others)-ms 56 kbps MODEM
4 Frames (40 ms)
60(G.723.1)/40(Others)-ms Limited bandwidth 2 Frames (60 ms)
The default is『20/30(G.723.1)ms』: 1 audio frame (30 ms voice) per RTP packet for G.723.1 CODEC, and 2
frames (20 ms voice) per packet for all other CODECs such as G.729A and G.711.
Before changing this setting, please refer to the following table “Packet Rate and VoIP Bandwidth
Consumption” to find out the optimal value fit into your environment. We suggest a reasonable packetization
should NOT longer than 40 milliseconds. Pick the longer packetization time unless you have a very limited
bandwidth access at the expense of more latency.
 [Back]: Return.
Note: YV2 will dynamically adjust its voice packetization on a per call basis if the “ptime” attribute is
specified in the peer’s SDP. For example: if “a=ptime:10” is specified, the YV2 will ignore the global voice
packetization and adjust to the negotiated packetization time:
From: 7751 <sip:[email protected]>;tag=22516
CSeq: 1 INVITE
Contact: "7751" <sip:[email protected]:5060>
User-Agent: SIP-Phone/1.1
Content-Length: 171
v=0
o=SIP-Phone 0 0 IN IP4 192.168.3.51
m=audio 5102 RTV/AVP 0 8 18
a=ptime:10
 Packet Rate and VoIP Bandwidth Consumption

Packet rate is especially important for sizing a network against a router because routers are not only
constrained by bandwidth but the number of packets per second (PPS) that they can process.
This table calculates the bandwidth consumption based on uni-directional voice only. So for the total
bandwidth used for media during a simple two-way call, you should multiply the value by two.
CODEC Frame per RTP IP Link Averaged Delay Mean
[110/129]
Opinion
Packet PPP Ethernet Band-width
(kbps) (ms) Score
(Packet Rate) (kbps) 802.3 Utilization
(MOS1)
1 (35 pps2) 16.8 18.4 21.6 28% 37.5
2 (17 pps) 11.2 12 13.6 43.2% 67.5
G.723.1 3 (12 pps) 9.3 9.9 10.9 52.8% 97.5
3.5-3.7
5.3 kbps 4 (9 pps) 8.4 8.8 9.6 59.3% 127.5
5 (7 pps) 7.8 8.2 8.8 64.1% 157.5
6 (6 pps) 7.5 7.7 8.3 67.7% 187.5
1 (35 pps) 17.9 19.5 22.7 31.9% 37.5
2 (17 pps) 12.3 13.1 14.7 47.9% 67.5
G.723.1 3 (12 pps) 10.5 11 12.1 57.1% 97.5
3.8-4.0
6.4 kbps 4 (9 pps) 9.5 9.9 10.7 63.8% 127.5
5 (7 pps) 9 9.3 9.9 68.1% 157.5
6 (6 pps) 8.6 8.9 9.4 71.4% 187.5
1 (100 pps) 42 46.8 56.4 16.5% 15
2 (50 pps) 25.2 27.6 32.4 28.2% 25
G.729ab 3 (35 pps) 19.6 21.2 24.4 36.8% 35
3.7-4.2
8 kbps 4 (25 pps) 16.8 18 20.4 43.5% 45
5 (20 pps) 15.1 16.1 18 48.8% 55
6 (17 pps) 14 14.8 16.4 53.1% 65
1 (100 pps) 100.8 105.6 115.2 59.9% 10
2 (50 pps) 84 86.4 91.2 73.4% 20
G.711 3 (33 pps) 78.4 80 83.2 79.5% 30
4.3-4.7
64 kbps 4 (25 pps) 75.6 76.8 79.2 82.9% 40
5 (20 pps) 73.9 74.9 76.8 85.1% 50
6 (17 pps) 72.8 73.6 75.2 86.6% 60
Those values in blue font are near optimal settings for the specific CODEC in terms of bandwidth consumption
and delay. In general, the more frames carried in one RTP packet, the less bandwidth it consumes but in the cost
of longer delay. We suggest a good setting should not introduce a delay for more than 60 ms.
9.1.3 Comfort Noise

4 . R F C 2 8 3 3 P T : 1 0 1
On/Off Back
You can prioritize those CODECs which are not CNG (Comfortable Noise Generation”) capable over those
are capable of CNG. If you disable “CNG”, then YV2 will first try those not-CNG capable CODECs (such as
G.729A and G.723.1 instead of G.729AB and G.723.1A) during call setup. Note, this is only a best-effort attempt:
whenever YV2 received a silence packet from the peer (SID packets if G.723.1A is adopted), it will automatically
switch to the corresponding CNG-capable CODEC, even you have disabled “CNG”.
Default is enabled.
1
Mean Opinion Score is very subjective and varies from one scoring episode to another depending on a variety of
things, e.g., sample size, acoustic environment, and methodology. The values presented here are for audio with no
packet loss. Some codecs fare better than others under packet-loss conditions. MOS ranges from 1 to 5, indicating
voice quality from poor to excellent, where a higher value inidcates better quality. Generally, the benchmark is PSTN
paid call, routed through telephony network, and its MOS is 5. MOS above 3.5 is think acceptable and a MOS higher
than 4 is suitable for commerial use, comparable to PSTN paid call.
2
PPS: Packet per Second.
[111/129]

 [Back]: Return.
9.1.4 RFC2833 PT
R F C 2 8 3 3 P a y l o a d T y p e :
[ 9 6 - 1 2 7 ]
1 0 1
Del Back
Specify the RTP payload type for the transmission of out-of-band DTMF over RTP as per RFC2833. Valid
value MUST be between 96 and 127 in decimal. Specify 128 to disable RFC2833 and transmit DTMF in-band;
i.e., mixing the DTMF tones with voice and transmitted in audio RTP stream.
Default is 101.

9.1.5 DTMF Relay by INFO

4 . R F C 2 8 3 3 P T : 1 0 1
On/Off Back
 [Back]: Return.
This terminal supports DTMF relay over SIP signaling channel by INFO method (RFC2976) in addition to
DTMF over RTP (by either RFC2833 or mixed with normal voice stream).
INFO sip:3101@ SIP.isp.com SIP/2.0
From: 7751 <sip:7751@ SIP.isp.com>;tag=22516
To: <sip:3101@ SIP.isp.com>;tag=99123
CSeq: 1356 INFO
Contact: "7751" <sip:[email protected]:5060>
User-Agent: IP PHONE/1.2.0
Content-Length: 24
Content-Type: application/dtmf-relay
Signal=1
Duration=250
Once enabled, all DTMF keys, 0-9*#&, will be sent by SIP INFO method; otherwise they will be transmitted
by DTMF over RTP.
[112/129]
9.2 System Admin

1 . R e - R e g i s t e r
2 . L o g o u t
3 . R e s t a r t
4 . S h u t D o w n
5 . R e s e t S y s t e m C o n f i g
6 . F a c t o r y V a l u e s
Back
 [Re-Register]: Re-REGISTER all activated SIP address-of-records to SIP service again without user
confirmation.
If the user has gone off-line (see below) explicitly, restart the auto-registration process to let user go
online again.
 [Logout]: Goes off-line without user confirmation.
Unregister all SIP address-of-records to all activated SIP service immediately and stop the regular auto-
registration scheduling to keep user offline. The phone will keep unregistered until the “Re-REGISTER”
command is issued to go online explicitly. This is useful when you are off work and want all calls go
directly to ring your home lines.
Note, reboot the terminal will clear this status and register all activated SIP address-of-records after
startup.
 [Restart]: Save all configuration data back to NVRAM and restart (reboot) YV2.
 [Shut Down]: Shut down terminal for turning off power.
After shutdown, all indicators (LEDs) and status LED will be off, then you could power off your
terminal safely.
 [Reset System Config]: Reset all system settings back to factory default values (such as restore both
privileged password and user-level password to “0000”) but leave general network settings and personal
information intact.
Those unchanged user data include:
 General Network Settings: Mode, Static Setings, PPPoE Settings, DNS Settings, and IP-TTL.
 Address Book
 Speed Dials
 Call Screen
 Favorite List
 IMPP List
 Call History: Missed, Received and Dialed Calls.
 Instant Messages: Inbox, Outbox and Draft
 Call Statistics
 [Factory Values]: Restore everything, including network settings and personal information, back to
factory values, then auto-reboot the terminal.
 【OK】: Execute command.

1 . R e - R e g i s t e r
2 . L o g o u t
3 . R e s t a r t
4 . S h u t D o w n
5 . R e s e t S y s t e m C o n f i g
6 . F a c t o r y V a l u e s
Yes No
 [Yes]: Confirm execution
 [No]: Return
[Re-Register] and [Logout] will not ask for user’s confirmation but executing immediately!
9.3 System Status

1 . N e t w o r k S t a t u s
2 . S y s t e m U p T i m e
[113/129]
3 . C a l l S t a t i s t i c s
4 . S y s t e m I n f o r m a t i o n
5 . V e r s i o n
Back
Display current system information.
 [Back]: Return
 【OK】: View selected item.
9.3.1 Network Status

I P : 1 9 2 . 1 6 8 . 3 . 2 0 0
M A C : 0 0 0 E C 3 1 2 3 4 5 6
D N S : 1 9 2 . 1 6 8 . 3 . 1
D N S : 1 9 2 . 1 6 8 . 3 . 2 5 4
L A N : 1 0 0 M , F u l l - D u p l e x
P C : : D o w n
Back
Display currently active host IP, Ethernet MAC address and the active DNS IP (secondary DNS will be
shown if available), LAN and PC Ethernet link status in order. The phone will set MAC as it host name in 12
hexadecimal characters.
Note, this item is identical to “6.Network” / “Active Status”.
 [Back]: Return
LAN and PC Ethernet link status are one of the following:

 Down: not connected
 Up: Ethernet jack connected
9.3.2 System Up Time

S y s t e m U p T i m e :
1 3 D 0 2 : 1 9 : 4 3
Back
Show the system up time since last boot-up in the format of “xxxD HH:MM::SS”.
 [Back]: Return
9.3.3 Call Statistics

1 . L a s t e s t 5 0 C a l l s
2 . L a s t 7 2 H o u r s
3 . A c c u m u l a t e d
Ping Back
 [Back]: Return.
 【OK】: Enter selected submenu.
[114/129]
9.3.3.1Lastest 50 Calls
1 . C o n n e c t e d C a l l s : 1 6
2 . F a i l R a t i o : 2 5 %
3 . A v g D u r a t i o n : 1 3 : 2 1
4 . A C K T i m e o u t : 3
Back
Summarize the latest 50 (or fewer if no traffic of such amount is available) calls. Those counters will reset to
zero whenever the system reboots.
1. Success calls: the counter of connected calls in the latest 50 calls.
2. Fail rate: the failed (not connected) ratio of the latest 50 calls.
3. Duration: the average conversation time of those successfully connected calls.
4. ACK timeout: the counter of inbound calls which disconnected due to ACK timeout. Only calls lasted
less than 32 seconds may disconnect due to ACK timeout.
 [Back]: Return
9.3.3.2Last 72 Hours/Accumulated
1 . C o n n e c t e d C a l l s
2 . C a l l A t t e m p t s
Back
“Last 72 hours” summarizes call statistics during the last 72 hours, and “Accumulated” summarizes call
statistics since system startup.
 [Back]: Return
 【OK】: Enter submeu
9.3.3.2.1 Connected Calls

1 . < 3 M i n u t e s
2 . 3 - 2 0 M i n u t e s
3 . 2 0 - 6 0 M i n u t e s
4 . > 1 H o u r
5 . T o t a l
Back
Summarize the connected calls during the last 72 hours or since system startup. Those counters will all reset
to zero whenever the system reboots.
1. <3 Minutes: Summarize those connected (either inbound or outbound) calls which lasted less than 3
minutes.
 Total Call: the accumulated counter during the last 72 hours or since system startup
 Avg. Duration (s): the average conversation time of this category; measured in seconds.
 Ratio (%): the ratio of calls of this category to the total connected (both inbound and outbound)
calls.
2. 3-20 Minutes: Summarize those connected (either inbound or outbound) calls which lasted between 3
and 20 minutes.
3. 20-60 Minutes: Summarize those connected (either inbound or outbound) calls which lasted longer than
20 minutes but shorter than one hour.
4. >1 Hour: Summarize those connected (either inbound or outbound) calls which lasted longer than one
hour.
5. Total: Summarize all connected (either inbound or outbound) calls during the last 72 hours or since
system startup.
[115/129]
 [Back]: Return
 【OK】: View record
9.3.3.2.1.1 <3 Minutes

1 . C a l l : 4 6
2 . D u r a t i o n : 1 : 2 8
3 . R a t i o : 6 3 %
Back
1. Total Call: the accumulated counter during the last 72 hours or since system startup.
2. Avg. Duration (s): the average conversation time of this category; measured in seconds.
3. Ratio (%): the ratio of calls of this category to the total connected (both inbound and outbound) calls.
 [Back]: Return
9.3.3.2.1.2 Total
1 . C a l l : 2 6 8
2 . D u r a t i o n : 4 : 2 1
Back
Summarize all connected (either inbound or outbound) calls during the last 72 hours or since system startup.
 [Back]: Return
9.3.3.2.2 Call Attempts

1 . I n b o u n d
2 . O u t b o u n d
3 . T o t a l
Back
Summarize the inbound and outbound call attempts during the last 72 hours or since system startup.
1. Inbound: Summarize the inbound calls, including
 Total Call: the total inbound calls during the last 72 hours or since system startup.
 Connected Call: the counter of connected (taken/picked-up) inbound calls.
 Fail ratio (%): the ratio of those calls failed to take to the total inbound calls.
 ACK Timeout: the accumulated counter of inbound calls which disconnected due to ACK timeout.
Only inbound calls lasted less than 32 seconds may disconnect due to ACK timeout.
2. Outbound: Summarize the outbound calls.
3. Total: Summarize all call attempts during the last 72 hours or since system startup, including both
inbound and outbound calls.
 [Back]: Return
 【OK】: View record.
9.3.3.2.2.1 Inbound/Outbound
1 . C a l l : 4 6
2 . C o n n e c t e d : 2 3
3 . F a i l R a t e : 5 0 %
4 . A C K T i m e o u t : 4
Back
[116/129]
1. Total Call: the total inbound calls during the last 72 hours or since system startup.
2. Connected Call: the counter of connected (taken/picked-up) inbound calls.
3. Fail ratio (%): the ratio of those calls failed to take to the total inbound calls.
4. ACK Timeout: the accumulated counter of inbound calls which disconnected due to ACK timeout.
 [Back]: Return
9.3.3.2.2.2 Total
1 . C a l l : 5 7
2 . C o n n e c t e d : 3 8
3 . F a i l R a t e : 3 4 %
Back
Summarize all call attempts during the last 72 hours or since system startup, including both inbound and
outbound calls.
1. Call: the total calls, including both inbound and outbound calls, during the last 72 hours or since system
startup.
2. Connected: the counter of all connected calls, which is as the same as the sum of inbound calls and
outbound calls.
3. Fail ratio: the ratio of all failed calls to the total call attempts.
 [Back]: Return
9.3.4 System Information

C P U : A C 4 9 x 1 5 0 M H z
D S P : A C 4 9 5
M e m o r y : 1 6 M B
F l a s h : 4 M B
S / N : 1 2 3 4 5 6 7 9 0 4
Back
Display the CPU family, hardware serial number and hardware information.
 [Back]: Return
9.3.5 Version
F i r m w a r e : Y V 2 1 . 0 . 3
A p p l i c a t i o n : 1 . 0 . 3 . 1 2
D r i v e r : 1 . 0 . 3 . 4
H a r d w a r e : 1 . 0 . 3 . 1
F a c t o r y V a l u e : 1 . 0 . 3 . 4
Back
Display the product version, software and firmware version of the phone.
 [Back]: Return
[117/129]
9.4 Log
1 . L o g C o m p o n e t
2 . L o g g e r T y p e
3 . N e t w o r k L o g g e r
Back
Modules could log debug information to either the embedded COMM port or a network log server. Note:
change log level may have negative impacts on the performance of YV2, such as intermittent voice and lagged
responses.
 [Back]: Return
9.4.1 Log Component

1 . U t i l i t y : F a t a l
2 . N e t w o r k : F a t a l
3 . R T P : F A t a l
4 . S D P : F A t a l
5 . S I P : F A t a l
6 . N e g o t I a t o r : F a t a l
7 . D e v M g R : F a t a l
Back
Define the log verbosity of each module.
 [Back]: Return.
0 . F a t a
1 . E r r o r
2 W a r n i n g
3 . E v e n t
4 . I n f o r m a t i v e
5 . T r a c e
6 . T r a c e 2
7 . T r a c e 3
8 . D e t a i l
9 . D e b u g
Back
Valid values include:
Log Level Verbosity
Fatal (Default) Not recoverable error
Error Recoverable Error
Warning
Event
Informative
Trace
Trace Level 2
Trace Level 3
Detail
Debug Everything
Note, Change the log level for each module may have negative effect on the performance of YV2,
such as intermittent voice and lagged response. Remember to lower the log level or set it back to
default level (Fatal level) whenever you have finished tracing debug information.
 [Back]: Return.
[118/129]
9.4.2 Logger Type

1 . F i l e L o g g e r
2 . T D P L o g g e r
3 . U D P L o g g e r
Back
1. File Logger: output logs to standard output, such as COMM port.
2. TCP Logger: output logs to a network log server by setting up a TCP stream.
3. UDP Logger: output logs to a network log server by UDP packets.
Default is File Logger.
 [Back]: Return.
9.4.3 Network Logger

N e t w o r k L o g g e r :

Assign the network server for log output. You can use either a dotted IP address or a DNS name. The phone
will use port 901 connects to default log server port 56447 (either TCP or UDP). To connect to a different server
port, please specify as “xxx.xxx.xxx.xxx:port”.

[119/129]
Appendix A – “Dial Plan”

10 Dial Plan
The programmable dial plan is designed for the service provider to customize the behavior of the YV2 for
collecting and sending dialed digits. The dial plan allows the terminal user to specify the events that trigger the
sending of dialed digits. These events include the following:
 The termination character has been entered.
 The specified dial string pattern has been accumulated.
 The specified number of dialed digits has been accumulated.
Dial Plan consists of alphanumeric string, and the maximum number of characters is 255.
System default dial plan is empty, which leads to recognize any dial strings ending with a pound, ‘#’, sign
(identical to the dial plan string “.>#-“, excluding the double quotes).
10.1 Dial Plan Commands
 Commands that can be used to create you own dial plans:

Command Description
. Wildcard, match any digit, “0-9#*”, entered.
- Additional (zero or more) digits can be entered. This command can be used only at the
end of a dial plan rule (for example, 1408t5- is legal usage of the - command, but
[120/129]
1408t5-3... is illegal).
Range, which means to match any single digit in the list. Use an underscore () to indicate
a range of digits. For example, [135] matches the digits 1, 3, and 5. Also, [1_5] matches
[] the digits 1, 2, 3, 4 and 5. The pound key (#) and asterisk (*) are not allowed in the
Range command. The Range command does not allow repeat or sub-rule matching.
Sub-rule matching. Using the ( ) and | operators allows you to specify multiple sub-rules
within a dial plan rule so that a sub-rule match is reached if the entered digits fit one of
the sub-rules. This can be used to reduce the length of the desired dial plan rule by
( subrule0 | concatenating the group of the sub-rules with the common rule.
subrule1 | ...|
For example, a dial plan rule of (1900|1800|17..)555.r3 or three dial plan rules of
subruleN ) 1900555.r3|1800555.r3|17..555.r3 are equivalent. A match is reached if 11 digits are
entered and the first three digits are either 1900, 1800, or 17..., and the fifth, sixth, and
seventh digits are all 5.
Defines the # character as a termination character. When the termination character is
entered, the dial string is automatically sent. The termination character can be entered
># only after at least one user-entered digit matches a dial plan rule. Alternatively, the
command >* can be used to define * as the termination character.
Repeat the last pattern n times, where n is 0-9 or a-z. The values a-z indicate a range of
10 to 35. Use the repeat modifier to specify more rules in less space.
rn Note The commands ># and tn are modifiers, not patterns, and are ignored by the rn
command.
| Used to separate multiple dial plan rules.
Logical not. Match any character except the character immediately following the ^
^ command. The ^ command can also be used as a negation instruction before the range or
sub-rule matching commands.
Seize rule matching. If a dial plan rule matches the sequence of digits entered by the user
to this point, and the modifier S is the next command in the dial plan rule, all other rules
are negated for the remainder of the call (for example, a dial plan beginning with *S will
be the only one in effect if the user first enters the * key).
S Note All rules apply in the order listed (whichever rule is completely matched first will
immediately send the dial string).
Note No syntax check is performed by the actual implementation. The administrator has
the responsibility of making sure that the dial plan is syntactically valid.
10.2 Dial Plan Rules
Dial plan rules supported by YV2

 (In Rule) for Dial Plan Blocking
 ‘P’ Rule to Support Dial Prefix
 ‘R’ Rule for Enhanced Prefix
 ‘A’ Rule for Suffix Appending
 ‘C’ Rule for Call Blocking
 ‘X’ Rule for Call Blocking and Call Forwarding Blocking
 ‘D’ Rule for Displaying Caller ID
10.2.1 (In Rule) for Dial Plan Blocking
Dial plan blocking can be used to reduce the occurrences of invalid dialed digits being sent and can prevent
the dialed string of a specified pattern from being sent. By adding dial plan blocking, dialed digits are discarded
after the inter-digit timer expires unless one of the specified matching rules is met.
[121/129]
In addition, the default four-second global inter-digit timeout value is also modified with the value specified
in the dial plan blocking command:
 Syntax: In
where n specifies the global inter-digit timeout and the valid values are 1-9 and a-z (10-35).
 Example: Ic | 911
This command specifies an inter-digit timeout of 12 seconds, and will discard dialed digits unless 911 is
entered.
Specifying your own inter-digit timeout also changes the behavior of the dial plan so that the entire dial
string, rather than being sent at timeout, is sent only as a result of a matching rule or time intended by a
matching rule.
10.2.2 ‘P’ Rule for Dial Prefix
65 This rule is for automatic pre-pending the dial string as entered by the user with a specified prefix.
 Syntax: Ptnnnn
Where t is a single leading trigger character; if t is the first entered digit when making a new call, it triggers
the prepending of a variable-length prefix (as specified by nnnn) in the dial string. The t character can take
one of the following values:
0-9,*,#, 'n' (= any of 1-9), 'N' (any of 'n' and 0), 'a' (any of 'n',* and #), or 'A' (any of 'a' and 0);
 Example: Pn12345
This rule prepends 12345 to the dial string when the first entered digit is any of 1-9. The triggered digit is
not removed from the dial string.
10.2.3 ‘R’ Rule for Enhanced Prefix
This enhanced prefix rule matches entire strings, whereas the ‘P’ rule matches only a single digit. The ‘R’
rule is for automatically prepending a specified prefix to the dialed string. The string must be an exact match to
trigger the rule. If more than one ‘R’ rule matches, the first matched ‘R’ rule is triggered.
The ‘R’ rule also uses negation to exclude one or more leading digits before prepending the defined prefix
string.
The number of dashes (-) after the R represents the number of leading digits that will be removed preceding
the prefix.
 Syntax: Rnnnn(tttt)
Where tttt is a trigger string. If the dialed numbers match this string, this match triggers the prepending of a
variable-length prefix (as specified by nnnn) to the dial string. The triggered string is not removed from the
dial string. The negation, sub-rule matching and range patterns can be applied to the trigger strings.
 Example 1: R1212([2_9]-)
This rule prepends 1212 to dial strings that have a leading digit of 2 to 9.
Note: ‘R’ rules can replace most ‘P’ rules; for example, Pn12345 is the same as R12345([1_9]-).
 Example 2: R-0033(0[1-9].r7)
This removes the first dialed digit, and then prepends 0033 to the dialed string. For example, if the number
0148336134 is dialed, the resulting string becomes 0033148336134.
 Example 3: R----0(0033[1-9].r7)
This removes the first four dialed digits, and then prepends 0 to the dialed string. For example, if the
number 0033148336134 is dialed, the resulting string becomes 0148336134.
[122/129]
10.2.4 ‘A’ Rule for Suffix Appending
This suffix appending rule matches entire strings. The ‘A’ rule is for automatically appending a specified
suffix at the tail of the dialed string. The string must be an exact match to trigger the rule. If more than one ‘A’
rule matches, the first matched ‘A’ rule is triggered.
The ‘A’ rule also uses negation to exclude one or more leading digits before appending the defined suffix.
 Syntax: Tnnnn(tttt)
Where tttt is a trigger string. If the dialed numbers match this string, this match triggers the appending of a
variable-length suffix (as specified by nnnn) to the dial string. The triggered string is not removed from the
dial string. The negation, sub-rule matching and range patterns can be applied to the trigger strings.
 Example 1: A1234([2_9]-)
This rule appends 1234 to dial strings that have a leading digit of 2 to 9.
 Example 2: [email protected](8[1-9].r3)
This appends “@10.0.100.1” to the dialed 4-digit string prefixed by a ‘8’. For example, if the number
“8192” is dialed, the resulting string becomes [email protected]
 Example 3: [email protected](00-)
This appends “@192.168.1.254” to any dialed string starts with ‘00’. For example, if the number
0033148336134 is dialed, the resulting string becomes [email protected].
10.2.5 ‘C’ Rule for Call Blocking
This rule is for blocking call numbers.
 Syntax: Cnnnn
Where nnnn is the leading set of digits of the blocked call number; nnnn can be composed with sub-rule
matching and range. The rule is triggered when the leading digits of a dialed string match the string nnnn.
The ‘C’ rule does not work with negation.
 Example: C1900|C1888 or C(1900|1888)

This rule blocks call numbers beginning with 1900 or 1888.
10.2.6 ‘X’ Rule for Call Blocking and Call Forwarding Blocking
This rule is for blocking call numbers and call forwarding numbers.
 Syntax: Xnnnn
Where nnnn is the leading set of digits of the blocked call number and blocked call forwarding number;
nnnn can be composed with sub-rule matching and range. The rule is triggered when the leading set of digits
of a dialed call number or forwarding number match the string nnnn. The ‘F’ rule does not work with
negation.
 Example: X1900|X1888 or X(1900|1888)

This rule blocks the call numbers and call forwarding numbers beginning with 1900 or 1888.
[123/129]
10.2.7 ‘D’ Rule for Displaying Caller ID
This rule is for displaying caller ID at the remote site. The number must be an exact match to trigger the rule.
 Syntax: Dnnnn
Where nnnn is the callee number. The caller ID will show to the callee; nnnn automatically becomes a valid
calling number. Also, nnnn can be composed with negation, sub-rule matching and range. The ‘D’ rule is
checked before the ‘R’ and ‘P’ rules.
 Example: D911
This rule shows the caller ID at the remote side when if the call number is 911.
10.3 Dial Plan Examples
10.3.1 Example 1
The following dial plan:

*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#-
consists of the following rules:
 *St4-: If the first digit entered is *, all other dial plan rules are voided. More digits can be entered after the
initial * digit, and the timeout before automatic dial string send is four seconds.
 #St4-: same as above, except for # as the initial digit entered.
 911—if the dial string 911 is entered, send it immediately.
 1>#t8.r9t2-: If the first digit entered is 1, the timeout before automatic send is eight seconds. The
terminating character # can be entered at any time to manually send the dial string. After the 11 th digit is
entered, the timeout before an automatic send changes to two seconds. The user can enter more digits until
the dial string is sent by the timeout or by the user entering the # character.
 0>#t811.rat4-: If the first digit entered is 0, the timeout before automatic send is eight seconds, and the
terminating character # can be entered at any time to manually send the dial string. If the first three digits
entered are 011, then, after an additional 11 digits are entered, the timeout before an automatic send changes
to four seconds. The user can enter more digits until the dial string is sent by the timeout or by the user
entering the # character.
^1t4>#-: If the first digit entered is anything other than 1, the timeout before an automatic send is four seconds.
The terminating character # can be entered at any time to manually send the dial string. The user can enter more
digits until the dial string is sent by the timeout or by the user entering the # character.
10.3.2 Example 2
The following dial plans:

.t7>#......t4-|911|1t7>#..........t1-|0t4>#.t7-
or
.t7>#r6t4-|911|1t7>#.r9t1-|0t4>#.t7-
consist of the following rules:
 .t7>#r6t4-: You must enter at least one digit. After the first digit is entered and matched by the dial plan, the
timeout before an automatic send is seven seconds, and the terminating character # can be entered at any
time to manually send the dial string. After seven digits are entered, the timeout before an automatic send
changes to two seconds. The - symbol at the end of the rule allows further digits to be entered until the dial
[124/129]
string is sent by the timeout or the user entering the # character.

 911: if the dial string 911 is entered send this string immediately.
 1t7>#.r9t1-: If the first digit entered is 1, the timeout before an automatic send is seven seconds, and the
terminating character # can be entered at any time to manually send the dial string. After the 11th digit is
entered, the timeout before an automatic send changes to one second. The user can enter more digits until
 0t4>#.t7-: If the first digit entered is 0, the timeout before an automatic send is four seconds, and the
terminating character # can be entered at any time to manually send the dial string. After the second digit is
entered, the timeout before an automatic send changes to seven seconds. The user can enter more digits until
10.3.3 Example 3
The following dial plan:

R1408([2_9].r5|[2_9].r6)|R9^(911|.r4)|X(1900|1888)
consists of the following rules:
 R1408([2_9].r5|[2_9].r6) : The prefix 1408 will be added to any call numbers with seven or eight digits
where the leading digit is in the range of 2 to 9. For example, 5551234 will become 14085551234, but
555123 does not match this rule.
 R9^(911|.r4) : The prefix 9 will added to any numbers except 911 and five-digit numbers. For example, 911
will still be 911, and 51234 will still be 51234.
 X(1900|1888) : There will be no calls or call forwarding to numbers beginning with 1900 or 1888.
[125/129]
Appendix B – “Available NTP Servers”

Service Level 2 NTP Server Service Area
North America
Ontario, Canada: University of tick.utoronto.ca Eastern Canada
Toronto tock.utoronto.ca
Quebec, Canada: Canadian ntp1.cmc.ec.gc.ca Eastern Canada
Meteorological Center ntp2.cmc.ec.gc.ca
Ontario, Canada: National time.chu.nrc.ca; Canada
Research Council of Canada time.nrc.ca
Service Area
Saskatchewan, Canada: 142.3.100.15: timelord.uregina.ca Canada
University of Regina
Mexico: Audiotel office ntp2a.audiotel.com.mx; Avantel, MCINet, and
ntp2c.audiotel.com.mx; Mexico
ntp2b.audiotel.com.mx
Santa Cruz, CA: Scruz-net, inc. 165.227.1.1: ns.scruz.net Western U.S.
San Diego, CA: UCSD Academic 132.239.254.49: ntp.ucsd.edu CERFNET; NSFNET,
Computing Services / Network SDSC region, and nearby
Operations
Quincy, California ntp1.mainecoon.com; North America
ntp2.mainecoon.com
Newark, DE: University of 128.175.1.3: louie.udel.edu CAIRN, Abilene, and
Delaware vBNS
Atlanta, GA: CNSG ntp.shorty.com Southeast U.S.
Kennesaw, GA: PeachNet rolex.peachnet.edu; PeachNet (Georgia) and
timex.peachnet.edu Southeast U.S.
Urbana-Champaign, IL: ntp-0.cso.uiuc.edu; CICNET, Midwest, and
University of Illinois ntp-1.cso.uiuc.edu; NCSA region
ntp-2.cso.uiuc.edu
Chicago, IL: Argonne National ntp-1.mcs.anl.gov; NSF / ANSNet, CICNet,
Laboratory ntp-2.mcs.anl.gov NetIllinois, and ESNet
West Lafayette, IN: Purdue gilbreth.ecn.purdue.edu; NSFNET and CICNET
University harbor.ecn.purdue.edu; area
molecule.ecn.purdue.edu
Manhattan, KS: KansasNet 199.240.130.1: ntp1.kansas.net Central U.S. and Great
OnLine Services 199.240.130.12: ntp2.kansas.net Plains
Boston, MA: UMass-Boston CS 158.121.104.4: New England
dept. timeserver.cs.umb.edu
Minneapolis / St Paul, MN: ns.nts.umn.edu; CICNET region
University of Minnesota nss.nts.umn.edu
Columbia, MO: University of 128.206.206.12: MOREnet
Missouri-Columbia everest.cclabs.missouri.edu
Omaha, NE: Radiks Internet 205.138.126.83: allison.radiks.net Midwest U.S.
Access
Las Vegas, NV: University of 131.216.1.101: cuckoo.nevada.edu NevadaNet, NSFNET, and
Nevada System Computing SDSC region
Services
Las Vegas, NV: UNLV College of tick.cs.unlv.edu; Sprintnet
Engineering tock.cs.unlv.edu
New York City, NY: Columbia ntp.ctr.columbia.edu Sprintlink and NYSERnet
University
Ithaca, NY: Cornell University 192.35.82.50: ntp0.cornell.edu NSFNET and NYSER
region
New York, NY: Morningside sundial.columbia.edu NYSERnet
[126/129]
Campus, Columbia University

New York City, NY: Columbia timex.cs.columbia.edu PSINET; NSFNET, and
University Computer Science NYSER region
Department
Norman, Oklahoma: University 129.15.22.8: Midnet
of Oklahoma constellation.ecn.uoknor.edu
Coos Bay, OR: Koala Computers 207.48.109.6: tick.koalas.com Northwestern U.S.
University Park, PA: Penn State 128.118.25.3: clock.psu.edu Internet2, vBNS,
University CERFnet(AT&T IP
Services), PSC / NCNE,
and CASC
Pittsburgh, PA: Pittsburgh 128.182.58.100: fuzz.psc.edu NSFNET and PSC region
Supercomputing Center
Pittsburgh, PA: Carnegie Mellon ntp-1.ece.cmu.edu; PREPNET and PSC region
Electrical and Computer ntp-2.ece.cmu.edu
Engineering
Dallas, TX: Cox School of 129.119.80.126: ntp.cox.smu.edu NSFNET and SESQUI
Business, Southern Methodist region
University
Hughes Springs, TX: First 209.144.20.76: ntp.fnbhs.com Northeast Texas
National Bank
Houston, Tx: Baylor College of 128.249.1.10: ntp.tmc.edu NSFNET and SESQUI
Medicine region
College Station, TX: Texas A&M 165.91.52.110: ntp5.tamu.edu NSFNET, SESQUI region,
University THEnet, and TAMUSDSN
Plano, TX: Greyware Automation tick.greyware.com; South-Central U.S.
Products tock.greyware.com
Blacksburg, VA: Virginia Tech ntp-1.vt.edu; Southeast U.S.
Computing Center ntp-2.vt.edu
Arlington, VA: Center for Seismic 140.162.1.3: ntp.cmr.gov NSFNET and SURA
Studies region
Richland, WA: Washington State 192.31.216.30: NSFNET and
University clock.tricity.wsu.edu NorthWestNet
Washington, U.S.: Mill Creek time.ultimeth.net Northwest U.S.
Wisconsin, U.S.: Computer ntp1.cs.wisc.edu; U.S. and any
Science Department, University ntp2.cs.wisc.edu;
of Wisconsin-Madison ntp3.cs.wisc.edu
South America
Buenos Aires, Argentina: 200.49.40.1: tick.nap.com.ar Argentina

Network Access Point 200.49.32.1: tock.nap.com.ar
Buenos Aires, Argentina: Sinectis time.sinectis.com.ar Argentina
S.A.
Brazil: Brazilian Research 200.144.121.33: ntp.cais.rnp.br Brazil
Network
Venezuela: VELUG, Grupo de 150.185.192.250: ntp.linux.org.ve Arica
Usuarios Linux de Venezuela
Europe
Zurich, Switzerland: Swiss Fed. 129.132.98.11: bernina.ethz.ch Switzerland and Europe
Inst. of Technology
Location: Copenhagen, Denmark, clock.netcetera.dk; Denmark, Scandinavia,
Europe clock2.netcetera.dk and Northern Europe
Spain: Altea slug.ctv.es Spain
Pieksamaki, Finland: Keski- tick.keso.fi; Finland
Savon Oppimiskeskus tock.keso.fi
Meudon, France: Observatoire de ntp.obspm.fr France and Europe
[127/129]
Paris-Meudon
Lyon, France: CISM ntp.univ-lyon1.fr France, Switzerland, Italy,
and Europe
Paris, France: VIA, Ecole 138.195.130.70: ntp.via.ecp.fr France and Europe
Centrale
Budapest, Hungary: KFKI 148.6.0.1: time.kfki.hu HUNGARNET
Research Institute for Particle and
Nuclear Physics
Italy: Net4u Srl, Vercelli, Italy 195.32.52.129: ntps.net4u.it Italy
Oslo, Norway: University of Oslo 129.240.64.3: fartein.ifi.uio.no NORDUnet
Oslo, Norway: Alcanet time.alcanet.no Europe
International
Krakow, Poland: Academic 149.156.4.11: info.cyf-kr.edu.pl Poland and Europe
Computer Centre
Lund, Sweden: Lund Institute of 130.235.20.3: ntp.lth.se Sweden and NORDUnet
Technology
Ljubljana, Slovenia: Institute of biofiz.mf.uni-lj.si 193.2.69.11 Slovenia and Europe
Biophysics, University of
Ljubljana
Ljubljana, Slovenia: hmljhp.rzs-hm.si Slovenia and Europe
Hydrometeorological Institute of
Slovenia
Ljubljana, Slovenia: Academic ntp1.arnes.si; Slovenia and Europe
and Research Network of ntp2.arnes.si
Slovenia
Ljubljana, Slovenia: J. Stefan time.ijs.si Slovenia and Europe
Institute
British Isles
Dublin, Ireland: School of ntp.cs.tcd.ie; Ireland and U.K.
Mathematics, Trinity College ntp.maths.tcd.ie;
ntp.tcd.ie
Glasgow, Scotland: Strathclyde ntp.cs.strath.ac.uk U.K., Europe, and any
University
Cambridge, U.K. ntp0.uk.uu.net UUNET (formerly known
ntp1.uk.uu.net; in the U.K. as PIPEX) and
ntp2.uk.uu.net U.K.
Manchester, England: University ntp2a.mcc.ac.uk; U.K.
of Manchester ntp2b.mcc.ac.uk;
ntp2c.mcc.ac.uk;
ntp2d.mcc.ac.uk
Buckinghamshire U.K.: Wibble tick.tanac.net Service area: U.K.
U.K., Aylesbury
Asia
Moscow, Russia: Landau Institute 193.233.9.7: ntp.landau.ac.ru Service area: Russia
for Theoretical Physics
Russia: Pushchino (near Moscow) 194.149.67.130: ntp.psn.ru Service area: Russia
Chernogolovka, Russia: 193.233.46.10: sign.chg.ru Russia
Chernogolovka Scientific Center
(near Moscow)
Far East and Pacific Rim
Tokyo, Japan: Cyber Fleet, Inc. 203.139.30.195: ntp.cyber-fleet.net Japan and East Asia
Seoul, Korea: Inet, Inc. time.nuri.net Korea, Japan, Hong Kong,
and East Asia
Hamilton, New Zealand: The truechimer.waikato.ac.nz; New Zealand
University of Waikato truechimer1.waikato.ac.nz;
truechimer2.waikato.ac.nz;
[128/129]
truechimer3.waikato.ac.nz
Singapore and the Philippines ntp.shim.org Singapore
Africa
Lagos, Nigeria: Supernet300 ntp.supernet300.com Western Africa (primarily
Nigerian NITEL
backbone)
Pietermaritzburg, South Africa: 143.128.82.200: ntp.cs.unp.ac.za South Africa
Natal University
Australia
Adelaide, South Australia: The 129.127.28.4: AARNet
University of Adelaide augean.eleceng.adelaide.edu.au
129.127.40.3: ntp.adelaide.edu.au
203.21.37.18: ntp.saard.net
Carlton, Victoria, Australia: eSec 203.21.84.4: time.esec.com.au
Limited
[129/129]

YV2 Administration

Uploaded by

Copyright:

Available Formats

YV2 Administration

Uploaded by

Document Information

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

YV2 Administration

Uploaded by

Copyright:

Available Formats

YV2 Administration Jan.

3.2.1.2.3 Call History......................................................................................................28

6.2.4 DNS Server....................................................................................................................60

7.2.1.2.3 qop Support......................................................................................................78

1.2.4. ‘A’ Rule for Suffix Appending.....................................................................................122

4x21 LCD with back-light Context-Sensitive

【Context-sensitive Soft-Keys】: 4 context-sensitive soft-keys

【→】：Move right (per characters)

Del Abc../123.. Clear Back

Call Del Back

1.1.1.1 View Contact

1.1.1.1.1 Edit Contact.Display

Del Abc../123.. Clear Back

 [Del]: Delete one character.

1.1.1.1.2 Edit Contct.Protocol

Call tel/sip Back

1.1.1.1.3 Edit Contact.Email-Like Address

Del Abc../123.. Clear Back

 [Del]: Delete one character.

 [Save]: Add contact into address book and return.

1.2.1 Add Contact.Display

Del Abc../123.. Clear Back

 [Del]: Delete one character.

1.2.2 Add Contact.Protocol

 [Save]: Add contact into address book and return.

1.2.3 Add Contact.Email-Like Address

Del Abc../123.. Clear Back

 [Del]: Delete one character.

 【OK】: Edit selected contact.

1.4 Address Book.【OK】

1.4.1 Speed Dials

1.4.2 Call Screen

1. List all contacts in black lsit in alphanumeric order.

Call Del Back

 [Back]: Return without any changes.

i. “Do-Not-Disturb”, DND, feature is enabled

 Shared-Line Appearance (SLA):

2.1 Missed/Dialed/Received Calls

2.1.1 View Record

3.2 Programable Keys

3.2.1 Programmable Keys.【OK】

DTMF List Format:

3.2.1.1.1 View DTMF

Dial Save Back

1. Tag: User-friendily display associated with this DTMF sequence numbers.

3.2.1.1.1.1 Edit DTMF

Del Abc../123.. Clear Back

 [Del]: Delete one character.

3.2.1.2 Set Programmable Keys

 [Back]: Return without any changes.

3.2.1.2.1 Address Book

3.2.1.2.3 Call History

3.2.1.2.4 Call Detail

o From/To: SIP address-of-record of the peer.

3.2.1.2.5 Call Return

3.2.1.2.6 Channel Info

3.2.1.2.7 Do Not Disturb (DND)

3.2.1.2.8 DTMF List

3.2.1.2.9 DTMF Relay by INFO

3.2.1.2.15 Network Status