V102
2-Port SIP VoIP Telephone Adaptor
User Manual
V1.10
Quick Guide
Step 1: Broadband (ADSL/Cable Modem) Connections for V102
A. Connect V102 LAN port to ADSL modem as the following connection.
B. Connect V102 PC port to Notebook PC LAN port using a Category 5 LAN cable.
C. Connect V102 RJ11 PHONE1 port to one analog telephone set.
D. Connect V102 RJ11 PHONE2 port to another analog telephone set.
E. Connect Power Adaptor. After power on, the POWER LED will be Green ON.
F. Pick up Phone1 or Phone 2, and the PHONE LED will be Green ON indicating Off-Hook.
G. When the PHONE LED is Green flashing, it indicats a successful SIP registration for Phone
1.
H. When the PHONE LED is Red flashing, it indicats a successful SIP registration for Phone
2.
I. Press #121# and #120# from the phone to listen to IVR and to check the DHCP status and
the IP address (e.g. 192.168.1.100) for V102. After the IP announcement, please hang up.
Figure A. ADSL Connections with NAT Router for V102
Step 2: Settings for V102 from PC Web Browser
A. This is an example for FWD ITSP provider using ADSL connection with NAT router as in
Figure A. Both parties sign up to FWD SIP server http://www.freeworlddialup.com
with
registered FWD phone number and password.
B. Pick up the phone and press #120# from the phone to listen to IVR and get the IP address
(e.g. 192.168.1.100) for V102.
C. Enter the IP address from PC Web browser for configuration settings.
Example: Enter http://192.168.1.100
from IE Web browser to display login page.
D. Enter the user name and password into the blank field. The default settings are Username:
root Password: test . Click the “Login” button to enter for configurations.
i
PC
PHONE1
NAT Router
ADSL Modem
Router IP: 192.168.1.254
V102 IP: 192.168.1.100
LAN
PC IP: 192.168.1.101
PHONE2
INTERNET
E. You need to set up the following web configurations: Network Settings, SIP Settings, NAT
Settings/STUN Settings. Remember to submit, save and reboot for new configurations.
F. The PHONE LED will be Green flashing showing a successful registration in the SIP
server. For further detail configurations, please refer the VoIP applications in the user
manual.
Step 4: Making Point-To-Point SIP Calls
A. Pick up the phone and you should hear a dial tone.
B. Press 123456# to call the party with the number 123456 registered in the SIP server. Note
# is used to send out the call immediately. In a moment, you should hear the ring back tone,
and wait for the called party to answer. For more applications, please refer to the user
manual
.
Note: If you have difficulties in configuring V102, please refer to the last chapter for trouble
shootings.
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TABLE OF CONTENTS
1. Introductions………………………………………………………………
2. Features ………………………………………………………………………
3. Packing Contents ……………………………………………………
4. LED Indicators……………………………………………………………
5. Installations & SIP Configurations ……………………
6. Default Reset from Telephone ……………………………
7. Configurations from Web Browser ……………………
8. Configurations from Telephone & IVR………………
9. V oIP Applications Examples…………………………………
SIP-to-SIP Calling/Answering
……………………………
SIP to Direct IP Calling …………………………………………
Direct IP to Direct IP Calling/Answering ………
Direct IP to Direct IP Calling within NAT…………
3-Way Conference Call, Call Waiting, Hold……
SIP-to-SIP Calling for FWD …………………………………
10. Advanced Settings for Embedded NAT …………
11. Trouble Shooting for Web Configurations ……
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3
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4
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29
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30
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31
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35
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1. Introduction
The V102 is a 2-port FXS Telephone Adaptor (TA) with SIP Protocols for Voice over IP (VoIP)
applications. Connecting to the Internet and two analog telephone sets, the V102 can
supports two concurrent VoIP calls over the Internet. V102 provides Ethernet LAN and PC
ports for ADSL and Notebook PC connections. It also provides two RJ11 connectors for
analog telephones (FXS). The two FXS ports can support T.38 features for FAX over
Internet. With an embedded NAT/DHCP server, V102 can be easily configured to fit for
different network diagrams by PC Web browser and telephone set, and it is very suitable for
ITSP (Internet Telephony Service Providers) and SOHO users to make VoIP calls.
Note that V102 requires an IP address, a subnet mask, and its gateway Router IP address for
its own use to connect to Internet. These three are available from your Internet service
provider. V102 may enable PPPoE or DHCP features to automatically get an assigned
dynamic IP from the ITSP. Please refer to Section 7 Configurations from Web browser for
detailed information.
2. Features
The V102 VoIP TA is equipped with two RJ11 connectors and two RJ45 connectors and is
featuring as the following
¾ SIP v1 (RFC2543), v2 (RFC3261) with MD5 authentication (RFC2069 and RFC 2617)
¾ RJ45 x 2 for Ethernet + RJ11 x 2 for FXS ports
¾ ITU-T G.711, G.723, G.726, G.729A/B, VAD and CNG for Speech Codec
¾ ITU-T G.165/168 Echo Cancellation
¾ Three LED Indicators for V102: POWER, PHONE, LAN
¾ Configurations by Web Browser and Telephone
¾ Embedded NAT/DHCP Server
¾ PPPoE/DHCP Client for Dynamic IP plus NAT, DNS, and DDNS Clients
¾ Support STUN server for NAT Traversal
¾ Interactive Voice Recording (IVR) for telephone IP status
¾ Speed Dial, Call Forward/Waiting/Transfer/Hold, and 3-Way Conference Call features
¾ Remote Firmware Upgraded with HTTP or TFTP server by Web PC
¾ Direct IP/URL Dial without SIP Proxy or Dial number via SIP server
¾ Telephone features: Volume Adjustment, Phone book, Speed Dial, Redial, and Flash
¾ Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info
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3. Packing Contents
Inside the package you should find:
(1) One V102 2-Port SIP TA
(2) One AC to 12VDC/1A Power Adaptor
(3) One User Manual CD
Please check if the packing is damaged or any component is missing. If so, please
contact your distributor.
4. LED Indicators
On the front panel of V102, there are three LED indicators as the following
POWER: “On” indicates the power is normal
PHONE: “Green On” indicates Off-Hook on Phone 1 or Phone 2.
“Green Flashing” indicates a successful SIP registration for Phone 1.
“Red Flashing” indicates a successful SIP registration for Phone 2.
“Yellow Flashing” indicates successful SIP registrations for both Phone 1 & 2.
LAN: “On” indicates the Ethernet Ports are in Connection.
“Flashing” indicates the data activity of Ethernet ports.
2
Power Phone 2
LAN
WAN
Phone 1
5. Installations & SIP Configurations
1. Connect V102 RJ45 LAN port to ADSL Modem using a Category 5 LAN cable.
2. Connect V102 RJ45 PC port to Notebook PC using a Category 5 LAN cable.
3. Connect V102 RJ11 PHONE1 port (next to PC port) to a Plain Old Telephone Set
(POTS).
4. Connect V102 RJ11 PHONE2 port to another Plain Old Telephone Set (POTS).
5. Connect the power adaptor (12VDC) to power on V102, and the POWER LED will be ON.
6. The PHONE LED indicators will be OFF for about 5 seconds and start flashing for 5
times, and remain OFF for VoIP configurations. The LAN LED will be ON when RJ45
WAN ports is connected. Pick up Phone1 or Phone 2, and the PHONE LED will be
Green ON indicating Off-Hook. If you hear a busy tone, please check if the WAN port is
connected.
7. Press #120# from Phone to check the assigned IP address for the V102. The default IP
address is 192.168.1.100. Please refer to Chapter 7, and you may enter this IP address
in IE Web browser for web configurations.
8. Please refer Chapter 9 for VoIP applications examples of SIP registrations, and register
Phone 1 & 2 into your SIP server.
9. The PHONE LED will be Green flashing for Phone 1 successful registration, and Red
flashing for Phone 2 successful registration, respectively. (Yellow flashing for both Phone
1 & 2 successful registrations.)
10. After successful registration into the SIP server, the PHONE LED will be Green flashing.
Pick up the Phone 1, and you should hear a dial tone. Press 123456# to call the party with the
number 123456 registered in the SIP server. Note that # will dial out the number
immediately. Dialing without # will not dial out until the auto dial timer (default=5 seconds)
elapsed. In a moment, you should hear a ring back tone, and wait for answer.
6. Default Reset from Telephone
V102 provides an easy way to reset to factory defaults by using Telephone.
Pick up the phone and press #198# to reset back to factory defaults, and the V102 will enter
into POWER ON cycle. The PHONE LED indicators will be OFF for about 4 seconds and
start flashing for 5 times. The POWER LED then will be lit constantly, and the PHONE LED
will be OFF.
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7. Configurations from Web Browser
You may enter the IP address from PC Web browser to configure V102. For example, enter
http://192.168.1.100
from IE web browser to display login page as follows.
7.1. Please enter the username and password into the blank field. The default settings are:
Username: root
Password: test
7.2. Click the “Login” button will enter the management information page for system setup.
Note that whenever you change the setting in each Web page, please remember to click the
“Submit” button in the page, and click the “Save” button to save into the non-volatile memory
and click the “Reboot” button to activate the new settings.
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System Information
7.3. You will see the system information like firmware version, Codec, etc in this page.
7.4. You may click the button list at the left hand side to configure the V102.
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Phone Book for Speed Dial
7.5. The Phone Book page specifies Speed Dial function.
7.6. For Speed Dial function you can add/delete Speed Dial number up to maximum 10 entries
in Speed Dial Phone List.
7.7. If you need to add a phone number into the Speed Dial list, you need to enter the
position, the name, and the phone number (by URL type). When you finished a new
phone list, just click the “Add Phone” button.
7.8. If you want to delete a phone number, please select the phone number you want to delete
then click “Delete Selected” button.
7.9. If you want to delete all phone numbers, please click “Delete All” button.
7.10. Example: Press 2# on telephone to Speed Dial the phone number 2 immediately.
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Call Settings
7.11. The subpages are as follows; Call Forward, SNTP, Volume, Block Settings, Caller ID,
Auto-Dial Timer, Flash Time (or hook switch), Call Waiting, and T.38 FAX functions.
Call Forward function:
7.12. You can select the forward mode and enter the forward URL.
All Forward: All incoming call will forward to the URL you choose.
Busy Forward: The incoming call will forward to the URL when the callee is busy.
No Answer Forward: The incoming call will forward to the URL when no answer.
Note you have to set the Time Out timer for system to start to forward the call to the
number you choosed. When you finish the setting, please click the “Submit” button.
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SNTP Setting:
7.13. You can setup the primary and second SNTP Server IP Address, to get the date/time
information. You may also set the Time Zone, and how long need to synchronize again.
When you finish the setting, please click the “Submit” button.
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Volume Setting:
7.14. You can setup the Handset Volume, Ringer Volume, and the Handset Gain in this page.
When you finish the setting, please click the “Submit” button. Handset Volume is to set
the volume to hear from the handset. Ringer Volume is to set the ringer volume. Handset
Gain is to set the volume send out to the other side’s handset.
Block Setting:
7.15. You can setup the Block Setting to keep the phone silence. You can choose either Always
Block or a Block period.
7.16. Always Block: All incoming call will be blocked until this feature is disabled.
7.17. Block Period: Set a time period and the phone will be blocked during the time period. If
the time in “From” is greater than that in “To” time, the Block time will be from Day 1 to
Day 2.
7.18. After you finish the setting, please click the “Submit” button.
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Caller ID Setting:
7.19. You may show caller ID in your PSTN Phone or IP Phone by selecting “Yes” in Single
Caller ID, and the desired Caller ID option for either FSK or DTMF. When you finish the
setting, please click the “Submit” button.
Auto Dial Setting:
7.20. You can set the timer for inter dial digit in this page. When the timer expires after finished
dialing, V102 will dial out the call automatically. When you finish the setting, please click
the “Submit” button.
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Flash Time Setting:
7.21. You can set the flash time duration for the telephone flash key or hook switch in this page.
The telephone flash key is used to switch to the other phone line or HOLD, and is quite
useful for the 3-way conference call and the call waiting function. When you finish the
setting, please click the “Submit” button.
Call Waiting Setting:
7.22. You can enable the call waiting function in this page. It allows answering another coming
call by pressing flash key while holding the current call. You may switch back to previous
call by pressing flash key again. When you finish the setting, please click the “Submit”
button.
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T.38 (FAX) Setting:
7.23. The T.38 FAX over IP function allows FAX machine to communicate with another FAX
machine over IP network. The defaults are OFF for Phone1 and Phone2. The T.38 ports
of Phone1 and Phone2 must be differents. When you finish the setting, please click the
“Submit” button.
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Network
7.24. You may configure the Bridge, NAT, and DDNS settings to show the Network status in this
section. If you have an external NAT router, then you must select Bridge ON in the Bridge
settings to disable embedded NAT. In this case, the two Fast Ethernet ports will be
bridged and transparent. Otherwise, you must select Bridge OFF to enable embedded
NAT and go on NAT/DDNS settings. The network status will show either bridge mode or
NAT mode depending on the selection of Bridge ON/OFF. The default is Bridge ON.
Network Status:
7.25. You can check and show the current Network settings in this page.
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Bridge Settings:
7.26. You can configure Bridge settings for V102 in this page.
7.27. The IP type for V102 is Fixed IP (192.168.1.100) at default. You may select a proper IP
type for your network requirements.
7.28. The Bridge mode can be ON/OFF. If you select the Bridge ON mode, then the two Fast
Ethernet ports will be bridged and transparent.
7.29. After you finish the setting, please click the “Submit” button.
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NAT Settings:
7.30. When you select Bridge OFF, the embedded NAT will be enabled. This is useful for ADSL
users without NAT router.
7.31. Enable the embedded DHCP Server at the LAN setting to automatically obtain a private
IP address for PC from the embedded DHCP server.
7.32. You may select PPPoE function at the WAN setting, and enter the given username and
password for ADSL. Note that the MAC settings for LAN and WAN must be different from
each other.
7.33. After you finish the setting, please click the “Submit” button.
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DDNS Setting:
7.34. You need to have a DDNS account before configuring the DDNS setting. Usually, most of
the VoIP applications are working with a SIP Proxy Server. Nonetheless, you may have a
DDNS account with a public IP address, and others can call you via the DDNS account.
When you finish the setting, please click the Submit button.
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SIP Settings
7.35. You can setup the Service Domain, Port Settngs, Codec Settings, RTP Setting, RPort
Setting and Other Settings for SIP Proxy Server registrations in this page.
Service Domain Settings:
7.36. You may register up to three SIP accounts in the V102. You can call your friends via firstly
enabled SIP account and receive the phone calls from all the three SIP accounts.
7.37. Click “Active” ON to enable the Service Domain, then enter the following items:
7.38. Display Name: enter the name you want to display.
7.39. User Name: enter the User Name given by your ITSP.
7.40. Register Name: enter the Register Name given by your ITSP.
7.41. Register Password: enter the Register Password given by your ITSP.
7.42. Domain Server: enter the Domain Server given by your ITSP.
7.43. Proxy Server: enter the Proxy Server given by your ITSP.
7.44. Outbound Proxy: enter the Outbound Proxy of ITSP. If not provided, you may skip
this.
7.45. Register Period: enter the Register Period in minute given by your ITSP.
7.46. When it shows “Registered” in the Register Status, it indicates a successful registration
to the ITSP, and the “PHONE” LED will start flashing. The V102 is then ready for VoIP
call.
7.47. If you have more than one SIP account, please follow the steps to register to other ITSPs.
7.48. After you finish the setting, please click the “Submit” button.
DTMF Settings:
7.49. You can setup the options for DTMF function in this page. The options include RFC2833
(Outband DTMF), Inband DTMF, and Send DTMF SIP info. The default is set at Inband
DTMF. If you are making two-stage callings for extension to PSTN, you might need to
select Outband DTMF option.
Port Settings:
7.50. You can setup the SIP and RTP port number in this page. Each ITSP provider might have
different SIP/RTP port setting, please refer to the ITSP to setup the port number correctly.
When you finished the setting, please click the “Submit” button. The defaults for SIP port
and RTP port are 5060 and 60000, respectively.
STUN Settings:
7.51. The STUN function must be enabled to work properly behind NAT when registered in SIP
server. You may enter the STUN server IP address and the STUN port number as shown
in the following example.
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Codec Settings:
7.52. You can setup the Codec priority, RTP packet length, and VAD function in this page. You
need to follow the ITSP recommendations to setup these items.
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Codec ID Settings:
7.53. You can setup the Codec ID in this page. You need to follow the ITSP suggestion to setup
these items.
Other Settings:
7.54. You can setup the Hold by RFC and QoS in this page. To change these settings please
follows your ITSP information. When you finished the setting, please click the Submit
button. The QoS is used to set the voice packet priority. Higher value other than zero will
get higher priority for the voice packets in Internet. However, the QoS function still needs
to cooperate with the other Internet devices.
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Auto Configuration Setting
7.55. Auto Configuration function can be used to download the original configurations stored in
the TFTP or FTP server. This is useful for the new user to automatically download a
predefined configuration setting. Before enabling this auto configuration, you must
select Bridge mode and Fixed IP type in Network settings. After enabling this
function, please click the “Submit” button. Remember to click “Save” in the Save Change
section. The V102 will then reboot and automatically download the original configurations
from the TFTP or FTP server.
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User Password
7.56. You may change the login name and password in this page.
Save Changes
7.57. You can save the changes you have made, and click the Save button. After clicking the
“Save” button, the V102 will automatically save the new settings.
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Update
7.58. V102 provides two methods, HTTP or TFTP, to update new firmware as the following
steps:
7.59. Select the firmware code type, Risc or DSP code. (mostly for Risc code)
7.60. Click the “Browse” button to choose the updated file location for HTTP download, or
7.61. Select TFTP and enter the IP address of TFTP server for firmware download, then click
the “Update” button.
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7.62. After clicking the “Update” button, the firmware list will be displayed from server to
indicate the available firmware for download.
7.63. Select the new file you want to download to the V102 then click the “Select” button.
7.64.
In 3 to 4 minutes, the PHONE LED indicators will start flashing 5 times to indicate successful
firmware update. Then, you need to login again new IP address which is available from IVR by
pressing #120# from phone.
7.65. NOTE: Do NOT power OFF the V102 after clicking the “Select” button, or you may damage
the V102.
7.66. NOTE: The remote TFTP download works only for public IP address.
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Default Setting:
7.67. You can restore the V102 to factory default in this page. By clicking the “Restore” button,
the V102 will restore to default and automatically restart again.
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Reboot
7.68. You may click the Reboot button to restart, then V102 will automatically reboot with the
stored configurations.
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8. Configurations from Telephone & IVR
You can use telephone to configure and to check the status of V102. Make sure that the LAN
port is connected to Ethernet, or you may hear a busy tone from the telephone.
Group IVR Action Phone Command Remarks
Status Check DHCP Type
#121#
IVR will announce if DHCP in enabled or
disabled. Hang up while hearing end tone.
Status Check TA LAN IP Address
#120#
IVR will announce the current TA IP
address. Hang up while hearing end tone.
Status Check Network Mask
#123#
IVR will announce the current TA network
mask. Hang up while hearing end tone.
Status Check Router IP Address
#124#
IVR will announce the current Router IP
address. Hang up while hearing end tone.
Status
Check Primary DNS
Server Setting
#125#
IVR will announce the current setting of
Primary DNS field.
Status
Check TA WAN IP
Address
#126#
IVR will announce the current TA WAN IP
address. Hang up while hearing end tone.
Status Check Firmware Version
#128#
IVR will announce the firmware version.
Setting Set DHCP client
#111#
This setting will enable DHCP Client.
Setting Set TA Static IP Address
#112xxx*xxx*xxx*xxx#
Ex: #112061* 066*159*009#
Note xxx must be 3 decimal digits.
This setting will disable DHCP Client.
Setting Set Network Mask
#113xxx*xxx*xxx*xxx#
Ex: #113255* 255*255*000#
Note xxx must be 3 decimal digits and
Must set TA Static IP first (#112).
Setting Set Router IP Address
#114xxx*xxx*xxx*xxx#
Ex: #114061* 066*159*254#
Note xxx must be 3 decimal digits and
Must set TA Static IP first (#112).
Setting Set Primary DNS Server
#115xxx*xxx*xxx*xxx#
Ex: #115159* 168*001*001#
Note xxx must be 3 decimal digits and
Must set TA Static IP first (#112).
Setting Set Codec
#130+[1-8]#
1:G.711 u-Law, 2: G.711 a-Law,
3: G.723.1, 4: G.729a,
5: G.726-16K, 6: G.726-24K,
7: G.726-32K, 8: G.726-40K.
Setting Set Handset Gain
#131+[00~15]#
Ex: #13107# and default is 06
Setting Set Handset Volume
#132+[00~12]#
Ex: #13209# and Default is 10
Setting Enable Call Waiting
#138#
This will disable call transfer function.
Setting Disable Call Waiting
#139#
This will enable call transfer function.
Setting Reboot
#195#
After you hear “Option Successful,” hangup and TA will reboot automatically.
Setting Factory Reset
#198#
T A will reset back to factory defaults.
WARNING: ALL “User-Changeable”
NONDEFAULT SETTINGS WILL BE
LOST!
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9. VoIP Application Examples
You can use PC Web browser to configure V102.
For example, enter http://192.168.1.100 from PC web browser.
A. ADSL Connections without NAT Router for V102
B. ADSL Connections with NAT Router for V102
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PC PHONE1
NAT Router
ADSL Modem
Router IP: 192.168.1.254
V102 IP: 192.168.1.100
LAN
INTERNET
PC
PHONE1
ADSL Modem
LAN
PHONE2
INTERNET
PHONE2
Example 1: SIP-to-SIP Calling/Answering
Applications:
The applications can be for ADSL connections as in both Diagrams A and B. Both parties are
registered to SIP server with either fixed real IP or private IP under NAT router. The SIP-to-SIP
calling works when both calling and answering parties are registered to SIP server with given
registered phone numbers. Please refer to Example 6 for more detailed SIP server regi strations.
Configurations:
1. Select “DHCP Client”, and bridge “ON” in the “Network / Network settings” pages,
2. Remember to click the “Submit” button,
3. Select Active “ON” in the “SIP settings / Service Domain” page s,
4. Enter the items of Register Name, Register Password, Proxy Server, and Outbound
Proxy,
5. Select “ON” in the “NAT settings / STUN setting” page, if Outbound Proxy is NOT available.
6. Upon successful SIP registration, the PHONE LED will start flashing.
Callings:
7. Pick up the phone, and you should hear a dial tone.
8. Press 1688 to call the party with the registered SIP phone number 1688. In a moment, you
should hear a ring back tone, and wait for the VoIP called party to answer.
Example 2: SIP to Direct IP Calling
Applications:
The application is for the calling party with ADSL connection as in either Diagrams A or B. The
calling party is registered to SIP server with either fixed real IP or private IP under NAT router. The
answering party is with fixed real IP.
Configurations:
1. Same as in Example 1.
2. Select “ON” in the “NAT settings / STUN setting” page, if Outbound Proxy is NOT available.
3. Make sure the PHONE LED is flashing continuously with a successful SIP registration.
Callings:
4. Pick up the phone, and you should hear a dial tone.
5.
Press 211*21*191*4# or 211*21*191*4 to call the party with the real IP address of
211.21.191.4. In a moment, you should hear another ring back tone, and wait for the
VoIP called party to answer.
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Example 3: Direct IP to Direct IP Calling/Answering
Applications:
The applications are for ADSL connection without NAT router as in Diagram A. Both
parties are with fixed real IP. The Direct IP calling works when both calling and answering
parties are with known fixed IP.
Configurations:
1. Select “Fixed IP”, and bridge “ON” in the “Network / Network settings” page,
2. Enter the items of IP, Subnet Mask, Gateway IP,
3. Click the “Submit” button.
4. Make sure the SIP server is OFF (default is OFF) and PHONE LED is NOT flashing.
Callings:
5. Pick up the phone, and you should hear a dial tone.
6.
Press 211*21*191*4# or 211*21*191*4 to call the party with the real IP address of
211.21.191.4. Note that # key will dial out the number immediately. Dialing without
# will not dial out until the auto dial timer (default=5 seconds) elapsed. In a
moment, you should hear a ring back tone, and wait for the VoIP called party to
answer.
Example 4: Direct IP to Direct IP Calling within NAT Router
Applications:
For the calling party in ADSL connection with NAT router as in Diagram B, this Direct IP
calling can work when the answering parties are with fixed private IP addresses within
the same VPN network, or with fixed real IP addresses.
Configurations:
1. Select “Fixed IP”, and bridge “ON” in the “Network / Network settings” page,
2. Enter the items of IP, Subnet Mask, Gateway IP,
3. Click the “Submit” button.
4. Make sure the SIP server is OFF (default is OFF) and PHONE LED is NOT flashing.
Callings:
5. Pick up the phone, and you should hear a dial tone.
6. Press 192*168*1*51# or 192*168*1*51 to call the party with the private IP address of
192.168.1.51. Press 211*21*191*4 to call the party with the real IP address of
211.21.191.4. In a moment, you should hear a ring back tone, and wait for the called
party to answer.
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Example 5: 3-Way Conference Call, Call Waiting, Call Hold
3-Way Conference Calling Application:
This is for 3-way conference call among Parties A, B, and C. Three parties are registered to SIP
server with either fixed real IP or private IP.
Callings:
1. Make a phone call from Party A to the first phone number Party B.
2. After the first call is established, press Flash key (or hook switch) from Party A to hold
the call, and Party A should hear a dial tone.
3. Make another phone call from Party A to the second phone number Party C.
4. After the second call is established, press Flash key (or hook switch) again from
Party A to join in Party B for three-way conference call.
Call Waiting Application:
When a new call is coming while you are talking, you can push the Flash key to switch
to the new call. You can push the Flash key to switch between the two calls.
Call Hold Application:
You may push the Hold key to hold the current call for a while, then push Hold key
again to resume talking.
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Example 6: SIP-to-SIP Calling for FreeWorld Dialup (FWD)
Applications:
This shows how to use FWD as an example for free ITSP provider. The applications are
for both parties registered to FWD SIP server.
1. Visit http://www.freeworlddialup.com
and sign up for a new registered account
number. Follow the instructions for registration.
2. After finished, you will receive a mail sent by the FWD mail system, and you will get
one FWD phone number and password in the mail. For example, the register
name/phone number is 636346 with password xxxx.
3. Login to the Web configuration page.
Configurations:
4. Bridge Settings
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5. SIP Settings
You have to enter the Display Name, User Name, Registered Name, Registered
Password, Domain Server (fwd.pulver.com), Proxy Server (fwd.pulver.com), Outbound
Proxy (fwdnat.pulver.com:5082). After finished the setting, click the Submit button and
the Save Change button. The system will reboot automatically. After system boot up,
the SIP setting page will show “Registered”, and the PHONE LED will start flashing.
FWD SIP Server Register Name: 636346 Password: xxxx
Domain Server: fwd.pulver.com, Proxy Server: fwd.pulver.com, S t un Server: stun.fwdnet.net
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Callings:
1. Pick up the phone, and you should hear a dial tone. (Your FWD phone number
636346).
7. Press 12345 to call the party with the registered phone number 12345. In a moment,
you should hear the ring back tone, and wait for the called party to answer.
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10. Advanced Settings for Embedded NAT
You may activate the embedded NAT server for V102. This NAT function is useful for ADSL
user without NAT router as in the Diagram A. First, enter the IP address in the Web page.
For example, enter http://192.168.1.100
from PC web browser.
Diagram A. ADSL Connections without NAT Router for V102
Web Login
1. Login Username: root and Password: test
2. Click Login button. The following System Information page will be shown.
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PC
PHONE1
ADSL Modem
LAN
PHONE2
INTERNET
NAT Settings
3. The NAT settings page will be shown as follows.
LAN Setting:
4. Enable the embedded DHCP Server function for V102 and PC to get private IP addresses
automatically from the embedded DHCP server.
WAN Setting:
5. You may select PPPoE function, and enter the given username and password for ADSL
connection. After finished, please click the Submit button. Note that the MAC settings for
LAN and WAN must be different from each other.
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6. Click the Save button to save these settings.
7. After clicking the Save button, the V102 will show a Note Information page as the following
and will automaitcally restart.
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11. Trouble Shooting for Web Configurations
11.1. DO NOT HEAR DIAL TONE?
When you pick up the phone and hear a busy tone, please make sure the ADSL Ethernet is
connected to the WAN port.
11.2. CAN NOT ACCESS WEB PAGE?
If you encounter the problem when accessing http://192.168.1.100 (V102’s default IP
address) from web browser, it’s likely that your PC is not in the same subnet as
192.168.1.xxx. In this case, you must change V102 IP address to the same subnet as PC
and NAT router. You can find your PC’s IP setting, using “ipconfig” command in “Command
Prompt” window. Then, change V102’s IP address to the same subnet as PC and NAT
router.
Example: To change V102 IP address to the same subnet as PC and NAT router. Assume
that PC and router are in the subnet of 192.168.62.xxx.
1. Pick up the phone and press
#111# from the phone to enable DHCP Client mode. V102 will
reboot, and LED will start flashing to get an IP address from NAT router.
2. Press #120# to obtain the V102 IP address from telephone IVR, for example,
192.168.62.51.
3. Enter from PC web browser http://192.168.62.51 to login V102 web page for
configurations.
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PC
PHONE1
NAT Router
LAN
PHONE2
INTERNET
11.3. CONFIGURE PC’S IP SETTINGS FOR V102 EMBEDDED NAT FUNCTION?
If you don’t have a router to connect both PC and V102 for sharing the only one IP address
from ADSL/Cable modem, you should enable the embedded NAT function inside V102. You
need to change your PC’s IP settings to recognize V102 as the default gateway. In this case,
you should enable the embedded NAT router of V102 to provide more than one private IP
addresses 192.168.1.xxx for PC and V102.
Example: To change PC’s IP setting to connect to V102
1. In Window XP
- At “Control Panel”, open "Network Connections".
- right click on "Local Area Connection", and then click on Properties.
- The "Local Area Connection Properties" window will pop up.
- Double click on "Internet Protocol (TCP/IP)".
- The "Internet Protocol (TCP/IP) Properties" window will pop up.
- Click on "Use the following IP Address".
- Enter IP: 192.168.1.50 (50 can be any number other than 100, which is used by V102).
- Enter Subnet mask: 255.255.255.0
- Enter Default gateway: 192.168.1.100
- Click on OK button.
2. You will lose internet connection at this time.
3. At IE browser, enter http://192.168.1.100
.
4. Follow the example "Chapter 11 Advanced Settings for Embedded NAT" for web login.
5. At LAN setting, turn on DHCP server.
6. At WAN setting, set as instructed by your ISP. (e.g. DHCP Client for cable modem, or
PPPoE for ADSL)
7. Save change, and reboot V102 to restart and enable embedded NAT function.
8. Change your PC's "Internet Protocol (TCP/IP) Properties" back to "obtain an IP
address automatically".
9. Go back to http://192.168.1.100
to configure V102 for SIP settings.
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PC
PHONE1
ADSL Modem
LAN
PHONE2
INTERNET
Funkwerk Enterprise Communications GmbH - Südwestpar k 94 - D-90449 Nürnberg
Telefon: +49 - 180 300 9191 0
Telefax: +49 - 180 300 9193 0
E-Mail: info@funkwerk-ec.com - www. funkwerk-ec.com
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