Funkwerk V101 User manual

V101
SIP VoIP Telephone Adaptor
User Manual
V1.1m

Quick Guide
Step 1: Broadband (ADSL/Cable Modem) Connections for V101
A. Connect V101 LAN port to ADSL NAT Router as the following connection.
B. Connect V101 PC port to PC LAN port using a Category 5 LAN cable.
C. Connect V101 RJ11 PHONE port to a Telephone Set.
D. Connect the power adaptor to power on V101, and the POWER LED will be lit. In 5
seconds, the PHONE LED will start flashing 5 times and be ready for configurations.
E. Pick up the phone, the PHONE LED will be lit, and you should hear a dial tone.
F. Press #121# and #120# from the phone to check the DHCP status and the IP address (e.g.
192.168.1.100) for V101. After the IP announcement, please hang up.
a. ADSL Connections with external NAT Router for V101
Step 2: Settings for V101 with NAT Router
A. V101 is equipped with DHCP agent to automatically get an IPAddress from NAT router
B. Press #120# from the phone to listen and check IP address (e.g. 192.168.1.100) for V101.
C. Enter the IP address from PC Web browser for V101 configuration settings.
Example: Enter http://192.168.1.100, if V101 IP address is 192.168.1.100
Step 3: Making Point-To-Point SIP Calls
A. While the PHONE LED is flashing continuously showing a successful registration in the SIP server.
B. Pick up the phone, and you should hear a dial tone.
C. 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 information, please refer to V101 user
manual.
PC Phone
NAT Router
ADSL Modem
Router IP: 192.168.1.1
V101 IP: 192.168.1.100
LAN
PC IP: 192.168.1.101

TABLE OF CONTENTS
1.Introductions………………………………………………………………
2.Features ………………………………………………………………………
3.Standard Compliances……………………………………………
4.Packing Contents ……………………………………………………
5.LED Indicators……………………………………………………………
6.Installations & SIP Configurations ……………………
7.Default Reset from Telephone ……………………………
8.Configurations from Web browser……………………
9.Configurations from Telephone …………………………
10.Applications Examples…………………………………………
11.Advanced Settings for Embedded NAT …………
12.Trouble Shooting for Web Configurations ……
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1. Introduction
The V101 is a single port Telephone Adaptor (TA) with SIP Protocols for Voice over IP (VoIP)
applications. Connecting to the Internet, the V101 can make a voice phone call over the
Internet from one IP to another. V101 provides two Ethernet LAN ports for connections to
Internet and Notebook PC, plus one phone port for telephone set. With an embedded
NAT/DHCP server, the V101 can be easily configured through the Web browser, and is very
suitable for ITSP (Internet Telephony Service Providers) customers and SOHO users to make
VoIP calls.
Each V101 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.
If the ITSP provides dynamic IP to customers, V101 user may enable PPPoE or DHCP
features to automatically get an assigned dynamic IP. Sometimes, the V101 Media Access
Control (MAC) hardware address may be required, and it is indicated at the bottom of V101.
Please refer to Section 8 Configuration from Web browser for detailed information.
2.Features
The V101 VoIP TA is equipped with one RJ11 connector for POTS, and two RJ45 connectors
for Router and PC connections. The V101 is featuring as the following
¾Three LED Indicators for V101: POWER, PHONE, LAN
¾RJ45 x 2 for Ethernet + RJ11 x 1 for FXS
¾Web Browser and Telephone Configurations
¾Embedded NAT/DHCP Server
¾TCP/IP, UDP, RTP/RTCP, ICMP, ARP/RARP
¾PPPoE/DHCP Client for Dynamic IP plus NAT, DNS, and DDNS Clients
¾Support STUN server for NAT Traversal
¾Interactive Voice Recording (IVR) for telephone IP status
¾Speed Dial, Call Forward/Waiting, Call Transfer/Hold and 3-Way Conference
¾Remote Firmware Upgraded with HTTP or TFTP server by Web PC
¾Direct IP/URL Dial without SIP Proxy or Dial number via SIP server
¾Telephone features: VolumeAdjustment, Phone book, Speed Dial, Redial, and Flash
¾Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info
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3. Standard Compliances
The V101 VoIP TA supports for the following standards
VoIP Protocol: IETF RFC3261 and RFC 2543 for SIP
SIPAuthentication: IETF RFC2069 and RFC 2617 for MD5
Speech Codec: ITU-T G.711, G.723, G.729A/B, VAD and CNG
Echo Cancellation: ITU-T G.165/168
4. Packing Contents
Inside the package you should find:
(1) One V101 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.
5. LED Indicators and Connectors
On the front panel of V101, there are three LED indicators as the following
POWER: “On” indicates the power is normal
PHONE: “On” indicates the telephone is Off-hook
“Flashing,” indicates successful registration at SIP server for V101
LAN: “On” indicates the Ethernet LAN port is in Connection
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Power Phone PC LAN
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6. Installations & SIP Configurations
1. Connect V101 RJ45 LAN port to NAT Router using a Category 5 LAN cable.
2. Connect V101 RJ45 PC port to Notebook PC using a Category 5 LAN cable.
3. Connect V101 RJ11 PHONE port to a Plain Old Telephone Set (POTS).
4. Connect the power adaptor to power on V101, and the POWER LED will be lit constantly.
5. 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 constantly ON
when any one of RJ45 ports is connected. If the PHONE LED keeps flashing, it indicates
that V101 has successfully registered in the SIP server.
6. Pick up the phone, the PHONE LED will be lit and you should hear a dial tone. If you
hear a busy tone, please check if the LAN port is connected properly.
7. Press #120# to listen and check the assigned IP address for the V101. The default IP
address is 192.168.1.100. This IP address will be used for the Web configurations from
Notebook PC. Please refer to Section 8 for Web configurations.
8. After successful registration to the SIP server, the PHONE LED will keep continuous
flashing. Press 123456 to call the party with the number 123456 registered in the SIP
server. In a moment (5 seconds), you should hear the ring back tone, and wait for answer.
Note that you may press 123456# to dial out the number immediately. Dialing without #
will not dial out until the auto dial timer (default=5 seconds) elapsed.
7. Default Reset from Telephone
V101 provide an easy way to reset to factory defaults by using Telephone.
From Telephone:
Pick up the phone and press #198#. The V101 will reset back to factory defaults, and enter
into POWER ON cycle. The PHONE LED indicators will be OFF for about 5 seconds and start
flashing for 5 times. The POWER LED then will be lit constantly, and the PHONE LED will be
OFF. If the PHONE LED keep flashing, it indicates that V101 has successfully registered in the
SIP server.
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8. Configurations from Web browser
You can use Web browser to configure V101. First enter the IP address in the Web page.
8.1. Please enter the default IP address http://192.168.1.100 from PC Web browser.
The following Web page should be displayed on PC. If you have difficulties accessing the
Web page from the PC Web browser, the subnet IP of PC might be different from
192.168.1.xxx.In this case, please refer to Chapter 12 for trouble shooting.
8.2. Please enter the username and password into the blank field. The default settings are:
Username: root
Password: test
8.3. 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 to activate the new settings.
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System Information
8.4. You will see the system information like firmware version, Codec, etc in this page.
8.5. You may click the button listed in the left hand side to configure the V101.
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Phone Book & Speed Dial
8.6. Phone Book page specifies Speed Dial function.
8.7. For Speed Dial function you can add/delete Speed Dial number up to maximum 10 entries
in Speed Dial Phone List.
8.8. 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.
8.9. If you want to delete a phone number, please select the phone number you want to delete
then click “Delete Selected” button.
8.10. If you want to delete all phone numbers, please click “Delete All” button.
8.11. Example: Press 2# on telephone will Speed Dial the phone number 2immediately.
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Call Settings
8.12. There are 8 sub pages as follows; Call Forward, SNTP, Volume, Block Settings, Caller ID,
Auto-Dial Timer, Telephone Flash key (or hook switch), and Call Waiting functions.
Call Forward function:
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.
When you finished the setting, please click the “Submit” button.
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SNTP Setting:
8.13. 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 finished the setting, please click the “Submit” button.
Volume Setting:
8.14. You can setup the Handset Volume, Ringer Volume, and the Handset Gain in this page.
When you finished the setting, please click the “Submit” button.
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Block Setting:
8.15. You can setup the Block Setting to keep the phone silence. You can choose either Always
Block or a Block period.
8.16. Always Block: All incoming call will be blocked until disable this feature.
8.17. 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.
8.18. When you finished the setting, please click the “Submit” button.
Caller ID Setting:
You may enable caller ID function by setting “Yes” in Single Caller ID, and select the
desired Caller ID options. When you finished the setting, please click the “Submit” button.
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Auto Dial Setting:
You can set the timer for inter dial digit in this page. When the timer expires after finished
dialing, V101 will start making the call. When you finished the setting, please click the
“Submit” button.
Flash Time Setting:
You can set the time duration for the telephone flash key/hook switch in this page. The
telephone flash key is quite useful for the 3-way conference call and the call waiting
function. When you finished the setting, please click the “Submit” button.
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Call Waiting Setting:
The call waiting function allows users to answer another coming call by pressing flash key
while holding the current call, and back to previous call by pressing flash key again.
When you finished the setting, please click the “Submit” button.
Call Transfer Setting:
The call transfer function allows users to answer a coming call and to hold the current call
by pressing flash key, and then transfer the current call to the desired party by dialing the
desired party number ended with # key. This function is exclusive with call waiting, and
you may enable call transfer function by disabling the call waiting function (#139#).
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Network
8.19. You can check the Network status, and configure the Network Settings and DDNS
settings in this section.
Network Status:
8.20. You can check and show the current Network setting in this page.
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Bridge Settings:
8.21. You can configure Network setting for V101 in this page.
8.22. The TCP/IP Configuration item is for the IP configuration of the LAN port network for
V101.
8.23. The PPPoE Configuration item is for the PPPoE Username and Password. If you have
the PPPoE account from your ITSP, you may enter the Username and the Password
here.
8.24. The Bridge mode is ON at default, and the two PC and LAN Ethernet ports will be
transparent. This is useful when V101 is connected with external NAT router.
8.25. When you finished the setting, please click the “Submit” button.
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NAT Settings:
8.26. You must disable Bridge mode when using embedded NAT setting for V101 in this page.
8.27. The PPPoE Configuration item is for the PPPoE Username and Password. If you have
the PPPoE account from your ITSP, you may enter the Username and the Password
here.
8.28. When you finished the setting, please click the “Submit” button.
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DDNS Setting:
8.29. You can configure the DDNS setting in this page. You need to have the DDNS account
before entering the information. When you finished the setting, please click the Submit
button.
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SIP Settings
8.30. You can setup the Service Domain, Codec Settings, Codec ID Settings, and Other
Settings for SIP in this page.
Service Domain Settings:
8.31. You may register up to three SIP accounts in the V101. You can call your friends via first
enabled SIP account and receive the phone from all the three SIP accounts.
8.32. Click “Active” ON to enable the Service Domain, then enter the following items:
8.33. Display Name: enter the name you want to display.
8.34. User Name: enter the User Name given by your ITSP.
8.35. Register Name: enter the Register Name given by your ITSP.
8.36. Register Password: enter the Register Password given by your ITSP.
8.37. Domain Server: enter the Domain Server given by your ITSP.
8.38. Proxy Server: enter the Proxy Server given by your ITSP.
8.39. Outbound Proxy: enter the Outbound Proxy of ITSP. If not provided, you may skip
this.
8.40. Register Period: enter the Register Period in minute given by your ITSP.
8.41. When it shows “Registered” in the Register Status, it indicate a successful registration to
the ITSP, and the “PHONE” LED will start flashing. The V101 is then ready for VoIP call.
8.42. If you have more than one SIP account, please follow the steps to register to other ITSPs.
8.43. When you finished the setting, please click the “Submit” button.
DTMF Settings:
8.44. You can setup the options for DTMF function in this page. The options include RFC2833
(Outband DTMF), Inband DTMF, and DTMF SIP info. The default is set at Inband DTMF.
If you are making two-stage callings, you may need to select Outband DTMF option.
Port Settings:
8.45. You can setup the SIP and RTP port number in this page. Each ITSP provider will 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:
8.46. You may enable the STUN server for SIP in this page. The STUN function must be
enabled for the V101 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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