AirLive EPHONE-2000S User manual

AirLiveePhone-2000S EnglishVersion
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User’sManual
ePhone-2000S
Wired IP Phone

AirLiveePhone-2000S EnglishVersion
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AirLive ePhone-2000S
<English Version>
CONTENTS
I. Airlive ePhone-2000S Diagram...................................................................................................1
1. Front View ..................................................................................................................................1
2. Rear View ...................................................................................................................................1
3. Back View ...................................................................................................................................1
II. Key Definition...........................................................................................................................2
III. Basic Configuration ................................................................................................................. 4
1. How to activate 『 Main Menu 』 ?............................................................................................4
2. Network Settings ........................................................................................................................4
(1) DHCP ............................................................................................................................. 4
(2) Static IP Environment（Fixed IP）............................................................................... 4
(3) PPPoE ............................................................................................................................ 4
3. SIP Service .................................................................................................................................5
IV. Phone Calls ............................................................................................................................... 7
1. How to dial out?.......................................................................................................................7
2. Dialing Method ..........................................................................................................................7
(1) Address book..................................................................................................................7
(2) Call History....................................................................................................................8
(3) ENUM Dialing...............................................................................................................8
(4) Intra-Domain Dialing..................................................................................................... 8
(5) Contact Dialing ..............................................................................................................8
(6) IP Dialing .......................................................................................................................8
(7) LAN Dialing ..................................................................................................................8
3. Add, Search, Delete and Modify Address Book.......................................................................... 9
(1) Add new address ............................................................................................................ 9
(2) Search.............................................................................................................................9
(3) Delete .............................................................................................................................9
(4) Modify............................................................................................................................ 9
4. Speed Dial.................................................................................................................................. 9
(1) Add speed dial number ................................................................................................10

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(2) Remove a specific speed dial entry..............................................................................10
(3) Speed Dial....................................................................................................................10
5. Call Screen ...............................................................................................................................10
(1) How to activate and deactivate call screen?........................................................................ 10
(2) List and Revoke Screened Numbers ..................................................................................... 10
V. Call Forward ..............................................................................................................................10
1. Set up forwarding number........................................................................................................11
2. Enable Call Forwarding Features ........................................................................................... 11
(1) All Calls Forward (Unconditionally Forward)............................................................. 11
(2) Busy Forward............................................................................................................... 11
(3) No Answer Forward ..................................................................................................... 11
3. Do Not Disturb (DND)..............................................................................................................12
VI. Phone Settings ........................................................................................................................13
1. Password:.................................................................................................................................13
2. DSS features............................................................................................................................. 13
3. Date/Time ................................................................................................................................. 14
4. Ringer.......................................................................................................................................14
5. Lock..........................................................................................................................................14
VII. NAT and Firewall................................................................................................................... 14
1. To determine whether you resides on public internet or local area network........................... 14
2. Public Internet Configuration..................................................................................................15
3. LAN Configuration to Traverse NAT/Firewall......................................................................... 15
3. Option-1: Contact your network administrator to configure your NAT/Firewall route. ......... 15
4. Option-2 Activate STUN ..........................................................................................................16
(1) To determine whether you are currently NATed or behind a firewall.......................... 17
(2) Activate STUN Mode .................................................................................................. 17
VIII. Specify CODEC..................................................................................................................17
IX. Advanced Call Processing .....................................................................................................18
1. Mute .........................................................................................................................................18
2. Hold and Retrieve .................................................................................................................... 18

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3. Handset, Speaker-phone, Ear-phone and Loud-speaker (Open-listening).............................. 18
4. Transfer .................................................................................................................................... 19
(1) Consultative transfer ........................................................................................................19
(2) Blind Transfer .................................................................................................................. 19
5. Conference ...............................................................................................................................20
(1) When you have set up two connected calls......................................................................20
(2) When you have only one connected call..........................................................................20
6. Auto Answer .............................................................................................................................21
(1) Server-side invoked auto-answer for urgent calls: ................................................................. 21
(2) Phone-set locally configured to auto-answer all incoming calls unconditionally. ................. 21
(3) Auto-Answer incoming calls arrive on specific registered SIP address-of-records............... 22

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I. Airlive ePhone-2000S Diagram
1. Front View
2. Rear View
3. Back View
10/100 RJ-45 Ethernet
Port (to PC)
10/100 RJ-45 Ethernet
Port (to LAN)
Power jack
Reset
Ear-phone Jack
(RJ-11)
Handset Jack
(RJ-11)
Wall Mount
2x16 LCD
Microphone
Handset
Speaker
Keypad

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II. Key Definition
【Name】：Functions in Call Mode | Functions in Menu Mode
【A/B Channel】：Call lines (2 concurrent calls at most) / Review the calling information on this
channel during conversation.
【Service Realm】：Display the registration status of each active service domain on idle; switch
target service domain (ISP) while making calls.
【Reject】：Reject incoming waiting calls
【MWI】：Message Waiting Indication, MWI: Access to voice mail system.
【MUTE】： Mute | Delete character
【FUNC】：Menu | Return to upper level submenu
【SPK】：Hands-free | Exit menu
【HOLD】：Hold | Confirm, Save
【SPD】：Speed Dial
【REDIAL】：Redial the last dialed number.
【Flash】：Take back transferring calls/Exit Conference mode
【XFER】：Transfer
【＋】：Volume UP (Ringer, headset, handset, speaker) | Next item 、 Move cursor to right
【－】：Volume Down (Ringer, headset, handset, speaker) | Previous item 、 Move cursor to left
FUNC
MUTE
SPK/Hands-free
HOLD
SPD
XFER
REDIAL
MWI
Volume【−,＋】
A/B Channel
Reject
FLASH
Service Realm

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【Registration】：Re-register. The LED indicates the registration status of each active service
domain:
Green LED On: Successfully register to all activated service domains.
Red LED On: At least one activated service domain could not be registered.
Green LED Flashes: Registration is in progress. Note, IP SIP Phone will regularly refresh
SIP Address-of-Record registration as necessary.
Red LED Flashes: None of those activated service domains can be successfully registered.
LED is off: Users explicitly logs out all SIP service and goes off-line ‘till user presses the
【Registration】key again to go on-line (re-register to all SIP service).
【Auto-redial】：Auto-redial the last dialed number ‘till connected (ringing). This feature can be
activated when busy tone is heard or as a shorthand of【Redial】+【Auto-redial】.
Once activated, the corresponding LED will be on.
【DND】：Do Not Disturb (red LED indicates on)
【Forward】：Forward incoming waiting calls
【URL】：Use keypad to enter alphabets and numbers (red LED on).
【Address Book】：Access to address book (search an entry or list all entries).
【Call History】：Missed calls/Received calls/Dialed Number. If the red LED is flashing, it indicates
there are unread records of missed calls.
【Conference】：3-party local conferencing
Conference
Call History
DND
Forward
Auto-redial Address Book
Registration URL

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III. Basic Configuration
1. How to activate 『Main Menu』?
1.Press 【FUNC】+ 【#】to configure via keypad。
2.After configuring the IP and network mask of your terminal, you may point any modern web
browser from you PC to configure it by HTTP. The default login password for both
privileged and user-level account is “0000”.
3.TELNETinto the phone on TCP port 23. The login password is same as phone lock
password, which default is “0000”.
4.Initially, your phone can only be configured via keypad since it bears no valid IP yet. After
finishing configuring your network, you can use either a web browser (HTTP port 80) or a
TELNETclient (TCP port 23) to configure your phone.
2. Network Settings
Please go to 『Main Menu』=>【6.Network】/【1.General】to configure your phone based
on your network environment: DHCP, static IP or PPPoE.
(1) DHCP
zPick【1.Mode】/【1.DHCP】
zDisable【4.Use Static DNS】by choosing 【2.DHCP】
1. If you want to assign a different domain name server instead of using those
obtained by DHCP, you should choose【1.Static DNS】and set the IP of your
specific DNS into 【6.Network】/【1.General】/【5.Static DNS】, such as “1.Primary
DNS” = 192.168.3.254*
(2) Static IP Environment（Fixed IP）
zPick【1.Mode】/【2.Static Assign】
zGo to【2.Static Settings】, and enter your network configurations based on your ISP.
For example:
1. Host IP = 210.201.210.132*
2. Network mask = 255.255.255.0*
3. Gateway IP = 210.201.210.128*
zEnable【4.Use Static DNS】by choosing 【1.Static DNS】
zAssign【5.Static DNS】, such as: “1.Primary DNS = 168.95.1.1*”
(3) PPPoE
zPick【1.Mode】/【1.DHCP】
zGo to【3.PPPoE Settings】and enter your PPPoE authentication information, such as:
1. Login ID = MyPPPoEAccount*

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2. Password = PPPoEDialupPassword*
3. Service Name = Optional, some ISPs may require it.
【*】Value should be adjusted to your real network environment
Note: if you reside on a LAN without gateway, you should specify the gateway IP as “0.0.0.0”
rather than assigning a non-existent or an invalid IP.
3. SIP Service
(1) Let’s assume that you have obtained the following information from your SIP ISP:
zAccount: Michael
zPassword: secret
zSIP AoR: [email protected]
zSIP Proxy/Registrar Server: sip.SIP.com, which serves on UDP port 5060
(2) Point web browser to your terminal and enter the following information on 『2.SIP
Settings』/『1st Domain』page.
(3) Alternatively, you may configure it via either keypad or TELNET by going to『Main

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Menu』=>『6.Network』/『2.SIP settings』/『2.Service domain』/『1.1st Realm』
zActivation: Enabled
zAuthentication: Note: 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).
1. Server login: Michael
2. Server passwd: secret
zAddress of record:
1. Display Name: Michael Jr. (Could be any combination of alphabets and numbers)
2. Protocol: sip
3. Sip:[email protected]
zProxy server:
1. Configuration: 2.Use default
2. Default proxy:
1. Proxy FQDN: sip.SIP.com
2. Transport: 1.UDP (or 3.TCP and UDP)
3. UDP port: 5060 （SIP proxy listening UDP port of your ISP.）
4. TCP port: 5060 (leave default intact)
zRegistrar
1. Configuration: 1.Same as proxy
2. Registrar svr: leave default intact since the registrar and proxy server provided by
your ISP are the same
(4) If you have applied for more than one service domains, please repeat step (1) and (2) until
all active domains are properly configured. IP SIP Phone supports three different active
service domains at most.
(5) Press 【SPK】to exit the configuration menu and the system will try to register to those
activated SIP domains. The flashing green LED of 【Registration】key indicates that the
registration is undergoing. Once the green LED stops flashing, you could know the
registration result by the LED:
Green LED On: Successfully register to all activated service domains.
Red LED On: At least one activated service domain could not be registered.
Green LED Flashes: Registration is in progress. Note, IP SIP Phone will regularly
refresh its SIP AoR registration as necessary.
Red LED Flashes: None of those activated service domains can be successfully
registered.
LED is off: Users explicitly logs out all SIP service and goes off-line ‘till user presses
the 【Registration】key again to go on-line (re-register to all SIP service).
Or press【Service Realm】to show registration status of each active service domain:

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0 : s i
p
. S I P . c o m
0 :
f
w d .
p
u l v e r . c o m
X
: u n k n o w n . c o m
0: Registered domain
X: Registration failed domain (Retry after 90 seconds).
IV. Phone Calls
Press【SPK】to switch between hands-free and handset mode. However, if you switch to
hands-free mode with handset lifted (off-hooked), the phone will be in loud-speaker mode:
it continues transmitting voice from handset, not from microphone, and played voice to both
speaker and handset receiver simultaneously. It will help clarify the voice of talker while
letting all persons in the same room listen. The phone will activate microphone for voice
transmission only when it is on hands-free mode and the handset is on hook. This applies to
ear-phone mode as well. In loud speaker mode with ear-phone enabled, the ear-phone could
hear voice but not speak-able.
1. How to dial out?
f
w d .
p
u l v e r . c o m
3 2 0 0
By default, you can gain access to active Service-Domain-N by line-N whenever
applicable. That is, the default service domain of [B call] is 2nd active service domain
if available; otherwise, it will use the 1st active service domain as default.
Moreover, you could change the target service domain while making calls by pressing
【Service Realm】to circle those active service domains (no matter whether they have
registered or not), but must before dialing the ‘#’, which finishes the dialing. The active
service domain will appear on the upper-right corner.
2. Dialing Method
In dialing phase, you could use【MUTE】to correct typos.
(1) Address book
z『Main Menu』=>『1.Address Book』/『1.Search』(By default【F4】is programmed
to access address book).
zPick the target number and press【REDIAL】to dial out.
Sip.Sip.com
fwd.pulver.com
Auto-locate

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(2) Call History
zPress【Call History】to choose desired call lists:Missed/Received/Dailed numbers.
zTraverse the list and pick up the target entry.
zPress 【REDIAL】to dial out.
Note: if you press 【Call History】while its red LED is flashing (indicating there are
unviewed missed calls), it will enter the “Missed calls” list directly.
(3) ENUM Dialing
zStarting with【#】for ENUM dialing
zDial the target number (6 digits at least), ending with a【#】to speed up dialing phase.
E.g.: to call “tel:[phone]”, please dial “#88635295000#”。
(4) Intra-Domain Dialing
zEnter the user part of the target AoR or dial alias number directly.
zThe target SIP service domain could be circle by press 【Service Domain】.
zPress 【＃】to finishing dialing.
Intra-domain dialing is applicable if both party registered to the same SIP domain
(possibly to the same SIP proxy/registrar server)。E.g., if sip:[email protected]
wants to place a call to sip:[email protected], he could just dial 【3200＃】 when
“Sip.com” has selected as target service domain.
(5) Contact Dialing
zUse 【*】for 『@』and『.』
zUse 【**】for 『:』if the target SIP agent does not listen on UDP port 5060.
zPress【＃】to finishing dialing
E.g. 1: to call sip:[email protected], please dial “3200＊192＊168＊10＊200”.
E.g. 2: to call sip:[email protected]:5070, please dial
”3200＊192＊168＊10＊200**5070”.
(6) IP Dialing
zUse【*】for 『@』and『.』
zUse【**】for “:” if the target SIP agent does not listen on UDP port 5060.
zDial【#】to finish dialing.
E.g. 1: to call sip:192.168.10.200, please dial “192＊168＊10＊200”.
E.g. 2: to call sip:192.168.10.200:5070, please dial ”192＊168＊10＊200**5070”.
(7) LAN Dialing
(Applicable only while both parties locate on the same LAN and their IP prefix, 24-bits,
are the same).
zStarts with a【*】
zDial the last suffix of target IP (the number must be between 1 and 255, inclusive).
zUse 【**】for “:” if the target SIP agent does not listen on UDP port 5060.
E.g., if “192.168.10.200” wants to dial “192.168.10.31”, please dial 【*31#】or

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【*031#】。
※ Note: 【*69】has been reserved for call return feature, please dial 【*069】
if you want to call a IP of “xxx.xxx.xxx.69”.
3. Add, Search, Delete and Modify Address Book
You may point web browser to your terminal’s IP and go to 『Address Book』to configure it.
Below illustrates how to configure address book by keypad or TELNET.
(1) Add new address
z『Main Menu』=>『1.Address Book』/『2.Add Entry』
zEnter Display Nam, such as Michael
zPick address type, default is SIP.
zEnter SIP AoR or ENUM number without protocol field, such as:
i. sip: 211.21.189.62
ii. sip: [email protected]
iii. tel: [phone]
(2) Search
z『Main Menu』=>『1.Address Book』/『1.Search』
zEnter the prefix to search or press 【Hold】to list all addresses
(3) Delete
Option 1:
z『Main Menu』=>『1.Address Book』/『1.Search』
zSearch for the address to remove and press【HOLD】to view the item.
zPick『2.Erase』from this address and press【HOLD】to delete it.
Option 2:
z『Main Menu』=>『1.Address Book』/【3.Erase】
zChoose either 『1.One by One』or『2.Erase All』to clear address book.
(4) Modify
z『Main Menu』=>『1.Address book』/『1.Search』
zSearch for the address you want to modify then press【HOLD】to view
zChoose 『1.Modify』from the menu of this address and modify it
4. Speed Dial
You may point web browser to your terminal’s IP and go to 『Address Book』/『Speed Dials』
to configure it. Below illustrates how to configure speed dials by keypad or TELNET.

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(1) Add speed dial number
z『Main Menu』=>『1.Address Book』/『4.Speed Dials』
zGo to the digit you want to set for speed dial and press 【HOLD】
zPick the target number from address book to associate with this speed dial digit.
Press 【HOLD】to save.
(2) Remove a specific speed dial entry
z『Main Menu』=>『1.Address Book』\『4.Speed Dials』
zGo to the digit you want to set for speed dial.
zPress 【MUTE】to remove the current speed dial mapping.
(3) Speed Dial
z[0-19] +【SPD】; e.g.: ‘1’+【SPD】or ‘01’ + 【SPD】
z【SPD】+ [00-19]; e.g.:【SPD】+ ‘01’.
Note:if you want to dial single digit only, such as
【 SPD 】 + 【 1 】 , please press an
additional
【 # 】 to finish dialing; otherwise, it will wait for the second digit until
inter-digit timeout, which default is 4 seconds. For example, 【 SPD 】 + 【 1 】 + 【 # 】 .
5. Call Screen
You may point web browser to your terminal’s IP and go to 『Address Book』/『Call Screen』
to configure it. Below illustrates how to configure it by keypad or TELNET.
(1) How to activate and deactivate call screen?
z『Main Menu』=>『1.Address Book』/『1.Search』
zLocate the target address that you want to screen and press 【HOLD】to choose
zGo to『3.Screen Call』and press【HOLD】to toggle call screen.
(2) List and Revoke Screened Numbers
z『Main Menu』=>『1.Address Book』/『5.Call Screening』to view all screened
addresses.
zLocate the target address that you want to remove from screening list and press
【HOLD】to choose.
zPick 『3.Revoke』and press 【HOLD】to remove it from the screened list.
V. Call Forward
You may point web browser to your terminal’s IP and go to 『Call Forward』to configure it.
Below illustrates how to configure call forward by keypad or TELNET.

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1. Set up forwarding number
z『Main Menu』=>『4.Call Forward』/『1. Target Number』
zPress 【HOLD】to pick an entry from address book. Use 【Õ】and 【Ö】to view
the current forwarding number. Press 【MUTE】to remove the current mapping of
forward number.
Note: on incoming calls arrival, both red LED of 【Reject】and【Forward】function keys will
be on. If you press 【Reject】on which time, the incoming call will be rejected as busy and
recorded as a received call, but it will not be forwarded in anyway. If you would like to
transfer it to your configured forwarding number, you should press【Forward】. If you have
not configured the forwarding number, the incoming call will be rejected as busy and
recorded as a missed call.
2. Enable Call Forwarding Features
(1) All Calls Forward (Unconditionally Forward)
z『Main Menu』=>『4.Call Forward』/ 『1.All Calls Fwd』
zPick『Enabled』and press【HOLD】to save your setting.
zOnce this feature is enabled, the red LED of【Forward】function key will flash to
remind you that all incoming calls will be silently forwarded to the forwarding
number you previously configured (no ringing tone would ever be heard!). If you
have not configured the forwarding number, the peer will be replied as ”480
Temporarily unavailable” and recorded as a missed call.
(2) Busy Forward
z『Main Menu』=>『4.Call Forward』/『3.Busy Forward』
zPick『Enabled』and press【HOLD】to save your setting.
zOnce this feature is enabled, the system will start to forward incoming calls to the
forwarding number you previously configured when the system is busy (on which
time you have already had 2 calls are waiting). If you have not configured such
forwarding number, it will reply the peer as busy and recorded as a missed call.
(3) No Answer Forward
z『Main Menu』=>『4.Call Forward』/ 『4.No Answer Fwd』/ 『1.Status』and

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pick 『Enabled』, press【HOLD】to save it.
z『Main Menu』=>『4.Call Forward』/ 『4.No Answer Fwd』/ 『2.No Answer
Time』, and configure the ringing expiration time before forwarding the incoming
call (system default is ringing for 30 seconds) . Press【HOLD】to save your
setting.
zOnce “forward on no-answer” feature is enabled, if you do not answer the
incoming call before the timer expires, it will be forwarded to the forwarding
number you previously configured. If you have not configured the forwarding
number, the peer will be replied as “no answer” and recorded as a missed call.
3. Do Not Disturb (DND)
zPress【DND】function key to toggle this feature: if the red LED is on, this feature
is enabled; otherwise it is disabled.
zWhen this feature is on, all incoming calls will be silently (no ringing tone will
ever be heard) forwarded to the forwarding number you previously configured. If
you have not configured the forwarding number, the peer will be replied as ”480
Temporarily unavailable” and recorded as a missed call.

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VI. Phone Settings
You may point web browser to your terminal’s IP and go to 『Phone Settings』to configure it.
Alternatively, you may configure your terminal via keypad or TELNET by going to『Main
Menu』=>『3. Phone Setting』. Below illustrate the keypad/TELNE configuration steps.
1. Password:
(needed on web configuration, telneting into the phone and unlocking the phone)
zEnter your old password then press【HOLD】(Default password is “0000”）
zEnter your new password then press【HOLD】(Accept only digits 0~9)
zRe-type your new password to confirm, then press【HOLD】to finish.
Web Page: 『 Phone Settings 』 / 『 Account 』 .
2. DSS features
Web Page: 『 Phone Settings 』 / 『 DSS Features 』 .
User programmable keys for easy access to various phone features.
zPick the function keys you want to program for short-cut key then press【HOLD】.
For example, you may program some hot-lines for one-touch dial out.
zBrowse the available functions, press【HOLD】to confirm.
The default mapping of these programmable function keys are:
【F1】:Forward menu – shortcut to activate forward menu.
【F2】: Channel info–show information of the last call on each call channel【A / B】.
【F3】: Call Detail - show detail records of the latest three connected and finished calls.
【F4】: Speed dials – activate speed dial list.
【 F5 】 : Messaging – Access to out-of-dialog instant messaging.
【F6】: Network Speed - adjust the voice quality based on network link speed.
【F7】: Network Info - show the current active host IP, MAC address and DNS IP(s).
DSS Keys

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【F8】: Call Return - place a call to the last incoming call, either a missed or received one
3. Date/Time
Web Page: 『 Phone Settings 』 / 『 Date / Time 』 .
zChoose the item and press【HOLD】to configure:
1. Date: System date
2. Time: System time
3. Date format: date displaying format
4. Time zone (default is GMT+00:00）
Use digits key【0】~【9】to switch between +/-
5. DST settings (day-light saving time, default is 0:00)
Note: if you want to keep the date and time you manually set, you must disable SNTP
synchronization from 7.3.1-SNTP Mode; otherwise, it will synchronize system
time with assigned NTP server.
4. Ringer
1. Ringer volume: (Adjust the ringing volume on incoming calls arrival).
z【＋】and【－】to increase or decrease ringing volume.
2. Ringer type: (Pick your favorite ringing tone)
z【＋】 and【－】to listen your favorite ringing, press 【HOLD】to save your
setting.
5. Lock
z『Main Menu』=>『3.Phone settings』/『8. Lock』
z『1.=Y』to lock out the phone such that it could not make outbound calls (but still could
take incoming calls, or『2.=N』to cancel this operation and press【HOLD】to confirm.
After locking out the phone, to unlock it, you need the phone password.
VII. NAT and Firewall
If this phone locates within a local area network and you want to place a call to public internet,
you must configure your phone to traverse the NAT and firewall currently behind. Please refer to
section-13 “NAT Traversal” on “IP SIP Phone v2 User’s Guide” for detail, and below illustrates
how to configure your terminal by keypad or TELNET.
1. To determine whether you resides on public internet or local area network
z『Main Menu』=>『8.Advanced』/『4.System status』or press 【F7】(default is

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“Network Info”).
zGo to 『1.Network』and use【＋】and【－】to navigate active network configuration:
host IP, MAC address, active primary DNS IP and optional secondary DNS IP (In that
order).
zIf your host IP is within any of the listed ranges, then your IP SIP Phone resides on
LAN, otherwise, it locates on public internet.
i. 10.0.0.0 ~ 10.255.255.255 (10.xxx.xxx.xxx).
ii. 172.16.0.0 ~ 172.31.255.255 (172.16.xxx.xxx ~ 172.31.xxx.xxx).
iii. 192.168.0.0 ~ 192.168.255.255 (192.168.xxx.xxx).
Note: if your host IP is 127.0.0.1, you should reconfigure your network to acquire a valid IP.
2. Public Internet Configuration
z『Main Menu』=>『6.Network』/『4.NAT & firewall』/『2.UDP traversal』
zEnable “1.Full access” by press 【HOLD】
3. LAN Configuration to Traverse NAT/Firewall
There are basically 2 options for CPE to traverse NAT/Firewall:
(a) Option 1: Set up a static route in the NAT gateway (Recommend)。
(b) Option 2: Use STUN to measure out ports.
Please adopt suitable option based on your network configuration.
Note: some SIP ISPs may provide SIP-aware router (NAT/Firewall) for their customers.
When you use a SIP-aware router, NAT detection should be set to "Off" as if you
were on the public internet and the configuration is the same as “Public Internet
Configuration”: 『6.Network』/『4.NAT & firewall』/『2.UDP traversal』/『1.Full
access』, which is to say that you do not have to worry about the NAT, nor firewall.
3. Option-1: Contact your network administrator to configure your NAT/Firewall route.
zContact your network administrator and acquire at least 7 consecutive UDP ports
mapped from the NAT to your IP SIP Phone.
For example:
i. NAT IP = 210.95.168.1 (you may detect your NAT IP by STUN from 『Main
Menu』=>『6.Network』/『4.NAT & firewall』/ 『1. STUN』/ 『2. Diagnose
NAT』)
ii. The network administrator has mapped UDP port 10101~10107 from NAT to
your IP SIP Phone [phone].
Note 1: Since the network administrator has to configure the NAT/firewall to map
those UDP ports to your IP SIP Phone, thus you should use static IP instead of
DHCP as your network configuration. Otherwise you take the risk that the IP
SIP Phone would get a different IP from the currently set into the mapping
when it reboot and re-get its IP by DHCP.

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Note 2: If there are several phones reside under the same NAT, their NAT port
mappings must not be overlapped since they all share the same NAT resource!
zConfigure SIP service signaling port:『Main Menu』=>『6.Network』/『2.SIP settings』
/『1.SIP service』. Take the scenario above as an example:
1. Transport: 1 (or 3, you must include UDP anyway).
2. SIP port: 10101
zConfigure RTP ports by going to 『Main Menu』=>『6.Network』/『3 RTP settings』.
Take the mapped ports as an example:
1. Base: 10102 (Must be an even number and between 3000 and 65532)
2. Range: 6 (Must be an even number and larger than or equal to 4).
Note: RTP ports are used for voice media transmission. Those ports must be
consecutive and start with an even number. Besides, at least 4 ports are needed
for a phone equips with a capacity of two concurrent calls, where each channel
needs 2 ports for RTP/RTCP.
zAssign static NAT IP by going to『Main Menu』=>『6.Network』/『4.NAT & firewall』
/『3.Static NAT IP』and fill in the acquired NAT IP from network administrator, such
as 210.95.168.1 mentioned above (you may also detect your NAT IP by STUN
from 『Main Menu』=>『6.Network』/『4.NAT & firewall』/ 『1. STUN』/ 『2.
Diagnose NAT』).
Note: if your NAT equipped with no fixed IP, such as those NATs dial into WAN by
PPPoE, then you must synchronize the NAT IP currently set into IP SIP Phone
whenever your NAT IP changed (Such as NAT re-dialup). By default, the
system will auto-detect the NAT IP (if the STUN server configured into
menu-6.4.1.1『STUN server』is viable) and notify you by sending an instant
message whenever it determine a mismatch between the NAT you configured
into menu 6.4.3-『Static NAT IP』and the one it auto-discovered by STUN
protocol. Alternatively, you may go to『Main Menu』=>『6.Network』/『4.NAT
& firewall』/『4.Dynamic NAT IP』and enable『2.Update by STUN』to
auto-refresh the newly changed NAT IP into menu-6.4.3『Static NAT IP』
whenever it detects an inconsistency.
zEnable Assigned static NAT IP by going to 『Main Menu』=>『6.Network』/『4.NAT
& firewall』/『2.UDP traversal』, selecting『2.Static NAT map』and press【HOLD】
to confirm activation.
4. Option-2 Activate STUN
Setting up the NAT router is impossible in many cases, and new equipment may be too
expensive. For these environments, "Simple Traversal of UDP Through NATs" (STUN)
has come to rescue. To enable STUN, you must acquire the IP of your served STUN
server and diagnose your NAT to see if STUN is viable.