Octtel communication OR222W Wireless VOIP Gateway User Manual

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Date Submitted	2007-11-06 00:00:00
Date Available	2007-11-07 00:00:00
Creation Date	2007-10-24 10:41:22
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THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE
SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND
RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED
WITHOUT WARRANTY OF ANY KIND, EXPRESSED OR IMPLIED. USERS MUST TAKE FULL
RESPONSIBILITIES FOR THEIR APPLICATION OF THE PRODUCTS.
THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET
FORTH IN THE INFORMATION PACKET THAT IS SHIPPED WITH THE PRODUCT AND ARE
INCORPORATED HEREIN BY THIS REFERENCE.
NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF
THESE SUPPLIERS ARE PROVIDED “AS IS” WITH ALL FAULTS. PRODUCT AND THE ABOVE-NAMED
SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT
LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE.
IN NO EVENT SHALL THE PRODUCT OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL,
CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS
OR LOSS OR DAMAGE TO DATA ARISING FROM THE USE OR INABILITY TO USE THIS MANUAL, EVEN
IF THE PRODUCT OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH
DAMAGES.
Operation Manual V1.1
COPYRIGHT ©2007 All rights reserved.
Contents
1. Introduction....................................................................................................1
Product Overview......................................................................................................................................... 1
Hardware Connections and Description ...................................................................................................... 2
OR SERIES...................................................................................................................................... 2
2. Installation and Applications ........................................................................5
OR SERIES Assigned with a Public IP Address .......................................................................................... 5
OR SERIES in a NAT network ..................................................................................................................... 6
OR SERIES assigned with a Public IP Address and serving as a Bridge ................................................... 7
3. Setting the OR SERIES through IVR ............................................................8
IVR (Interactive Voice Response) ................................................................................................................ 8
IP Configuration Settings—Setting IP Configuration of WAN Port................................................. 11
Recorded Voice File ................................................................................................................................... 12
PPPoE Character Conversion Table .......................................................................................................... 13
4. Setting a OR SERIES with WEB Browser ..................................................14
Basic Network Settings .............................................................................................................................. 15
WAN ............................................................................................................................................... 15
SIP.................................................................................................................................................. 20
Phone Book.................................................................................................................................... 26
Basic Voice Services.................................................................................................................................. 28
Caller ID ......................................................................................................................................... 28
Hot Line .......................................................................................................................................... 30
Calling Features ............................................................................................................................. 34
Calling Features - Advanced Setting.............................................................................................. 36
PSTN Control ................................................................................................................................. 37
Emergency No................................................................................................................................ 37
Advanced Network Settings ....................................................................................................................... 38
LAN ................................................................................................................................................ 38
NAT Traversal................................................................................................................................. 41
DDNS ............................................................................................................................................. 42
Caller Filter ..................................................................................................................................... 43
PPTP Client.................................................................................................................................... 43
SIP Advanced................................................................................................................................. 44
Virtual Server.................................................................................................................................. 52
DMZ................................................................................................................................................ 52
Port Filtering ................................................................................................................................... 53
IP Filtering ...................................................................................................................................... 54
Advanced Voice Services........................................................................................................................... 55
Line Settings................................................................................................................................... 55
Codec Settings ............................................................................................................................... 62
FAX Settings................................................................................................................................... 63
Other Settings ............................................................................................................................................ 64
Digit Map ........................................................................................................................................ 64
DTMF & Pulse ................................................................................................................................ 70
CPT/Cadence Settings................................................................................................................... 72
Provision Settings .......................................................................................................................... 74
Transit Call Control......................................................................................................................... 76
Long-Distance Control Table .......................................................................................................... 77
Long Distance Exception Table...................................................................................................... 77
Status and Tools ......................................................................................................................................... 78
Current Status ................................................................................................................................ 78
RTP Packet Summary.................................................................................................................... 78
System Information ........................................................................................................................ 79
Ping Test......................................................................................................................................... 80
STUN Inquiry.................................................................................................................................. 80
System Settings ......................................................................................................................................... 81
NTP ................................................................................................................................................ 81
Login Account................................................................................................................................. 81
Backup/Restore.............................................................................................................................. 82
System Operations......................................................................................................................... 83
Software Upgrade .......................................................................................................................... 84
Logout ............................................................................................................................................ 84
5. TCP/IP Setting ..............................................................................................85
Terminal Adapter
1. Introduction
Product Overview
The stand-alone OR SERIES carries both voice and facsimile over the IP network. It supports
SIP industry standard call control protocol to be compatible with free registration services or
VoIP service providers’ systems. As a standard user agent, it is compatible to all well-known
Soft Switches and VSP(Voice Service Provider)/ SIP proxy servers
OR SERIES can be seamlessly integrated to existing network by connecting to a phone set,
fax machine or PSTN line. With only a broadband connection such as ADSL bridge/router,
Cable Modem or leased line router, it allows you to gain access to voice and FAX services over
the IP in order to get the convenient of VoIP and reduce the cost of international and long
distance calls.
In addition, the in-built router supports comprehensive Internet gateway functions to
accommodate other PCs or IP devices to share the same broadband stream. QoS function
allows voice and data traffic to flow through where voice traffic is transmitted in the highest
priority. With TOS bit enabled, it guarantees voice packets to have first priority to pass through
a TOS enabled router.
With the support of DDNS, it makes OR SERIES reachable by its domain name where the ISP
dynamically assigns the IP address.
OR SERIES can be assigned with a fixed IP address or by DHCP, PPPoE. It adopts the G.711,
G.729A or G.723.1 voice compression format to save the network bandwidth while providing
real-time and toll quality voice. In addition, in the event that the power supply fails or Internet
connection is lost, OR SERIES can automatically divert the FXS end to the PSTN network on
the PSTN port so users can still use the conventional PSTN line to make calls. This feature is
especially useful while dialing emergency calls (i.e. 911).
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Hardware Connections and Description
The diagram shows how OR SERIES connects to other devices in your network.
OR SERIES
Front Panel
OR201LW
Power: Power LED. A steady light indicates a proper connection to a power source.
Prov./Alm.: A blinking light indicates the VoIP Gateway is attempting to connect with the Provisioning
server. Once the service connects, the LED will turn off. The LED will light solid if the self-test or boot-up
fails.
Reg.: The Register LED will turn on when the VoIP Gateway is connected to a VoIP service provider.
The LED will turn off if not connected to a service provider.
WAN: When a connection is established the 10 or 100 LED will light up solid. The LED will blink to
indicate activity. If the 10 or 100 LED does not light up when a cable is connected, verify the cable
connections and make sure your devices are powered on.
WLAN: A steady light indicates a wireless connection. A blinking light indicates that the VoIP Gateway is
receiving/transmitting from/to the wireless network.
LAN(L1-L4): When a connection is established the 10 or 100 LED will light up solid on the appropriate
port. The LEDs will blink to indicate activity. If the 10 or 100 LED does not light up when a cable is
connected, verify the cable connections and make sure your devices are powered on.
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Phone: This LED displays the VoIP status and Hook/Ringing activity on the phone port that is used to
connect your normal telephone(s). If a phone connected to a phone port is off the hook or in use, this
LED will light solid. When a phone is ringing, the indicator will blink.
Line: Light on means the line is in use (off-hook), and vice versa.
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Model Description
2S1LW: It includes 2FXS+1LifeLine+Wireless Network. FXS stands for Phone 1-2 which are connected to
your analog telephone, and Life Line stand for Line port which is connected to your original telephone line on
the wall jack with RJ-11 cable. Phone 1 will be relayed to Line port when the user enter the feature code (refer
to Force Calling Thru PSTN code function) before FXS dials out via PSTN line or for emergency calls in the
occasion of a power failure. With wireless function enabled, you can easily build a wireless network.
Rear Panel
Line: Connect to your original telephone line on the wall jack with RJ-11 cable.
Phone Port (1-2): Connect to your phones using standard phone cabling (RJ-11).
LAN: Connect to your Ethernet enabled computers using Ethernet cabling.
WAN: Connect to your broadband modem using an Ethernet cable.
Power Receptor: Receptor for the provided power adapter.
Ground: A conducting connection with the earth. Connect with the ground so as to make the earth a
part of an electrical circuit using metal wire.
Antenna: Connect to a wireless network.
WARNING: DO NOT (1) connect the phone ports to each other (FXS to FXS) or (2) connect any
phone port directly to a PSTN line (FXS to PSTN) or to an internal PBX line (FXS to PBX
extension). Doing so may damage your VoIP Gateway.
Use Reset Button to restore factory default settings:
1.
Power on.
2.
Press and hold the reset button for 5 seconds.
Release the reset button. Factory settings will be restored.
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Terminal Adapter
2. Installation and Applications
The network interface is divided into 3 basic modes as described below:
OR SERIES can be assigned with a Public IP Address
OR SERIES can be built under the existing NAT
OR SERIES can be assigned with a Public IP address and serves as a Bridge device
OR SERIES Assigned with a Public IP Address
OR SERIES will have a Public IP address for Internet connection regardless of whether it is a
static IP address, DHCP (using a Cable Modem), or PPPoE (Dialup / ADSL).
OR SERIES IP Settings
Need to be set up as static IP, DHCP, or PPPoE
NAT/STUN Settings
Unnecessary (Disabled)
DDNS Settings
Unnecessary (Disabled)
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Terminal Adapter
OR SERIES in a NAT network
OR SERIES uses a virtual IP address and the IP sharing function of other systems to connect
to the Internet.
LAN IP address of IP sharing
Please avoid IP address 192.168.8.1-192.168.8.254 (You may
need to change the settings of IP sharing or change SIP series
Gateway LAN Port IP address)
OR SERIES IP Settings
Set as static IP address, and assign the LAN IP address of the
IP sharing to the Default Gateway.
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Terminal Adapter
OR SERIES assigned with a Public IP Address and
serving as a Bridge
OR SERIES will have a Public IP address regardless of whether it is a static IP application,
DHCP (using a Cable Modem), or PPPoE (To connect to your ADSL account), which can then
use the functions of built-in Bridge function to allow a PC to be on-line at the same time.
OR SERIES IP Settings
Need to be set up as static IP, DHCP, or PPPoE
NAT/STUN Settings
Unnecessary (Disabled)
DDNS Settings
Unnecessary (Disabled)
For settings at PC end
PC uses the original IP address
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Terminal Adapter
3. Setting the OR SERIES through IVR
VoIP transmits voice data (packet) via the Internet to achieve telecommunications. This means
that the telecommunication quality is closely related to the whole network environment. If any one
of the telecommunicating parties has insufficient bandwidth or frequent packet loss, the
telecommunication quality will be poor. Therefore, an excellent telecommunication can only be
created when OR SERIES is connected to the Internet and when network environment is stable.
Preparation
Install the OR SERIES according to instructions. Connect the power supply, telephone set,
telephone cable, and network cable properly as described in Chapter 2.
If a static IP is used, confirm the desired IP settings of the WAN Port (IP address, Subnet
Mask, and Default gateway). Please contact your local Internet Service Provider (ISP) if you
have any questions.
If using dialup ADSL (PPPoE) for network connection, confirm the dialup account number
and password.
If users wish to build OR SERIES under the NAT, OR SERIES WAN Port IP address and
LAN Port should not use the same range. This is to avoid network failures.
IVR (Interactive Voice Response)
OR SERIES provides convenient IVR functions. Users only need to pick up a handset and enter
the function code for the query and setting without using a PC.
Note: After finishing the settings, make sure the new settings are saved. This is so that the
new settings will take effect after OR SERIES is restarted.
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Instructions
FXS Port: When you have set the password in WEB-GUI with English character. To access
OR SERIES IVR function is different. Instead of **[password]#. You should press
***[password]#. The character to number conversion can be acquired from PPPoE Character
Conversion Table.
Example:
1. The factory default code is blank. Enter **#. You are now in IVR setting mode, enter the IVR
function code. Please refer to IVR Function Table for IVR function code.
2. if the password is 1234, then enter **1234#.
3. If your password is abc123 then you access IVR by pressing ***414243010203# .
FXO Port: To use IVR functions, dial the phone number of FXO Port using an external line.
You will hear the prompt “enter value”, and then enter a PIN number. The factory default code
is blank. Enter “**#” as above. You are now in IVR setting mode.
Once the first setting or query has been completed, you will hear a dial tone. Then use the
same procedure to make a second query or setting. To exit IVR mode, simply hang up the
phone.
Example:
1. Enter **# . You are now in IVR setting mode.
2. Enter 101 (to query IP address) . OR SERIES responds with an IP address.
3. You can continue with more settings or queries: enter 111 (to set IP address)
192*168*1*3 (IP number).
enter
Save Settings
After entering IVR mode, dial 509 (Save Settings). Wait for about 3 seconds and after hearing a
confirmation tone “1”, hang up the phone. Please reboot OR SERIES to enable the new settings.
To inquire about current OR SERIES’s WAN Port IP address
After entering IVR mode, dial 101. OR SERIES will repeat the current WAN Port IP address.
If OR SERIES does not repeat the IP address, it indicates that OR SERIES is not currently
connected to the Internet. Please check if the cable connection, account number, and password
are correct.
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Terminal Adapter
IVR Functions Table:
Function
Code
Description
111/101
Set/Query WAN Port IP address
112/102
Set/Query WAN Port Subnet Mask
113/103
Set/Query WAN Port Default OR SERIES
114/104
Set/Query WAN Port IP Type
(1: Static IP, 2.DHCP, 3.PPPoE)
116/106
Set/Query Phone Book Manager Server IP address
117/107
Remark
Use in conjunction with
function code 114, select
1 for a Static IP function.
Must use 116/106,
117/107 in conjunction
Set/Query whether or not to login Phone Book Manager
with each other.
(0: Disable 1:Enable)
066
Querying the connection to Phone books manager
118
Restart
121
Setting PPPoE Account
122
Setting PPPoE Password
311/301
Set/Query LAN Port IP address
131/132
Play/Record greeting message
OR SERIES ONLY
133
Saving greeting message
OR SERIES ONLY
215/205
Set/Query OR SERIES Telephone Number
(Representative Number)
216/206
Set/Query the extension number of Line 1.
Use in conjunction with
function code 114, select
3 for a PPPoE function
109
Restoring factory default setting of IP
409
Restoring factory default settings
509
Save settings
900
Set IVR and the language used on the Web GUI
(1: English, 2: Traditional Chinese, 3: Simplified Chinese)
209
Soft Upgrade
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SIP OPERATION MANUAL
The default of Static IP
IP: 192.168.1.2
Mask: 255.255.255.0
Gateway: 192.168.1.254
Terminal Adapter
IP Configuration Settings—Setting IP Configuration of WAN Port
Static IP Settings
Note: Before setting Static IP, you must have IP address (111), Subnet Mask (112) and
Default Gateway (113) provided by your local Internet Service Provider (ISP).
Function
Command
After entering IVR mode, dial 114.
After hearing “Enter value”, dial 1 (select static IP)
Select a Static IP
IP address Settings
After entering IVR mode, dial 111. After hearing “Enter value”,
enter your IP address and # (speed up dialing).
Example: If the IP address is 192.168.1.200, dial 192*168*1*200#.
Subnet Mask Settings
After entering IVR mode, dial 112. After hearing “Enter value”,
enter your subnet mask and # (speed up dialing).
Example: If the mask value is 255.255.255.0, dial 255*255*255*0#.
Default Gateway Setting
After entering IVR mode, dial 113. After hearing “Enter value”,
enter your default OR SERIES’s IP address and # (speed up
dialing).
Example: If the Default Gateway is 192.168.1.1, dial 192*168*1*1#.
Save Settings and Restart
Dial 509 to save settings.
Dial 118 to reboot OR SERIES.
Wait for about 40 seconds for restart, and then enter 101 to
check if the IP address is retained. If the IP address is not
repeated, OR SERIES has not been successfully connected to
the Internet, please check if the cable connection and IP address
are correct.
Dynamic IP (DHCP) Settings
After entering IVR mode, dial 114.
You will hear “Enter value”,
Dial 2 to select DHCP.
Dial 509 to save settings.
Dial 118 to reboot OR SERIES.
Wait for about 40 seconds for restart, and then enter 101 to check if the IP address is retained. If
the IP address is not repeated, OR SERIES has not been successfully connected to the Internet,
please check if the cable connection is correct.
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Terminal Adapter
ADSL PPPoE Settings
NOTE: Before setting PPPoE, you must have PPPoE account (121) and PPPoE password
(122) provided by your local Internet Service Provider (ISP).
Select a PPPoE
After entering IVR mode, dial 114.
You will hear “Enter value”.
Dial 3 to select PPPoE.
Set PPPoE account
After entering IVR mode, dial 121.
You will hear “Enter value”.
Enter account number and # (speed up dialing).
Example: If the account is “84943122 @ hinet.net”, please enter 08 04 09 04 03 01 02 02 71 48 49 54 45 60 72
54 45 60 #.
Please note that it is necessary to enter two digits for each character/number; for example,
enter 01 for 1 and 11 for A.
PPPoE Password Setting
After entering IVR mode, dial 122.
You will hear “Enter value”.
Enter password number and # (speed up dialing).
Example: If the password is “3ttixike”, please enter “03 60 60 49 64 49 51 45#”.
Save Settings and Restart
Dial 509 to save settings.
Dial 118 to reboot OR SERIES.
Wait for about 40 seconds for restart, and then enter 101 to check if the IP address is retained.
If the IP address is not repeated, OR SERIES has not been successfully connected to the
Internet, please check if the cable connection, account, or password are correct.
Recorded Voice File
OR SERIES allows users to record their incoming call greeting messages, when calling via
FXO.
After entering IVR mode, dial 132. After hearing “Enter value”, record the incoming call
greeting message. To end recording, simply hang up.
After recording, to listen to the recorded message, press 131. Press 133 to save the message.
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Terminal Adapter
PPPoE Character Conversion Table
Number Input Key
Upper Case
Input Key
Letter
Lower Case
Input Key Symbol Input Key
Letter
00
11
41
71
01
12
42
•
72
02
13
43
73
03
14
44
74
04
15
45
75
05
16
46
76
06
17
47
77
07
18
48
78
08
19
49
79
09
20
50
80
21
51
81
22
52
82
23
53
83
24
54
84
25
55
85
26
56
86
27
57
87
28
58
88
29
59
89
30
60
90
31
61
91
32
62
92
33
63
93
34
64
94
35
65
95
36
66
96
97
98
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Terminal Adapter
4. Setting a OR SERIES with WEB Browser
OR SERIES allows users to make settings with a web browser. Activate your browser, and then
enter OR SERIES’s IP address (e.g. http：//192.168.8.254.) in the Location (for IE) or Address
field and press Enter. And you will see the WEB page as following figure. You can also dial 101 on
your phone’s keypad to inquire the current WAN Port IP address. The factory default LAN Port IP
address is 192.168.8.254.
Instructions
Open a web browser.
Enter OR SERIES’s LAN Port IP address (Default is 192.168.8.254) in Address field (for IE)
and make sure your PC is correctly connected to OR SERIES and IP addresses are also in
the same network.
The following registration screen will appear (The factory default settings for Login ID and
Password are left blank).
Change the default settings of Administrator’s Name, Password and Web UI Login ID,
Password in Login Account.
After completing and confirming the settings, some of the settings will take effect immediately.
But network related settings would take effect after OR SERIES is restarted. Please go to
System Operation to save the settings before restarting OR SERIES.
For security concern, OR SERIES only accepts one user to login WEB UI for configuration at a
time. Please remember to logout or restart OR SERIES before leaving.
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Terminal Adapter
Basic Network Settings
WAN, SIP and Phone Book are basic Network settings. You have to choose one of SIP and
Phone Book for registration. It is recommended to use SIP if you’re not sure which one to use.
After completing these settings, OR SERIES will be able to make VoIP calls.
WAN
WAN Configuration includes the method of obtaining IP, the setting of DNS (Domain Name
Server), etc.
Setup Hint:
1. Choose the correct access type that your ISP supports.
2. Set DNS (Domain Name Server) to Auto if you don’t know the DNS server address.
3. WAN QoS, Clone MAC and VLAN are optional.
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Terminal Adapter
It is the IP address of WAN port.
When you use DHCP or PPPoE to obtain IP address, you can check the Current WAN IP
Address field to know if OR SERIES has obtained IP address. N/A is no IP address.
IP Configuration
There are five methods of obtaining a WAN port IP address:
1. DHCP, means a Dynamic IP (Cable Modem)
2. Static IP
3. PPPoE (Dialup ADSL)
4. PPTP.
5. BigPond Cable
Using DHCP and PPPoE for obtaining an IP address may vary. If you are not familiar with the
network connection, please contact your local ISP.
Item
Description
DHCP
This is the default Internet access type. It will obtain IP address
from DHCP server of ISP.
Static IP
If OR SERIES is connected to a router that request OR
SERIES to have a static IP address, fill in the proper IP
address, Subnet Mask and Default Gateway (IP address of the
router).
PPPoE
Enter PPPoE account and password and make sure they are
correct.
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Terminal Adapter
IP Configuration (continued)
Item
Description
PPTP
Enter IP address, Subnet mask, PPTP server address, PPTP
ID and Password. It only obtains an IP address from PPTP
server and does not provide VPN function.
BigPond Cable
Enter user name and password. Login Server is option.
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Terminal Adapter
Domain Name Server (DNS)
OR SERIES will look up the IP address from the DNS provided by ISP while it is accessing
another VoIP devices or computer with a hostname. In most cases ISP servers will assign DNS
information to OR SERIES automatically.
Note: Without correct DNS setting OR SERIES may not be able to provide services.
Item
Description
Domain Name Server
Assignment
Auto
ISP.
Manual
Domain Name Server IP
Enter correct DNS server address
: OR SERIES uses DNS IP automatically provide by
: Use it if OR SERIES has a static IP address
VLAN
It is optional. It works with the Router or Switch that supports VLAN.
Note: Please do not change anything here unless requested by your ISP.
Item
Description
Enable VLAN Tagging
It is to tag the packets for VLAN Router or Switch identifying.
VLAN ID
It is to assign uniquely a user-defined ID to each packet.
Priority
It is the proprietary to VLAN Router or Switch.
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Terminal Adapter
WAN QoS
It is effective when OR SERIES is as a Bridge. Using QoS is able to ensure that voices have
higher priority than data flow, and it also restricts upstream data flow.
Item
Description
QoS
It is to set an external bandwidth to ensure sound quality
during transmission (When this function is enabled, the voice
packet has the highest priority to ensure telecommunication
quality while less bandwidth is assigned for data transmission).
ToS (Type of Service)/
DiffServ(DSCP)
The voice packet has the highest priority to ensure
telecommunication quality, and the larger the value you set,
the higher priority you will get.
Clone MAC
Some Internet Service Providers (ISP) assigns the IP via the MAC (Media Access Control)
Address. Click the Clone button to copy the MAC address of the Ethernet Card installed in the
computer used to configure the device. It is only necessary to fill in the field if required by your
ISP.
Your MAC Address will be blank as you log in through WAN port.
Note: Please do not change anything here unless requested by your ISP.
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Terminal Adapter
SIP
In this section, you should have one or more VoIP service accounts from Voice Service
Provider(VSP) and enter the related parameters of VSP.
Setup Hint:
1. Enter the SIP telephone number.
2. Tick register and invite with ID/Account.
3. Enter user ID/Account and password.
4. Enter the VSP IP address or URL (Uniform Resource Locator) and VSP listen port
number.
5. Enter SIP domain if the VSP address is not IP.
6. OutBound Proxy is optional.
Accounts Settings
Item
Description
Number
Enter the SIP telephone number assigned by your VSP
Register
Tick the check box to register the number before making calls.
Invite with ID / Account
Tick the check box if SIP server requests authentication.
User ID / Account
Password
Authentication information required by VSP
FXS Group
Select group-hunting priority. When there is an incoming call,
OR SERIES will automatically assign an unassigned call
according to Hunting Priority. If Line 2 does not want to be set
as an assigned line to receive any inbound calls, set it to “0”.
Note: There are two ways to register if you have one more accounts.
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Terminal Adapter
Registration by each line: If your VoIP account and password are individual, the settings
should be as below.
Registration by FXS Representative Number: If you have one VoIP account and password,
the settings should be as below.
Item
Description
Number
Enter the SIP telephone number assigned by your VSP
Register
It is to register this number before making calls
Invite with ID / Account
Tick the check box if SIP server request authentication.
User ID / Account
Password
Authentication information required by VSP
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VSP (Voice Service Provider) Settings
Note: If you fail to make a call, please contact your VSP.
Item
Description
Use DNS SRV
Tick the check box to make OR SERIES register to VSP.
DNS SRV Auto Prefix
The default is that OR SERIES will use
_sip._udp.domain.com to query IP. If you untick the check
box, OR SERIES will use domain.com to query IP.
Proxy Fallback Interval
Defines the time that OR SERIES registers to the main server
if OR SERIES has registered to the secondary server.
Item
Description
Enable Support of SIP Proxy
Server / Soft Switch
Tick the check box to make OR SERIES register to VSP.
Enable SIP Proxy 1
SIP Proxy 1 is the main server. When SIP Proxy 1 and 2 are
enabled, OR SERIES will register to SIP Proxy 2 which is a
backup server after all lines are failed to register to SIP Proxy
1.
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Item
Description
Proxy Server IP/Domain
Enter the SIP Server IP address or URL (Uniform Resource
Locator)
Proxy Server Port
Enter the Proxy Server listen port number (The default value is
5060).
Proxy Sever Realm
Enter the correct registered Proxy Server Realm name to avoid
registration failure. Set it by default if you are not sure.
TTL (Registration interval)
The interval that OR SERIES will report to the Proxy Server
periodically. Set it by default if you are not sure.
SIP Domain
Enter SIP Domain (URI) if required by VSP(Voice Service
Provider).
Use Domain to Register
Tick the check box to make OR SERIES register with SIP
Domain; otherwise it will register with SIP Server IP address.
Outbound Proxy
This is optional. An outbound proxy server handles SIP call signaling as a standard VSP would.
Furthermore, it receives and transmits phone conversation traffic(media) between two talking
VoIP devices. This option tells OR SERIES to send and receive all SIP packets to the destined
outbound proxy server rather than the remote VoIP device. This might help VoIP calls to pass
through any NAT protected network without additional settings or techniques.
Note: Make sure your Voice Service Provider requires this feature before enable it. VSP
gives parameters.
Item
Description
All Call through OutBound
Proxy
Tick the check box to make OR SERIES register to OutBound
Proxy Server / Soft Switch.
OutBound Proxy IP/Domain
Enter the OutBound Proxy IP address or URL (Uniform
Resource Locator).
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E.164
This is optional. E.164 is to replace number that you dial out into [country code]+[area code] +
[destination number]. This is done automatically by OR SERIES without changing user dialing
habit.
If your VSP accept only E.164 numbering rule in SIP invite. You will have to fill information in the
current VoIP IAD according to the dialing habit. These information are, what will user dial when he
tries to make international call? What is the country code of the VoIP IAD? What will user dial
when he wants to dial long distance call? What is the local area code? If all information are filled,
the dial out invite will be changed from [destination number] to [country code]+[area
code]+[destination number].
Note: If you fail to make a call, please contact your VSP.
Item
Description
International Call Prefix Digit
Enter the International call prefix.
Country Code
Users please select the desired country code.
Long Distance Call Prefix Digit The long-distance prefix digit for making a long-distance call.
Area Code
Enter the area code.
Item
Description
To Invite Proxy
Invite Proxy to follow the E.164 rule.
Transform to Transit Out
The call from FXO to PSTN follows the E.164 rule. It applies to
one-stage dialing. (Only OR SERIES has this function).
ENUM Header Exception
Defines OR SERIES not to change the prefix..
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Example of To Invite Proxy:
International Call Prefix Digit: 00
Country Code: 1
Long Distance Call Prefix Digit: 0
Area Code: 567
ENUM Head Exception: 070
Phone Number Dialed
By The User
The True Phone Number
Description
Dialed By Gateway
23456789
1 567 23456789
Exclude International Call Prefix Digit and
Long Distance Call Prefix Digit.
Add Country Code(1) and Area Code(567).
0 223 98765432
1 223 98765432
Include Long Distance Call Prefix Digit.
Delete Long Distance Call Prefix Digit(0) and
add Country Code(1).
00 852 987654321
852 987654321
Include International Call Prefix Digit.
Delete International Call Prefix Digit(00).
070 12345678
070 12345678
Include ENUM Head Exception(070).
Do not change the number.
Example of Transform to Transit Out:
International Call Prefix Digit: 00
Country Code: 1
Long Distance Call Prefix Digit: 0
Area Code: 567
ENUM Head Exception: 070
Phone Number Dialed The True Phone Number
To FXO From the
Dialed By Gateway From Description
Remote End
FXO to PSTN
23456789
Include Country Code(1), Area Code(567).
Delete Country Code and Area Code.
1 765 8527413
0765 8527413
Include Country Code(1) and exclude Area
Code(567).
Delete Country Code(1) and add Long Distance
Call Prefix Digit(0).
852 987654321
00 852 987654321
Exclude Country Code.
Add International Call Prefix Digit(00).
070 12345678
070 12345678
Include ENUM Head Exception(070).
Do not change the number.
1 567 23456789
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Phone Book
Some peer information needs to be added to this section before OR SERIES makes peer-to-peer
calls.
Phone Book Manager: VoIP devices register to Phone Book Manager. When you make calls
from OR SERIES to the peer VoIP device, it will get the number and IP from Phone Book
Manager.
Phone Book: Some peer information is added to Phone Book. OR SERIES can set up and store
100 phone numbers into Phone Book and provide an IP address query when calling to other VoIP
devices.
Using Phone Book Manager
Item
Description
Register to
Phone Book Manager
Tick the check box to register to the Phone Book Manager.
VoIP failure announcement
If OR SERIES fails to register to the Phone Book Manager, it
will play a voice announcement when FXS is off-hook.
Gateway Name for Phone
Book Manager
The alias registered with the Phone Book Manager.
Phone Book Manager Login
Password
Enter the registered password that is the same with Phone
Book Manger.
Phone Book Manager IP /
Domain
Enter the IP address for the Phone Book Manager. It supports
URL (Uniform Resource Locator).
Phone Book Manager Listen
Port
The protocol communication port for transmitting signals
between the Phone Book Manager and OR SERIES.
Note: Make sure that Phone Book Manager Login Password and Phone Book Manager
Listen Port are same as that of the Phone Book Manager.
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Using Phone Book
OR SERIES can set up and store 100 phone numbers to a phone book. If there is no Phone
Books Manager exiting in private network, all OR SERIESs in a group have to set up each
gateway’s number one by one to communicate with each other.
Note: If the VoIP peer is in a NAT network, the listen port may vary or unreachable
depend on settings of that NAT router.
Item
Description
Gateway Name
Enter an easy-to-remember name to identify each VoIP device
listed in the phone book. This parameter is optional.
Gateway Number
Enter the telephone number of other VoIP device.
IP/Domain Name
Enter the IP address or URL of other VoIP device.
Port
Enter the listen port of other VoIP devices.
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Basic Voice Services
OR SERIES supports some voices such as display Caller ID, call forwarding, call hold, call
transfer, call-waiting, three-way calling, Emergency No., etc.
Caller ID
In this section, it allows you to set Caller ID generation.. There are two type of FSK Caller ID.
Choose the proper type for you.
Item
Description
FXS Caller ID Generation
Tick the check box to display the phone number of the calling
party on your phone set when there is an incoming call.
FXO Caller ID Detection
Tick the check box to detect Caller ID delivered from PSTN
port.
Detection Level
It is the gain volume that could be adjusted while detecting
caller ID.
FSK Caller ID Type
In most cases, Bellcore is preferred in North America and ETSI
in Europe.
Note: You have to enable “Hot Line->Wait for Caller ID before FXO / Trunk pick up” to
ensure detect Caller ID correctly.
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Transit In Caller ID Strip / Replace
You can change the information of the calling party while making calls to Internet.
Note: Available in OR SERIES only.
Item
Description
Scan code
Defines the rule of the Caller IDs detected by FXO. It can be a
prefix or a full number.
Substitude
Defines the changed Caller ID while making calls to Internet by
FXO. It will change two places of displaying the caller id. One
is From-Header Display Name, and the other one is Remote
Party ID Display Name.
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Hot Line
Item
Description
Enable
All lines are enabled by default. Untick the check box to disable
it if the line is not in use (Pause Function).
Hot Line
While picking up the phone, OR SERIES will automatically dial
the assigned Hot Line number. At the moment, dialing any
number out is denied.
Hot Line No.
Enter the Hot Line number for an automatic dial.
Warm Line (Hot Line Delay)
A user can dial any number within the time. After the time
expires, OR SERIES will divert incoming calls from an outside
line to the Hot Line Number.
PSTN Busy-Out with FXS
Pick-up
OR SERIES will reject a call from FXO while FXS is getting
DTMF. If you would like to disable it, set the value as “0”.
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Item
Description
As making a VoIP call, a waiting call from PSTN is allowed.
Before starting, do the following settings first:
VoIP Call Allow PSTN In
1. Tick the check box to enable VoIP Call Allow PSTN In.
2. Tick the check box to enable Call Hold.
(Calling Feature → Call Hold)
3. Set PSTN Busy-Out With FXS Pick-Up as 0.
As making a PSTN call, a waiting call from VoIP is allowed.
Before starting, do the following settings first:
PSTN Call Allow VoIP In
1. Tick the check box to enable PSTN Call Allow VoIP In.
2. Tick the check box to enable Call Hold and Call Waiting.
(Calling Feature → Call Hold and Call Waiting)
Wait for Caller ID before
FXO / Trunk pick up
It is to detect caller ID from PSTN port.
Item
Description
Enable
All lines are enabled by default. Untick the check box to disable
it if the line is not in use (Pause Function).
Hot Line
While picking up the phone, dialing any number out is denied,
since OR SERIES will automatically dial the assigned Hot Line
number.
Hot Line No.
Enter the Hot Line number for an automatic dial.
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Item
Description
Warm Line (Hot Line Delay)
A user can dial any number within the time. After the time
expires, OR SERIES will divert incoming calls from an outside
line to the Hot Line Number.
Item
Description
Enable
All lines are enabled by default. Untick the check box to disable
it if the line is not in use (Pause Function).
Hot Line
While picking up the phone, dialing any number out is denied,
since OR SERIES will automatically dial the assigned Hot Line
number if set Warm Line to 0.
Hot Line No.
Enter the Hot Line number for an automatic dial.
Warm Line (Hot Line Delay)
A user can dial any number within the time. After the time
expires, OR SERIES will divert incoming calls from an outside
line to the Hot Line Number.
Dial-Out Prefix
It is the number dialed automatically by FXO port before the
FXO interface diverts a VoIP call to PSTN.
FXO Line Default Dial-Out
Before starting to configure, you should set FXO Line VoIP call
in option to Default Dial-Out. When FXO receives a call from
VoIP, it will dial to PSTN with the default number.
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Item
Description
Caller Indicate Dial-Out: When there is a call from WAN
FXO Hunting VoIP call in
option
interface to FXO port, it will dial to PSTN with the number
assigned in SIP packet.
Default Dial-Out: When there is a call from WAN interface to
FXO port, it will dial to PSTN with the number filled in FXO Line
Default Dial-Out field.
Select the greeting type. When FXO receives an inbound call,
Trunk Incoming Prompt Voice the caller can hear the greeting. (If you would like to record a
voice file, you must use the IVR 132 function).
Custom Greeting Upload /
Backup
It is to upload or backup the recorded voice file. The format
must be G.723.1.
When FXO is connected to different PBX or PSTN, or under
Enable FXO/Trunk Extension
special circumstances, the caller can choose one of them to
Number
call out. It MUST be ticked while registering to a Proxy.
Pick up Line by Dialing
Extension Number
When there is a call from WAN interface and assigned FXO
extension number, FXO goes off-hook and waits for the caller
to dial the number to PSTN. It MUST be enabled while
registering to a Proxy.
Wait for Caller ID before FXO /
Detect caller ID from FXO port.
Trunk pick up
Transit in Busy Tone Limit
Define the duration of a busy tone before FXO hook-on. Notify
the caller from PSTN that this call is finished.
Detect FXO Line Presence
Tick the check box to detect the line presence that FXO port is
connected to PBX or a PSTN line. Untick the check box to
disable this function if it mis-detect line presence on FXO port
while ringing.
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Calling Features
OR SERIES provides Call Forward, Call Hold, Call Transfer and Call Waiting.
OR SERIES also provides Three-Way Calling based on Nortel Soft Switch. It also works with the
conference call supported by VSP.
Item
Description
Do Not Disturb
Tick the check box to reject all incoming calls from WAN
interface. It allows only to make an outgoing call.
Unconditional Forward
All incoming calls will be forwarded to the Forwarding Number
automatically. If the call is forwarded to FXO port, FXO is
off-hook instead of dialing out.
Busy Forward
It is to forward the incoming call to Forwarding Number when
the line is busy.
No Answer Forward
It is to forward the incoming call to Forwarding Number after the
time expires without answer.
Call Hold
Tick the check box to enable call hold for specific FXS port.
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Item
Description
Call Transfer
Tick the check box to enable call transfer for specific FXS port.
Call Waiting
Tick the check box to enable call-waiting for specific FXS port.
Three-Way Calling /
Service ID
It is for conference all based on Nortel Soft Switch and must
work with Proxy Server that supports Three-Way Calling
service.
Calling Feature Instructions:
Call Hold: While pressing FLASH button on the phone. The call is held.
Call Transfer: Ongoing call will be put on hold after FLASH button pressed on local phone
set. Meanwhile, the local user can dial out to another number after dial tone observed. After
the handset is back on the hook, the call on hold will then be transferred to the new call
regardless of the status of the new call. If wrong number is dialed for the new call, just press
the FLASH button to get back the call on hold. In another case, if the local user does not
hang up the phone after new call sets up, press FLASH button to switch between the first
call and the new call. If a phone set is connected directly to the FXS port of OR SERIES
and not functioning to FLASH, please adjust the settings in Flash Detect Time in category
“Line Settings”.
Call Waiting: When you are on the phone and a second call comes in, you will hear
“Beep-Beep” tone to notify that there is another call. Press the FLASH button to hold the
first call and take the second call. After finishing the second call, press the FLASH button
again to take the first call.
Example of a Three-Way calling:
1.
2.
3.
4.
Or
1.
2.
3.
4.
Alex calls Bob, Bob answers the call.
Alex presses Flash and calls Coral (Bob is on hold), Coral answers the call.
Alex dials *61 and then presses Flash.
Thus the conference call is established.
Alex calls Bob, Bob answers the call.
Coral calls Alex (Call Waiting), presses Flash and talks to Coral.
Alex dials *61 and then presses Flash.
Thus the conference call is established.
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Calling Features - Advanced Setting
OR SERIES provides advanced settings: Call Pickup and Automatic Redial.
Note: Automatic Redial is only used for the latest call (NO two calls reserved for
Automatic Redial). The duration of Automatic Redial is set to 10 minutes. If the callee is
still not available after 10 minutes, OR SERIES will not dial again.
Function Code
Description
*40#
Call Pickup: The user can use the function of call pickup to
answering others calls. When one of FXS is ringing and there
is no one to answer the call. The user can use another FXS
port to pick up the ringing call with this function code.
For Example: If Alice calls Bob (9901701) who does not
answer. Carol can pick up the call by dialing *40 9901701#.
*41#
Automatic Redial: The remote party is initially busy when you
call. Hang up the phone and then pick up to dial *41# and then
hang up. You are hearing a ring tone when the remote party is
available. You are alerted and then pick up the phone to wait
for the remote party answering.
*42#
It is to cancel the latest automatic redial function.
*43#
It is to query how long shall OR SERIES wait to redial (ms).
*44#
It is to adjust the duration of waiting for automatic redial.
Method: Dial *44 + Expiry Time#
*45#
It is to query the duration of waiting for automatic redial (ms).
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PSTN Control
Note: Available in OR SERIES only
This rule only applies to one-stage dialing. It is to replace the prefix number before diverting the
number to PSTN dial out. It also restricts the number by checking the prefix number.
Example: If you transit out with 01907123456, OR SERIES will replace the number to 190601
907123456. If you transit out with 008621123456, OR SERIES will replace it with 190200
8621123456.
Item
Description
Trunk Dial Out Verify
Trunk Dial Out Replace
Before the number is diverted to PSTN by FXO port, OR
SERIES will verify the numbers in Trunk Dial Out Verify filed
and replace them with the numbers in Trunk Dial Out Replace
field.
Trunk Dial Out Deny
OR SERIES will deny the call with the leading number filled in
this column.
Emergency No
Emergency numbers is defined here. You can call out to PSTN (Telco line) with the numbers that
your VSP does not support (i.e. Toll free service numbers).
Note: Available in OR SERIES only
Item
Description
Enable
Tick the check box to make this entry effective.
Scan Code
Fill in the leading number for OR SERIES to scan or the full
number.
User Dial Length
Set the total digit count of user dialed.
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Advanced Network Settings
OR SERIES provides interface for advanced network settings to enhance your network security.
LAN
This is about LAN configuration. There are LAN interface mode that is to set OR SERIES as a
router or a bridge, LAN IP and subnet mask, DHCP settings.
LAN interface mode
Item
Description
Router
OR SERIES serves as a router with NAT.
Bridge
OR SERIES serves as a bridge between WAN port and LAN
port without NAT. (LAN default gateway will still be accessible
for configuration).
Item
Description
Bridge Mode VLAN Tagging
It is to tag the packets for VLAN Router or Switch identifying
when OR SERIES serves as a Bridge.
VLAN ID
It is to assign uniquely a user-defined ID to each packet.
Priority
It is the proprietary to Router or Switch.
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LAN Settings
Note: OR SERIES LAN port IP address cannot be in the same section as the NAT LAN port
IP address.
Example: If the LAN IP address of the Internet Sharing Device is 192.168.8.1, then OR SERIES’s
LAN IP address cannot be in the range between 192.168.8.1 ~ 192.168.8.254. You can set
192.168.99.254 for the LAN IP.
Item
Description
LAN IP/LAN default Gateway LAN Port IP address and the subnet mask value. Please note
that OR SERIES is built with NAT
Subnet mask
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DHCP Settings
Item
Description
Enable DHCP Server
Tick the check box to enable DHCP server service of OR
SERIES.
IP Pool Starting Address
IP Pool Ending Address
The first IP address to be assigned to DHCP clients.
The last IP address to be assigned to DHCP clients.
IP Pool Uses Other Default
GW
Tick the check box to give DHCP client the other default
gateway.
IP Pool Default Gateway
IP Pool Subnet mask
Assign the default gateway and subnet mask to DHCP client.
Lease Time
The valid period of an assigned IP address.
Domain Name Server
Assignment
Domain Name Server IP
Auto
Manual
: Assign DNS obtained from WAN port to the DHCP
clients.
: Manually assign DNS for DHCP clients.
It is to manually assign DNS to DHCP client, a correct DNS IP
address must be filled.
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NAT Traversal
If OR SERIES is set up behind an IP sharing device or a router, you can select either the NAT or
STUN protocol.
Note: NAT IP/Domain must be the same with Hostname (in DDNS page), if OR SERIES is
behind a NAT Server that uses a dynamic IP and registers to DDNS.
The ports that need to set the Virtual Server Mapping in the NAT server are below.
1. Listen Port (UDP): 5060 is default.
2. RTP Port (UDP): 9000~9001. These ports are used for telecommunication.
3. Http Port (TCP): The default is 80.
Item
Description
NAT Public IP
Tick the check box to enable NAT.
NAT IP/Domain
Enter the NAT Server IP address (Real External IP address of
NAT Server) then fill in the URL (Uniform Resource Locator).
Enable STUN Client
Tick the check box to use STUN protocol prevents problems
with setting the IP sharing function, but some NAT do not
support this protocol.
STUN Server IP/Domain
STUN Server Port
Enter the STUN server IP address and Listen Port number.
Enable UPnP Control Point
It only works when the NAT server supports UPnP. Tick the
check box to enable OR SERIES to pass through the NAT
server.
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DDNS
These settings are only necessary when OR SERIES is set up behind a NAT that uses a dynamic
IP address and do not support DDNS.
First of all, you need to apply an account from one of the servers. OR SERIES allows users to
choose one of DynDNS, TZO, 3322.org, PeanutHull or a private server.
Item
Description
Register to DDNS
Tick the check box to enable DDNS and choose a DDNS
Server as below to register.
Item
Description
Server address
Enter the IP address or URL (Uniform Resource Locator) of the
DDNS Server.
Hostname
The URL of OR SERIES (or NAT) – provided by a domain
name registration providers. (e.g. www.dyndns.org).
Login ID
Password
The ID and password are used to login the DDNS server.
Behind NAT
Tick the check box to enable this function only when OR
SERIES is set up behind a NAT.
Custom
Only DynDNS has. Tick the check box if you have a custom
hostname in DynDNS.
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Caller Filter
This function is used to allow or deny SIP Invite from the list. The IP address of VSP is allowed
while registering to VSP.
Item
Description
Allow
Deny
Choose the IP addresses in the table are allowed to call in or
deny.
Enable
Tick the check box to make this effective.
Filter IP address
Enter the start IP you would like to allow/deny.
Subnet mask
Enter the subnet mask you would like to allow/deny.
PPTP Client
This is optional. ISP gives all parameters.
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SIP Advanced
In this section, you can set the listen port and RTP port of OR SERIES.
There are some parameters with VSP (Voice Service Provider).
Session Timer: It is to identify the connection of a session which is defined in RFC 4028.
SIP Timeout Adjustment: It is to set SIP message resend time and maximum response time.
Supplementary Features: Other features work with VSP (Voice Service Provider).
Item
Description
Listen Port UDP
The listen port of OR SERIES.
RTP Starting Port UDP
The initial value of port number for transmitting voice data
among OR SERIES(s). Each line requires 2 ports
(RTP/RTCP). It is not necessary to change these.
For example, if the starting port is 9000, then Line 1 is using
9000 (RTP) and 9001 (RTCP), and Line 2 is using 9002 and
9003.
Session Timer
Item
Description
Session Expiration
It is to avoid the billing of abnormal dropping the call because
of Internet. The default is disabled.
Session Refresh Request
The method of refreshing for Session Timer.
Session Refresher
The role OR SERIES plays in Session Timer. UAS is an
originator, and UAC is a replier.
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SIP Message Timeout Adjustment
Item
Description
SIP Message Resend Timer
Base
SIP packet will resend if response dose not arrive in the base
time set in this column. The max of resend time is 4 sec.
It will send again at "base time" *2, and send again at
"base time" *2 *2. Resend will stop/restart when total resend
20sec has reached.
Max. Response Time for Invite If the remote party does not reply in the set time after the first
invite, this call is failed.
SIP Proxy Server / Soft Switch Settings
Item
Description
VoIP failure announcement
As soon as the registration to proxy server is failed, OR
SERIES will drive IVR system to play out failure
announcements for the user.
Bind Proxy Interval for NAT
OR SERIES will always send two packets in N seconds to VSP
to bind the tunnel. The VSP can always send SIP packets to
OR SERIES that is setup behind an NAT.
Initial Unregister
OR SERIES will send un-register packet to VSP as it is
initialing.
Support Message Waiting
Indication (MWI)
Tick the check box to enable voice mail function. OR SERIES
will play a tone to notify user if there are messages in the voice
mail.
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Item
Description
MWI Subscribe Interval
The subscribe interval is for OR SERIES check of the voice
mail.
Supplementary Features
Note: Enable Anonymous Caller ID or Anonymous Transit in W/O Caller ID, you may be
unable to make a call since OR SERIES doesn’t send the number for authorization.
Item
Description
Anonymous Caller ID (CLIR)
Tick the check box to dial out with “anonymous” as caller
identification by FXS. Sometimes it may require proxy server to
identify by Caller ID, so disable it while the call is failed.
VoIP Call Out Notification
OR SERIES will play a tone to notify the call is through VoIP.
Enable Built-in Call Hold
Music
The default setting is that when receiving a call hold request,
OR SERIES will play music on hold. Untick the check box to
disable the function.
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Item
Description
Use Second CPT after SIP
registered
This function is usually applied when the user set VoIP as the
primary path for outgoing calls and PSTN as the backup. OR
SERIES will generate a different set of tones to inform the user
that VoIP is in service. When VoIP call is failed, the user will
hear PSTN tones instead of the second set CPT. (for CPT
settings, refer CPT Parameters Table)
Enable Non-SIP Inbox Call
Tick the check box to disable Non-SIP inbox call if all calls
need to go through VSP.
Delay PSTN Hangup
Detection
The default is that OR SERIES detects dully if PSTN hangs up.
Tick the check box to make OR SERIES detect PSTN status
sensitively.
Enable P-Asserted
Tick the check box to use anonymous caller ID for protection if
the SIP proxy has this function.
Privacy Type
Privacy requested for Third-Party Asserted.
Invite URL need ‘user=phone’
It will contain “user=phone” in Invite Packet. Some Proxy
Servers can’t accept “user=phone”, just disable it.
Reliability of Provisional
Responses
Defines a type of SIP responses that provide information on
the progress of the request processing. Tick the check box to
achieve reliability for provisional responses.
Compact Form
Defines the header packet size will be shortened with signaling
compression to enhance bandwidth. Tick the check box to
enable this function.
Defines from which part of the SIP packet will the gateway
obtain caller ID. There are several places where you can put
your caller ID.
SIP Caller ID Obtaining
Remote-Party-Id Display Name: It is locate at SIP→
Remote-Party-ID→Before [ Click on Control Panel -> Double-click on Network and Dial-up Connection ->
Click on Open Local Area Connection ->
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Click Properties.
Highlight Internet Protocol (TCP/IP) and then click Properties.
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Select Use the following IP Address. Set IP address, Subnet mask and Default gateway. The IP
Address must be within the same range as OR SERIES (If the IP Address of OR SERIES is
192.168.8.254. You can assign 192.168.8.100 for your PC). Then, enter the DNS server IP
address (varies in different networks. consult your ISP’s service for information). Click on the OK
button to make settings take effect.
87
SIP OPERATION MANUAL
FEDERAL COMMUNICATIONS COMMISSION INTERFERENCE STATEMENT
This equipment has been tested and found to comply with the limits for a Class B digital
device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide
reasonable protection against harmful interference in a residential installation. This
equipment generates, uses and can radiate radio frequency energy and, if not installed
and used in accordance with the instructions, may cause harmful interference to radio
communications. However, there is no guarantee that interference will not occur in a
particular installation. If this equipment does cause harmful interference to radio or
television reception, which can be determined by turning the equipment off and on, the
user is encouraged to try to correct the interference by one or more of the following
measures:
– Reorient or relocate the receiving antenna.
– Increase the separation between the equipment and receiver.
– Connect the equipment into an outlet on a circuit different from that to which the
receiver is connected.
– Consult the dealer or an experienced radio/TV technician for help.
CAUTION:
Any changes or modifications not expressly approved by the party responsible for
compliance could void the user's authority to operate the equipment.
This device complies with Part 15 of the FCC Rules. Operation is subject to the following
two conditions:
(1) This device may not cause harmful interference and
(2) This device must accept any interference received, including interference that may
cause undesired operation.
RF exposure warning ·
This equipment must be installed and operated in accordance with provided instructions
and the antenna(s) used for this transmitter must be installed to provide a separation
distance of at least 20 cm from all persons and must not be co-located or operating in
conjunction with any other antenna or transmitter. End-users and installers must be
provide with antenna installation instructions and transmitter operating conditions for
satisfying RF exposure compliance.

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