urmet domus 4501 Troubleshooting guide

DS 4501-006A LBT 8321
Mod.
4501
VOIP INTERFACE “DOMUS VOIP ATA”
Ref. 4501/30
QUICK INSTALLATION GUIDE
&
USER MANUAL
VOIP ATA

2DS4501-006A
INFORMATIVE NOTES
INTENDED USE OF THE DEVICE
This device has been designed and realized for connection to national telephone network (PSTN) or for Urmet switchboard (PABX) extension
using, or for a local network (LAN). Therefore his installation, and his programming – where necessary – must be done by qualified staff.
PACKAGING
Check that packing and content are not visibly damaged. Contact the retailer immediately if parts are either missing or damaged. Do not
attempt to use the device in this case. Send the product back in its original packing if it is damaged.
RECOMMENDED INSTALLATION LOCATION
It is advisable to use the device in sheltered environments with temperature in the range from 5 to 40° and relative humidity in the range
from 20 to 75%.
Keep the device away from directly sunlight or sources of heat.
DEVICE INSTALLATION ADVICE
The VoIP ATA device was designed to work with power supply.
CLEANING THE DEVICE
Use a moist or antistatic cloth for cleaning the device only. Do not use volatile liquids, like petrol, alcohol, solvents, etc., or chemically
treated cloths; they could damage the surface finish.
Normally the phone feet do not mark the surface it is rested on but in consideration of the variety of materials available, it is advisable to
clean the resting points regularly.
CONTENTS OF THE BOX
• n.1 VoIP ATA
• n.1 telephone plug/plug line cord
• n.1 Ethernet cable (RJ45)
• n.1 adapter kit
• n.1 “CONSUMER RIGHTS” list
• n.1 “WEEE DISPOSAL INFORMATION NOTE” sheet
• n.1 instruction manual
• n.1 CD
Urmet reserves the right to change the product specifications and performance without prior notice.
INDEX
QUICK INSTALLATION GUIDE ........................................................... 3
USER MANUAL .................................................................................. 7
1. Introduction ........................................................................... 7
2. Features................................................................................. 7
3. Standard Compliances .......................................................... 7
4. LED Indicators ....................................................................... 7
5. Installations & SIP Configurations ......................................... 7
6. Default reset by telephone..................................................... 7
7. Configurations by Web Browser............................................ 7
8. Configurations by Telephone & IVR ..................................... 35
9. VoIP Application Examples.................................................. 36
10. Trouble Shooting for Web Configurations............................ 38
11. Glossary............................................................................... 39

DS4501-006A 3
QUICK INSTALLATION GUIDE
STEP1: VOIP ATA LED AND REAR PANEL
LINE PHONE2 PHONE1 LAN WAN
RET
POWER
DESCRIPTION
Connection to the main power
Connection to the WAN (i.e. modem/router ADSL)
Connection for analog device (telephone or PBX)
Not used
Connection to PSTN line
Connection to LAN (i.e. PC for programming)
CONNECTION
POWER
WAN
Phone1
Phone2
Line
LAN
DESCRIPTION
ON when the main power is present
ON when there’s a correct WAN connection
ON when there’s a successful SIP registration
Blinking with phone in use
Blinking with PSTN line in use
ON when a PC is connected to the LAN port
LED
POWER
WAN
REG
Phone
Line
LAN
LAN REG Phone LineWANPOWER
VOIP ATA
STEP 2: BROADBAND (ADSL MODEM/CABLE). CONNECTIONS FOR VOIP ATA
1. Connect the VoIP ATA RJ45 WAN port to the ADSL Router/Modem using a direct network cable.
2. Connect the VoIP ATA RJ45 LAN port to the LAN port of the PC using a straight, direct cable.
3. Connect the VoIP ATA RJ11 Line port to a PSTN telephone line.
4. Connect the VoIP ATA RJ11 Phone port to an analog telephone.
5. Connect the DC adapter. After power on, the power LED will be ON.
6. REG LED will be on for a successful SIP registration. Pick up the phone and Phone LED will be on indicating VoIP mode.
7. Press 0* key, and the Line LED will be on, indicating PSTN mode.
Example of connection of the VoIP ATA to an ADSL Router/Modem
LAN
192.168.123.1
(default)
Analog
Telephone
Line
Phone
Telephone
network
Router/Modem
ADSL
192.168.123.150
WAN
Audio
Data
Tel n
Splitter
Tel 1
192.168.101.1
2 wires
Ethernet
STEP 3: CONFIGURATION VIA WEB BROWSER AND PC
1. Connect a PC to the LAN port and start a browser to access web configuration.
2. Default LAN port IP address is 192.168.123.1, type http://192.168.123.1:9999 in the web browser address field to connect to the web
configuration page.
3. Login using the default username and password:
Username: root
Password: test
Click on “Login” to access configurations.

4DS4501-006A
4. The following system information page will appear on the web browser.
5. Program the following network configurations in the left-hand column: Phone, Network WAN/LAN Settings, SIP Settings. Remember to
click on “Save Change” and then save to confirm and activate settings: the device will be restarted with the new settings.
6. Three different modes are available for WAN configurations (WAN mode).
DHCP:
This mode is used if the device is connected to an ADSL router with enabled DHCP server, i.e. which can automatically assign addresses to
the PCs and to the devices connected to it.
This option is used for assigning a dynamic IP address and is generally chosen as WAN IP address. When this option is selected, the IP
address, the Subnet mask and the Gateway IP address will be automatically assigned. Remember to click on Submit to make the changes
effective.
PPPoE:
This mode is used if the device is directly connected to an ADSL modem, like a PC.
When this option is selected, the PPPoE username and password must be entered and the IP address will be automatically assigned to the
WAN port.
Fixed IP:
This option is used to manually program the IP address, the Subnet mask and the Gateway IP, if this is necessary due to the presence of a
previously configured network or specific address allocations (static IP connection) given by a provider (ITSP).

DS4501-006A 5
Two configuration modes are possible for the LAN (LAN mode):
NAT (default):
By using this mode, the WAN and LAN ports refer to two separate interfaces; in this case, as shown in the example above, the PC connected
to the LAN port will obtain the address from the device (192.168.123.x).
Bridge:
By using this mode, the device will behave transparently (as an Ethernet hub/bridge) and in this case, the PC connected to the LAN port will
obtain the address from the ADSL Router/Modem to which the device is connected (192.168.101.x in the mentioned example).
This mode cannot work without a router which assigns the addresses and may be useful in presence of a company network in which you
want the PC to take the address from the network.
7. SIP configurations:
In order to use the VoIP service, a SIP account registered at one of the many SIP providers is needed. Register to obtain the account
connection data.
Registration to the Eutelia server is shown by way of example. Connect to the site.
http://www.euteliavoip.com
Follow the instructions for registering and activating the account. You will be asked to choose a device number and will receive connection
data like those shown below):
Login 01119838123
(this is the user name of account, which is the same in this case of the device number chosen)
Password ********
Domain voip.eutelia.it
Sip Proxy Server voip.eutelia.it
Sip Proxy Socket 5060
Stun Server stun.voip.eutelia.it
Stun Server Socket 3478
These data will then be requested during the SIP account configuration step on the device, as we will see below.
The Stun connection data may be needed if the device is connected to a router to which several PC and devices are connected,
typically a large network, where the router works as a NAT, i.e. filters and re-routes packages to the various devices.
Refer to the following paragraph for entering Stun data.
There are many VoIP providers offering this type of services, in addition to Eutelia. Only some are listed here.
Messagenet (http://www.messagenet.it/)
CheapVoIP (http://www.cheapnet.it/supporto/guide/voip/)
Squillo (http://www.ngi.it/squillo/)
The various providers offer different services with different prices and policies. Choose the provider, from the list or others, who offer the
service most in line with your needs.
The user can configure up to 3 SIP registrations (realms) in the Service Domain Settings configurations.
It may be useful to have several configured accounts if several different people use the device or if you want to use several SIP providers at
the same time because they are more convenient for certain services.
Account (realm) 1 (default for the first SIP account registration). For example:
Active (state of the device): ON
Display Name (name of the user): Mario
User Name (registered at the SIP provider): 01119838123
Register Name (registered at the SIP provider): 01119838123
Register Password (password registered at the SIP provider): ********
Proxy Server (server + registration port, provided by the SIP provider): voip.eutelia.it:5060
Domain Server (server + registration port, provided by the SIP provider): voip.eutelia.it:5060
Outbound Proxy:
Accounts (realm) 2 & 3 (for the second and third SIP account registration, similar to the previous registration)
Active: ON
Display Name: Lucia
User Name: 01119838124
Register Name: 01119838124
Register Password: ********
Proxy Server: voip.eutelia.it:5060
Domain Server: voip.eutelia.it:5060
Outbound Proxy:
Number selection for SIP account
With three registered SIP accounts, the VoIP ATA can receive all incoming calls from the three SIP servers. Accounts (Realms) 1, 2, and
3are used to configure the registered numbers for the first, second and third SIP service, respectively. The VoIP ATA can register different
telephone numbers on different SIP servers. Lift the handset and press 2_* to select the second SIP server. Similarly, press 3_* to select the
third SIP server. The first account is used by default. Make sure that the Accounts (Realm) 1, 2 and 3 are all registered and that the REG LED
is ON.
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6DS4501-006A
STEP 4: HOW TO MAKE POINT-POINT SIP CALLS
1. Select one of the three SIP servers, if you have more, otherwise simply pick up the handset: you will hear the dial tone.
2. Key in 123456# to call number 123456 registered to the SIP server. Note that #is used to forward the call immediately. You will hear the
calling tone. Wait for the called party to pick up.
3. For other applications, refer to the user manual for other VoIP application examples.
STEP 5: HOW TO MAKE PSTN CALLS
1. Press the 0* key to engage the line.
2. Dial the required number.
If you have difficulties in configuring VoIP ATA, please refer to the chapter 10 for trouble shootings.§

DS4501-006A 7
USER MANUAL
1. INTRODUCTION
The Domus VoIP ATA is a SIP protocol based IP gateway for Voice Over IP applications (VoIP). It is provided with a PSTN (Line) port for
connection to the traditional telephone line and a phone port for an analog telephone.
Connected to the Internet, to a phone and to a PSTN line , the Domus VoIP ATA can be used to make VoIP calls and traditional telephone
calls.
The Domus VoIP ATA has a WAN port for ADSL connections, a LAN port for connection to the PC and two RJ11 ports for connection to the PSTN
telephone line (FXO) and analog telephone (FXS). By means of the internal DHCP, VoIP ATA can be easily configured to adapt to the various
network architectures by means of a web browser and telephone keypad.
Note that the VoIP ATA requires an IP address, a subnet mask and the Router IP gateway address for use on the Internet. These three parameters
will be supplied by your Internet Service Provider. The VoIP ATA can activate PPPoE or DHCP functions for automatically obtaining a dynamic
IP address assigned by the Provider. Refer to the configurations made by means of a Web Browser described in section 7 for more detailed
information.
2. FEATURES
Equipped with two RJ11 connectors and two RJ45 connectors VoIP ATA is featuring as the following:
Six LED Indicators: POWER, WAN, LAN, REG, Phone, Line.
RJ45 x 2 for WAN and LAN ports + RJ11 x 2 for FXS and FXO ports.
Configurations by Web Browser and Telephone.
Embedded NAT/DHCP Server.
PPPoE/DHCP Client for Dynamic IP plus NAT, DNS, and DDNS Clients.
Support STUN server for NAT Traversal.
Support registrations for up to 3 SIP Servers.
Hot Line Mode.
Dial Plan Settings.
T.38 Fax over IP.
Interactive Voice Recording (IVR) for telephone IP status.
Phone Book, Call Forward/Transfer/Waiting/Hold, and 3-Way Conference Call features.
Auto Configurations by TFTP, HTTP, or FTP server.
Remote Firmware Upgraded with HTTP or TFTP server by Web PC.
Direct IP/URL Dial without SIP Proxy or Dial number via SIP server.
Telephone features: Volume Adjustment, Phone book and Flash.
Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info.
3. STANDARD COMPLIANCES
The Domus VoIP ATA supports for the following standards:
VoIP Protocol: IETF RFC3261
SIP Authentication: 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. LED INDICATORS
On the front panel of VoIP ATA, there are six LED indicators as the following
POWER: “On” indicates the power is normal
REG: “On” indicates successful registration at SIP server.
Phone: “Flashing” indicates the Phone is in use (off-hook) for VoIP call.
Line : “Flashing” indicates the Phone and the PSTN Line are in use for PSTN call.
WAN: “On” indicates the WAN port is in Connection.
LAN: “On” indicates the WAN port is in Connection.
5. INSTALLATIONS & SIP CONFIGURATIONS
Refer to the ‘Quick Installation Manual’ for SIP installation and configuration.
6. DEFAULT RESET BY TELEPHONE
VoIP ATA provides an easy way to reset to factory defaults by using Telephone. Pick up the phone and press #190# and you will hear “do do, do
do” of audio then hung up the phone to unlock. Then pick up the phone again and press #198# to reset back to factory defaults, and the VoIP
ATA 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.
7. CONFIGURATIONS BY WEB BROWSER
You may enter the IP address from PC Web browser to configure VoIP ATA.
Please enter the default IP address http://192.168.123.1:9999 from PC Web browser.
The following Web page shall 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 192.168.123.xxx. In this case, please refer to Chapter 10 for trouble shooting.
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8DS4501-006A
Please enter the username and password into the blank field. The default settings are:
Username: root
Password: test
Click the “Login” button to enter the System information page for web configurations.
Whenever you change the setting in each Web page, 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.
SYSTEM INFORMATION
The system information shows firmware version, Codec, etc.
You may click the buttons, for example Phone Book, at the left hand side to configure the VoIP ATA.

DS4501-006A 9
PHONE BOOK
PHONE BOOK SETTINGS
You may add/delete Name up to maximum 140 entries in Phone book list.
To add a phone name, you need to enter the position, the name, and the phone URL. When you finished a new phone list, just click the “Add
Phone” button.
To delete a phone name, please select the phone name then click “Delete Selected” button.
To delete all phone names, please click “Delete All” button.
After dialing a phone number, the VoIP ATA will first match with the Name in the phone book. If matched, VoIP ATA will send out the corresponding
URL instead. If not, the same dialed number will be sent out.
Example 1: Name: 333, URL: 192.168.1.100
Press 333# on telephone, and the phone at 192.168.1.100 will start ringing.
Example 2: Name: 201, URL: 14081234567
Press 201# on telephone, and VoIP ATA will call the registered number 14081234567.
Example 3: Name: 301, URL: 192.168.1.100:5062
Press 301# on telephone, and VoIP ATA will call at 192.168.1.100 with port 5062.
Example 4: No Name 401 in phone book.
Press 401# on telephone, and VoIP ATA will call the registered number 401.
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10 DS4501-006A
PHONE SETTING
CALL FORWARD SETTING
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 you are busy and on the phone.
No Answer Forward: The incoming call will forward to the URL when no answer.
You need to set the Time Out ring which will initiate No-Answer forwarding to the number you choose. When you finished the setting, please
click the “Submit” button.

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SNTP SETTINGS
You can setup the primary and the second SNTP Server IP Addresses 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 SETTINGS
You can setup the Handset Volume/Gain, PSTN-Out Volume, and PSTN-In Gain in this page. Handset Volume is to set the volume hearing from
the handset. PSTN-Out Volume is to set the PSTN volume for you to hear. Handset Gain is to set the volume send out to the other side’s handset.
PSTN-In Gain is to set the volume send out to the other side’s handset.

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DND SETTINGS
You can setup the DND (Do Not Disturb) to keep the phone slience. You can choose DND Always or DND period..
DND Always: All incoming call will be blocked until the feature is disabled.
DND Period: Set a time period and the phone will be blocked during the time period. If the “From” time is large than the “To” time, the
block time will from Day 1 to Day 2.
When you finished the setting, please click the Submit button.

DS4501-006A 13
AUTO ANSWER SETTING
You may enable the Auto Answer function to answer the incoming call by VoIP ATA FXO port. When the ring count exceeds the number set in
Auto Answer Counter, the FXO port will auto answer and allow for extension calls from VoIP to PSTN and vice vesa.
For the incoming call from the Internet, the FXO port will answer with a PSTN dial tone and allow caller to redial to PSTN phone number. For the
incoming call from PSTN, the VoIP ATA FXO port will answer with a short beep tone and allow caller to redial to VoIP number. PIN Code is used
to prevent from call piracy. The caller needs to enter the right PIN code followed by “#” to get the PSTN dial tone. Incorrect PIN Code will result
in call disconnect. The Anto Answer is disabled at default.
CALLER ID SETTING
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 finished the setting, please click the “Submit” button.
DIAL PLAN SETTINGS
Dial plan and auto dial timer settings can be set in this page. The dial plan allows you to map the dialing into an easy-to-remember phone number
system. The auto dial timer specifies the elapse time between the dialing digits. When Drop prefix is ON and the dialing prefix is matched, the
prefix will be dropped and replaced by the rule digits and followed by the rest of dialing digits. When Drop prefix is OFF and the dialing prefix is
matched, the rule digits will be added before the dialing digits in accord with the settings:
Symbol Representations
x or X
+
0, 1, 2, 3, 4, 5, 6, 7, 8, 9
or

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Example 1: Drop Prefix: No, Replace rule 1: 002, 8613+8662
a) Pressing 8613xxx will result in dialing out 002+8613+xxx.
b) Pressing 8662xxx will result in dialing out 002+8662+xxx.
Example 2: Drop Prefix: Yes, Replace rule 2: 006, 002+003+004+005+007+009
a) Pressing 002xxx will result in dialing out 006+xxx.
b) Pressing 003xxxx will result in dialing out 006+xxxx.
Example 3: Drop Prefix: No, Replace rule 3: 009, 12
a) Pressing 12xxx will result in dialing out 009+12xxx.
Example 4: Drop Prefix: No, Replace rule 4: 007, 5xxx+35xx+21xx
a) Pressing 5xxx will result in dialing out 007+5xxx.
b) Pressing 534 will result in dialing out 534 (not matched for the rest 3 digits).
c) Pressing 35xx will result in dialing out 007+35xx.
d) Pressing 356 will result in dialing out 356 (not matched for the rest 2 digits).
e) Pressing 35668 will result in dialing out 35668 (not matched for the rest 2 digits).
Example 5: Dial Now: *xx+#xx+11x+xxxxxxxx
a) Pressing *00, *01, *02.. *99 will result in dialing out the same *xx immediately.
b) Pressing #00, #01, #02.. #99 will result in dialing out the same #xx immediately.
c) Pressing 110, 111,.. 119 will result in dialing out 11x immediately.
d) Pressing 12345678 (8 digits) will result in dialing out 12345678 immediately. This implies that the phone numbers with 9 or more digits are
prohibited.
Auto Dial Timer: The inter-digit timer. Default is 5 seconds.
When you finish the setting, please click the Submit button.
Click the Save button. The changes you have made will be saved into the system and the system will reboot automatically.

DS4501-006A 15
FLASH TIME SETTING
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 finished the setting, please click
the “Submit” button.
CALL WAITING SETTING
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 finished the setting, please click the “Submit” button.

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CALL TRANSFER FUNCTION
The call transfer function allows users to answer an incoming 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. The call transfer function is exclusive with call waiting function.
You may enable call transfer function by disabling the call waiting function (#139#), or disable call transfer function by enabling the call waiting
function (#138#).
T.38 (FAX) SETTING
T.38 function can be used for FAX transmission over IP. Note that T.38 function must be enabled for both side of FAX over IP. You may enable
or disable the T.38 function. When you finished the setting, please click the Submit button.
HOT LINE SETTINGS
You can enable/disable and set Hot Line phone number in this page.
When Hot Line mode is enabled, you just pick up the phone and the VoIP ATA will call the Hot line number immediately. The default for Hot Line
mode is disabled.

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ALARM SETTINGS
You can configure the Alarm setting in this page.
NETWORK
You can check the Network status, and configure the WAN, LAN, DDNS, and VLAN settings in this section.
NETWORK STATUS
You can check and show the current Network setting in this page.

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WAN SETTINGS
The WAN setting is used to configure the Ethernet port connects to the ADSL Modem/Router.
The default setting is NAT mode for IP PHone, and this enables the embedded NAT router between the LAN port and PC port. You may change
to Bridge Mode if you need NOT use the embedded NAT router. When setting to Bridge Mode, the WAN and the LAN ports will be bridged.
There are three selections for WAN: Fixed IP, DHCP Client, and PPPoE modes. The WAN default is at DHCP Client Mode.
For Fix IP Mode, please make sure the IP address. Net Mask, Gateway, and DNS settings are suitable in your current network environment.
For PPPoE Mode, you have to enter correct username and password to get the IP address from your Internet Service Provider.
When you finished the setting, please click the Submit button.

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LAN SETTINGS
The default IP address is 192.168.123.1 for VoIP ATA, with Net Mask 255.255.255.0., and DHCP Server enabled. The IP addresses for DHCP
are from 150 to 200.
Connect your PC to the PC port, set your PC as DHCP Client mode, and then the PC will get an IP address from the VoIP ATA automatically.
When you finished the setting, please click the Submit button.

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DDNS SETTINGS
You can configure the DDNS setting in this page. You need to have the DDNS account and input the informations properly. You need a DDNS
account with a public IP address then others can call you via the DDNS account. Most of the VoIP applications are working with a SIP Proxy
Server as well as DDNS Server. When you finished the setting, please click the Submit button.