Octtel communication OR222W Wireless VOIP Gateway User Manual

Octtel communication Co., LTD Wireless VOIP Gateway Users Manual

Users Manual

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
SIP OPERATION MANUALTerminal Adapter 11. 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).
SIP OPERATION MANUALTerminal Adapter 2 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.
SIP OPERATION MANUALTerminal Adapter 3Phone: 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.
SIP OPERATION MANUALTerminal Adapter 4 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.
SIP OPERATION MANUALTerminal Adapter 5 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)
SIP OPERATION MANUALTerminal Adapter 6 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.
SIP OPERATION MANUALTerminal Adapter 7OR 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
SIP OPERATION MANUALTerminal Adapter 8 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.
SIP OPERATION MANUALTerminal Adapter 9Instructions  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) enter 192*168*1*3 (IP number). 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.
SIP OPERATION MANUALTerminal Adapter 10 IVR Functions Table: Function Code Description Remark 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 Use in conjunction with function code 114, select 1 for a Static IP function.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 Set/Query whether or not to login Phone Book Manager (0: Disable 1:Enable) Must use 116/106, 117/107 in conjunction with each other. 066 Querying the connection to Phone books manager 118 Restart 121 Setting PPPoE Account 122 Setting PPPoE Password Use in conjunction with function code 114, select 3 for a PPPoE function 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. 109 Restoring factory default setting of IP The default of Static IP IP: 192.168.1.2 Mask: 255.255.255.0 Gateway: 192.168.1.254409 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
SIP OPERATION MANUALTerminal Adapter 11IP 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 Select a Static IP  After entering IVR mode, dial 114.  After hearing “Enter value”, dial 1 (select 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.
SIP OPERATION MANUALTerminal Adapter 12 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.
SIP OPERATION MANUALTerminal Adapter 13 PPPoE Character Conversion Table Number Input Key Upper Case Letter Input Key Lower Case Letter Input Key Symbol Input Key0 00 A 11 a 41 @ 71 1 01 B 12 b 42 • 72 2 02 C 13 c 43 ! 73 3 03 D 14 d 44 " 74 4 04 E 15 e 45 $ 75 5 05 F 16 f 46 % 76 6 06 G 17 g 47 & 77 7 07 H 18 h 48 ' 78 8 08 I 19 i 49 ( 79 9 09 J 20 j 50 ) 80 K 21 k 51 + 81 L 22 l 52 , 82 M 23 m 53 - 83 N 24 n 54 / 84 O 25 o 55 : 85 P 26 p 56 ; 86 Q 27 q 57 < 87 R 28 r 58 = 88 S 29 s 59 > 89 T 30 t 60 ? 90 U 31 u 61 [ 91 V 32 v 62 \ 92 W 33 w 63 ] 93 X 34 x 64 ^ 94 Y 35 y 65 _ 95 Z 36 z 66 { 96 | 97 } 98
SIP OPERATION MANUALTerminal Adapter 14 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.
SIP OPERATION MANUALTerminal Adapter 15Basic 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.
SIP OPERATION MANUALTerminal Adapter 16 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.
SIP OPERATION MANUALTerminal Adapter 17IP 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.
SIP OPERATION MANUALTerminal Adapter 18 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 : OR SERIES uses DNS IP automatically provide by ISP. Manual : Use it if OR SERIES has a static IP address Domain Name Server IP Enter correct DNS server 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.
SIP OPERATION MANUALTerminal Adapter 19WAN 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.
SIP OPERATION MANUALTerminal Adapter 20 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.
SIP OPERATION MANUALTerminal Adapter 21 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
SIP OPERATION MANUALTerminal Adapter 22 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.
SIP OPERATION MANUALTerminal Adapter 23Item 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 Serverperiodically. 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).
SIP OPERATION MANUALTerminal Adapter 24 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..
SIP OPERATION MANUALTerminal Adapter 25Example 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 Dialed By Gateway Description 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 To FXO From the Remote End The True Phone Number Dialed By Gateway From FXO to PSTN Description 1 567 23456789 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.
SIP OPERATION MANUALTerminal Adapter 26 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.
SIP OPERATION MANUALTerminal Adapter 27Using 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.
SIP OPERATION MANUALTerminal Adapter 28 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.
SIP OPERATION MANUALTerminal Adapter 29 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.
SIP OPERATION MANUALTerminal Adapter 30 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”.
SIP OPERATION MANUALTerminal Adapter 31Item Description VoIP Call Allow PSTN In As making a VoIP call, a waiting call from PSTN is allowed. Before starting, do the following settings first: 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. PSTN Call Allow VoIP In As making a PSTN call, a waiting call from VoIP is allowed. Before starting, do the following settings first: 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.
SIP OPERATION MANUALTerminal Adapter 32 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.
SIP OPERATION MANUALTerminal Adapter 33 Item Description FXO Hunting VoIP call in option Caller Indicate Dial-Out: When there is a call from WAN 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. Trunk Incoming Prompt Voice Select the greeting type. When FXO receives an inbound call, 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. Enable FXO/Trunk Extension Number When FXO is connected to different PBX or PSTN, or under special circumstances, the caller can choose one of them to 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 / Trunk pick up Detect caller ID from FXO port. 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.
SIP OPERATION MANUALTerminal Adapter 34 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 Numberautomatically. 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.
SIP OPERATION MANUALTerminal Adapter 35Item 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. Alex calls Bob, Bob answers the call. 2. Alex presses Flash and calls Coral (Bob is on hold), Coral answers the call. 3. Alex dials *61 and then presses Flash. 4. Thus the conference call is established. Or 1. Alex calls Bob, Bob answers the call. 2. Coral calls Alex (Call Waiting), presses Flash and talks to Coral. 3. Alex dials *61 and then presses Flash. 4. Thus the conference call is established.
SIP OPERATION MANUALTerminal Adapter 36 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).
SIP OPERATION MANUALTerminal Adapter 37PSTN 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 Replacefield. 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.
SIP OPERATION MANUALTerminal Adapter 38 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.
SIP OPERATION MANUALTerminal Adapter 39LAN 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 Subnet mask LAN Port IP address and the subnet mask value. Please note that OR SERIES is built with NAT
SIP OPERATION MANUALTerminal Adapter 40 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 Auto : Assign DNS obtained from WAN port to the DHCP clients. Manual : Manually assign DNS for DHCP clients. Domain Name Server IP It is to manually assign DNS to DHCP client, a correct DNS IP address must be filled.
SIP OPERATION MANUALTerminal Adapter 41NAT 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.
SIP OPERATION MANUALTerminal Adapter 42 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 domainname 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.
SIP OPERATION MANUALTerminal Adapter 43Caller 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.
SIP OPERATION MANUALTerminal Adapter 44 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.
SIP OPERATION MANUALTerminal Adapter 45SIP 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.
SIP OPERATION MANUALTerminal Adapter 46 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.
SIP OPERATION MANUALTerminal Adapter 47Item 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. SIP Caller ID Obtaining 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. Remote-Party-Id Display Name: It is locate at SIP→Remote-Party-ID→Before [<sip:] Remote-Party-Id User Name: It is locate at SIP → Remote-Party-ID → After [<sip:], Before [@] From-Header Display Name: The standard way is in SIP → Message Header → From → SIP Display info. Support URI Percent-Encoding (RFC 3986) It follows RFC 3986 to encode some letters as charactertriplet, consisting of the percent character "%" followed by the two hexadecimal digits representing that octet's numeric value.The unreserved characters that are not encoded are uppercase and lowercase letters, decimal digits, hyphen (or dash), period (or dot), underscore (or underline), exclamation, tilde, asterisk (star or multiplication), single quote, parenthesis, bracket, ampersand, equal, plus sign, dollar sign, comma, semicolon, question mark, slash, colon, at sign and back slash.
SIP OPERATION MANUALTerminal Adapter 48 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. 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.
SIP OPERATION MANUALTerminal Adapter 49Item Description 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. SIP CallerId Obtaining 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. Remote-Party-Id Display Name: It is locate at SIP→Remote-Party-ID→Before [<sip:] Remote-Party-Id User Name: It is locate at SIP → Remote-Party-ID → After [<sip:], Before [@] From-Header Display Name: The standard way is in SIP → Message Header → From → SIP Display info. Support URI Percent-Encoding (RFC 3986) It follows RFC 3986 to encode some letters as charactertriplet, consisting of the percent character "%" followed by the two hexadecimal digits representing that octet's numeric value.The unreserved characters that are not encoded are uppercase and lowercase letters, decimal digits, hyphen (or dash), period (or dot), underscore (or underline), exclamation, tilde, asterisk (star or multiplication), single quote, parenthesis, bracket, ampersand, equal, plus sign, dollar sign, comma, semicolon, question mark, slash, colon, at sign and back slash.
SIP OPERATION MANUALTerminal Adapter 50 Note: Enable Anonymous Caller ID or Anonymous Transit in W/O Caller ID, you may be unable to make a call since OR SERIES does not 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. CLIR At Transit in W/O Caller ID Disable it, if FXO detects caller ID from PSTN, OR SERIES will use the detected caller ID as caller identification; if FXO cannot detect caller ID from PSTN, OR SERIES will use “anonymous” as caller identification. When enabled, OR SERIES will always use “anonymous” as caller identification. 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. Enable Non-SIP Inbox Call Tick the check box to disable Non-SIP inbox call if all calls need to go through VSP.
SIP OPERATION MANUALTerminal Adapter 51Item 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 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 It decreases the size of SIP header. Tick the check box to enable this function. SIP CallerId Obtaining Defines from which part of the SIP packet will the gatewayobtain caller ID. There are several places where you can put your caller ID. Remote-Party-Id Display Name: It is locate at SIP →Remote-Party-ID → Before [<sip:] Remote-Party-Id User Name: It is locate at SIP → Remote-Party-ID → After [<sip:], Before [@] From-Header Display Name: The standard way is in SIP → Message Header → From → SIP Display info. Support URI Percent-Encoding (RFC 3986) It follows RFC 3986 to encode some letters as charactertriplet, consisting of the percent character "%" followed by the two hexadecimal digits representing that octet's numeric value.The unreserved characters that are not encoded are uppercase and lowercase letters, decimal digits, hyphen (or dash), period (or dot), underscore (or underline), exclamation, tilde, asterisk (star or multiplication), single quote, parenthesis, bracket, ampersand, equal, plus sign, dollar sign, comma, semicolon, question mark, slash, colon, at sign and back slash.Compare SIP ‘To’ Header for Transit Out When there is a call from WAN interface to FXO and the number of Request line and “To” is different, FXO will use the number of “To” to dial out. Please consult your Internet Telephony Service Provider about the format of invite packet from VSP.
SIP OPERATION MANUALTerminal Adapter 52 Virtual Server Enable users on Internet to access the WWW, FTP and other services from your NAT. It is also known as port forwarding. When remote users are accessing Web or FTP servers through WAN IP address, it will be routed to the server with LAN IP address. Item Description Enable Virtual Server Tick the check box to enable virtual server function. WAN Port Range Enter the port on WAN. TCP/UDP Select the communication protocols used by the server—TCP or UDP. LAN Host IP Address Enter IP address that the server provides various services. Server Port Range Enter the port used by the server on LAN. Remark The space reserved for notations. DMZ Demilitarized Zone lets the server on the LAN to be directly exposed to the Internet for accessing data. Either this function or the virtual server can be selected for use. Item Description Enable DMZ Tick the check box to enable this function. DMZ Host IP Address Enter the LAN host IP address.
SIP OPERATION MANUALTerminal Adapter 53Port Filtering Port filtering enables you to control all data that can be transmitted in routers. Note: When the port used at the source end is within the limited scope, it will be filtered without transmission. Item Description Enable Port Filtering Tick the check box to make this effective Port Range Set the range of port to be filtered. If set 80 and protocol is Both or TCP, all computers will be unable to use the services of http (port 80) — will be unable to browse normal WebPages. TCP/UDP Select the communication protocols used by the server—TCP or UDP. Remark The space reserved for notations.
SIP OPERATION MANUALTerminal Adapter 54 IP Filtering IP Filtering is to limit intranet users from accessing the Internet. Item Description Enable IP Filtering Tick the check box to make this effective IP Enter the IP address that you want to filter; the limited IP address will be unable to transmit the data to the Internet TCP/UDP Select the communication protocols used by the server—TCP or UDP. Remark The space reserved for notations.
SIP OPERATION MANUALTerminal Adapter 55Advanced Voice Services OR SERIES provides function for advanced voice settings, such as FAX, Codec, Speaking and Listening volume, etc. Line Settings You can adjust listening volume, speaking volume and tone volume here. Item Description Listening Volume It is to adjust the hearing volume. Speaking Volume It is to adjust the speaking volume. Tone Volume It is to adjust the tone volume. It will be applied to all tones volume generated by OR SERIES including Dial Tone, Busy Tone, and so on. Min. FXS Hook Flash Time It is to set the minimum flash time for FXS detecting. Flash Time FXS: Enter the maximum detecting period of flash signal from the phone set connected to the FXS port. For example, if pressing the HOLD key will disconnect a call, increase the “Flash Time” should fix this issue. PSTN: It is the time of PSTN port going on-hook. If on-hook time of PSTN is longer than the flash time of PSTN. Enable Polarity Reversal As the remote party answer this call, the polarity will be reversed.
SIP OPERATION MANUALTerminal Adapter 56 Item Description PSTN Ring OFF Length It is used to detect if the PSTN remoter party is on-hook through the ring length from PSTN by PSTN port. If the ring length form PSTN is larger than this setting, it is going on-hook by PSTN port, and it makes FXS stop ringing. CO Line Type Choose PSTN if the PSTN port is connected to PSTN line. Choose PABX if the PSTN port is connected to PABX line. FXS Chip Option 1 It is to avoid mis-detecting the loop state of a subscriber line or PBX user loop by FXS interface. In some places, the voltage of off-hook makes it mis-detect the idle state and the active state by FXS interface. Untick this variable if it mis-detects the state by FXS interface in your place. Item Description Ring (Early Media) Time Limit Specify the interval of ring time to cancel a call when no one answers a call. Enable End of Digit Tone OR SERIES will play a “Beep-Beep” tone to notify the call is in progress. It will play when invite packet is sent. Force Calling Thru PSTN Code Set the preferred code you set to force calling through PSTN. For example: If the code is set to *33 and you would like to dial “23456789” through PSTN, just dial “*33 23456789”. Early Media Treatment It refers to media that is delivered before call answer to inform the remote user about the session establishment. If you fail to make a call, please disable it.
SIP OPERATION MANUALTerminal Adapter 57Item Description Loop Current Drop Trigger Time It is to set the trigger time for dropping loop current by FXS port. A setting of zero is to disable this function. It is used to avoid the line engaged if FXS port is connected to PBX. Loop Current Drop Duration It is to set the drop duration. Enable ROH OR SERIES will play Receiver Off-Hook tone to notify user of hanging up the phone set. Item Description Listening Volume It is to adjust the hearing volume. Speaking Volume It is to adjust the speaking volume. Tone Volume It is to adjust the tone volume. It will be applied to all tones volume generated by OR SERIES including Dial Tone, Busy Tone, and so on. Min. FXS Hook Flash Time It is to set the minimum flash time for FXS detecting. Flash Time It is to adjust the maximum detecting period of flash signal from the phone set connected to the FXS port. For example, if pressing the HOLD key will disconnect a call, increase the “Flash Detect Time” should fix this issue. Enable Polarity Reversal As the remote party answer this call, the polarity will be reversed.
SIP OPERATION MANUALTerminal Adapter 58 Item Description FXS Chip Option 1 It is to avoid mis-detecting the loop state of a subscriber line or PBX user loop by FXS interface. In some places, the voltage of off-hook makes it mis-detect the idle state and the active state by FXS interface. Untick this variable if it mis-detects the state by FXS interface in your place. Item Description Ring (Early Media) Time Limit Specify the interval of ring time to cancel a call when no one answers a call. Enable End of Digit Tone OR SERIES will play a “Beep-Beep” tone to notify the call is in progress. It will play when invite packet is sent. Early Media Treatment It refers to media that is delivered before call answer to inform the remote user about the session establishment. If you fail to make a call, please disable it. Loop Current Drop Trigger Time It is to set the trigger time for dropping loop current by FXS port. A setting of zero is to disable this function. It is used to avoid the line engaged if FXS port is connected to PBX. Loop Current Drop Duration It is to set the drop duration. Enable ROH OR SERIES will play Receiver Off-Hook tone to notify user of hanging up the phone set.
SIP OPERATION MANUALTerminal Adapter 59 Item Description Listening Volume It is to adjust the hearing volume. Speaking Volume It is to adjust the speaking volume. Tone Volume It is to adjust the tone volume. It will be applied to all tones volume generated by OR SERIES including Dial Tone, Busy Tone, and so on. Flash Time It is to adjust the maximum detecting period of flash signal from the phone set connected to the FXS port. For example, if pressing the HOLD key will disconnect a call, increase the “Flash Detect Time” should fix this issue. Enable Polarity Reversal As the remote party answer this call or FXS picks up, the polarity will be reversed. PSTN Answer Detection This is used for VSP only. When there is call from WAN interface to FXO port, it could identify if the called party of PSTN answers this call. After it dials to PSTN, it will send “183” to the calling party. After the called party of PSTN answers this call, it will send “200 ok” to another the calling party and the VSP starts to charge. PSTN Ring OFF Length It is used to detect if the PSTN remoter party is on-hook through the ring length from PSTN by PSTN port. If the ring length form PSTN is larger than this setting, it is going on-hook by PSTN port, and it makes FXS stop ringing. FXS Chip Option 1 It is to avoid mis-detecting the loop state of a subscriber line or PBX user loop by FXS interface. In some places, the voltage of off-hook makes it mis-detect the idle state and the active state by FXS interface. Untick this variable if it mis-detects the state by FXS interface in your place.
SIP OPERATION MANUALTerminal Adapter 60 Item Description Ring (Early Media) Time Limit Specify the interval of ring time to cancel a call when no one answers a call. Enable End of Digit Tone OR SERIES will play a “Beep-Beep” tone to notify the call is in progress. Force Calling Thru PSTN Code Set the preferred code you set to force calling through PSTN. For example: If the code is set to *33 and you would like to dial “23456789” through PSTN, just dial “*33 23456789”. Trunk Early Media Options Early Media refers to media that is generated prior to connection or answer of a call is established by the called party. It may be unidirectional or bidirectional, and can be generated by the caller, the callee, or both. The gateway supports three early media mechanisms. These mechanisms occur from the moment “200 OK” being sent in response to an “INVITE” message. Both Way Voice: Use bidirectional early media to obtain information between caller and callee prior to the connection of a call. One Way Voice: Only the caller can hear early media from the callee prior to the connection of a call. Ring Back: Playing ring back tone for the caller, indicating that the callee is being alerted prior to the connection of a call. Early Media Treatment It refers to media that is delivered before the call is answered to inform the called party about the session establishment. If you fail to make a call, please disable it. Loop Current Drop Trigger Time It is to set the trigger time for dropping loop current by FXS port. A setting of zero is to disable this function. It is used to avoid the line engaged if FXS port is connected to PBX.
SIP OPERATION MANUALTerminal Adapter 61Item Description Loop Current Drop Duration It is to set the drop duration. Enable ROH OR SERIES will play Receiver Off-Hook tone to notify user of hanging up the phone set. Termination Impedance Choose correct impedance in your country/area. The wrong impedance will cause voice failure. Drop Inactive Call This is used as a standard for FXS and FXO interface to determine whether or not to back to the idle state. OR SERIES will back to the idle state automatically to avoid keeping the line engaged while the time expires and the detected volume is lower than Silence Detection Threshold. Item Description Silence Detection Threshold Set the ceiling threshold of voice energy to be identified as silence. Drop Silent Call Timeout Set the silence period to wait for before dropping a call. Voice Menu Options This is used to enable or disable IVR function or Call Feature Code. When disabled, call pickup/repeat, dialing/unattend transfer will be disabled.
SIP OPERATION MANUALTerminal Adapter 62 Codec Settings You can set the preferred codec, Jitter Buffer, Silence Detection/Suppression and Echo Cancellation in this section. Item Description Preferred Codec Type Since different voice codec have different compression ratios, so the sound quality and occupied bandwidths are also different. It is recommended to use the default provided (G.723.1) because it occupies less bandwidth and will provide better sound quality. Jitter Buffer It is to adjust the jitter to receive a packet. If the jitter range is too large, it will delay voice transmission. Silence Detection/ Suppression If one side of a connection is not speaking, OR SERIES will stop sending voice data (package) to decrease bandwidth usage. Echo Cancellation It is to prevent poor telecommunication quality caused by echo interference. Packet Time Defines how long OR SERIES sends a RTP packet (voice packet) to the remote party. The smaller the value, the more bandwidth usage. The larger the value, the more voice delay.Approximate Bandwidth Require The bandwidth required varies with codec format and packet time.
SIP OPERATION MANUALTerminal Adapter 63FAX Settings The line will detect FAX automatically if you choose T.30 Fax, T.38 Fax, T.30/Modem or T.30 Only. Choose the type of FAX protocol and set the related settings. Item Description Disable The line do not detect FAX automatically. T.30 Fax OR SERIES uses T.30 as the protocol for fax transmission. The parameter settings are the same as for voice transmission. However, enabling the fax function will consumemore network resources and will affect transmission quality. T.38 Fax OR SERIES uses T.38 as the protocol for fax transmission. T.38 is used for better and faster facsimile transmission. It is recommended to enable T.38 to gain better fax quality without setting fax and voice parameter. T.30 Fax/Modem Choose T.30 Fax/Modem as the protocol for transmission if OR SERIES is connected to Modem. T.30 Only Choose T.30 as the protocol for transmission. OR SERIES only accept the fax protocol of T.30. Item Description Enable High Quality OR SERIES sends the same FAX frame twice to get a high quality of the FAX when the line is using T.38 Fax. It requires more bandwidth. FAX Codec OR SERIES provides G.711 and G.726 for T.30 fax transmission. It is recommended to use G.711 for T.30. FAX Jitter Buffer It is to adjusts the jitter to receive fax packets. If the jitter range is too large, it will delay fax transmission.
SIP OPERATION MANUALTerminal Adapter 64 Other Settings OR SERIES provides advanced settings to apply to various situations. Here are Digit Map, DTMF & Pulse, CPT/Cadence Settings and Provision Settings. Digit Map Digit Map now is combined the original feature of Digit Map and Speed Dial. You can use “?” or “%” in the column of Scan Code, VoIP Dial-out and PSTN Dial-out. “?” is a single digit, and “%” is wildcard. It provides a mapping between the number received from user and the replaced or modified number for real dial out. With this function, user can easily add certain leading digits to replace full number. There are 50 sets of leading digit entries to choose voice routing interface. Digit Map Table Item Description Alert if Auto fails Tick the check box to play a voice announcement before calling out. It reminds user that this call is through PSTN. Enable Pound Key ‘#’ Function It is to speed up the connection of a call by entering ' # ' after a complete phone number is dialed. Default Call Route Define the default call route of OR SERIES. If Default Call Route is Deny, all numbers that are not match the Digit Map Table will be denied. Enable Tick the check box to make this entry effective. Scan Code Define the leading digits for OR SERIES to scan while the user is dialing. VoIP Dial-out Define the dialed number rule for OR SERIES calling through Internet.
SIP OPERATION MANUALTerminal Adapter 65Item Description PSTN Dial-out Define the dialed number rule for the gateway to call through PSTN/FXO port. User Dial Length Define total number of digits that user dialed. A setting of zero tells the gateway scans digits only and disregards the total digit count. Route It is to determine the interface calls should go through if above conditions satisfied. Digit Map Table Item Description Enable Pound Key ‘#’ Function It is to speed up the connection of a call by entering ' # ' after a complete phone number is dialed. Default Call Route Define the default call route of OR SERIES. If Default Call Route is Deny, all numbers that are not match the Digit Map Table will be denied. Enable Tick the check box to make this entry effective. Scan Code Define the leading digits for OR SERIES to scan while the user is dialing. VoIP Dial-out Define the dialed number rule for OR SERIES calling through Internet. User Dial Length Define total number of digits that user dialed. A setting of zero tells the gateway scans digits only and disregards the total digit count.
SIP OPERATION MANUALTerminal Adapter 66 Item Description Route Determine the interface calls should go through if above conditions satisfied. Digit Map Testing Item Description Test Dial No. You have to set some rules in Digit Map Setting first and enter the number for test. Result OR SERIES will show the number for VoIP Dial-out and PSTN Dial-out according to the Digit Map Table. Methods of Digit Map: Method 1- Single mapping: Fill a short code into the Scan Code column, and enter the desired phone number into the VoIP Dial-out or PSTN Dial-out column. Example - Single mapping, Scan Code: 55 VoIP Dial-out: 07021234567 User Dial Length: 2 Route: VoIP
SIP OPERATION MANUALTerminal Adapter 67Pick up the handset and dial 55 and OR SERIES will dial 07021234567. You also can use Digit Map Testing to know that OR SERIES will dial 07021234567 and go through Internet. Method 2- Multi mapping; Fill the prefix code into the Scan Code column and the format to transfer into the VoIP Dial-out or PSTN Dial-out column. Example 1 - Multi mapping, Scan Code: 2??? PSTN Dial-out: 351006??? User Dial Length: 4 Route: PSTN Pick up the handset and dial 2301. OR SERIES will dial 351006301 and go through PSTN/FXO. You also can use Digit Map Testing to know that OR SERIES will dial 07021234567 and go through PSTN/FXO.
SIP OPERATION MANUALTerminal Adapter 68 Example 2 - Multi mapping, Scan Code: 0% VoIP Dial-out: 0% PSTN Dial-out: 1805% User Dial Length: 0 Route: Auto Pick up the handset and dial 0423456789. OR SERIES will dial 0423456789 and go through Internet first. If the call is fail to Internet, OR SERIES will dial 1805423456789 and go through PSTN/FXO. You also can use Digit Map Testing to know that OR SERIES will dial 0423456789 to Internet and 1805423456789 to PSTN/FXO. Method 3- Substitution; It helps you dial to destination that you can not dial by phone. Destination like: test@1.1.1.1. Fill the number into the Scan Code column and enter the desired name into the VoIP Dial-out column. Example, Scan Code: 11 VoIP Dial-out: test User Dial Length: 2 Route: Auto
SIP OPERATION MANUALTerminal Adapter 69Pick up the handset and dial 11. OR SERIES will dial “test” and go through Internet. You also can use Digit Map Testing to know the dialing result. NOTE: In the example of Method 3, the result also shows that OR SERIES will dial 11 and go through PSTN. That means OR SERIES will dial 11 to PSTN if the call is fail to Internet. Please select the route is VoIP in this rule if the route is only able to Internet.
SIP OPERATION MANUALTerminal Adapter 70 DTMF & Pulse You can change these parameters if you have problems in dialing number. DTMF Settings Item Description Dial Wait Timeout It is to set the waiting time for the user’s first key pressing when dialing a number. The user will hear busy tone if the first key is not pressed within the set time frame. Inter Digits Time Out It is to set the waiting time between each key pressing. If the caller does not press the next number before the time expires, OR SERIES will play busy tone. Item Description Minimum DTMF ON Length Minimum DTMF OFF Length Define the length of diverting a call to another extension line. (Adjust length between Dail_on and Dail_off). DTMF Detection Sensitivity It is to adjust the sensitivity of detecting numbers for OR SERIES. FXO Dial Type Select dial type for FXO. There are DTMF and Pulse. Pulse Dial Mark/Space Ratio Duration and break of pulse dial ration.
SIP OPERATION MANUALTerminal Adapter 71Out-of-Band DTMF Item Description Enable Out-of-Band DTMF Tick the check box to send DTMF keys (0~9, *, #,) follow the RFC2833 rules or via SIP Info. Enable Hook Flash Event According to RFC2833 or SIP info, OR SERIES will deliver Hook Flash signal to the remote party. Volume Defines the DTMF volume of RFC 2833. Payload Type Payload type of RFC2833.
SIP OPERATION MANUALTerminal Adapter 72 CPT/Cadence Settings OR SERIES will generate the tones by the call process tone parameters table. CPT parameters Table The CPT has 2 sets of parameter tables. Please adjust the parameters based on local PSTN.
SIP OPERATION MANUALTerminal Adapter 73Busy Tone Cadence Measurement CPT/Cadence setting parameters serve as the basis of an FXO interface to determine whether or not a PSTN-call receiving party has hung up the phone. If the following parameters differ from the parameters of the actual assigned lines, it could cause the FXO to continue to engage a line. Item Description Auto Learning It is to learn the busy tone automatically by FXO port. BTC Detection Sensitivity The more sensitivity, the more quickly it will cut off the call by FXO port. If it often cut off an un-finished call by FXO port, select less sensitivity. BTC Volume Threshold The detection level for BTC.
SIP OPERATION MANUALTerminal Adapter 74 Provision Settings Provision Server is used to provision, configure, manage and maintain subscribers and network users. OR SERIES, acts as a part of subscribers, can be controlled by Provision Server. OR SERIES provides a simply way for users to connect and send request to Provision Server by enabling this setting. With this system, the Server can not only easily modify a configuration file to change gateway settings but to assign latest firmware for specific gateways to upgrade. Besides, Provision Server also reports the status of OR SERIES and all actions will be recorded in log file that offers users to trouble shouting effectively. Item Description Enable Auto Provisioning Tick the check box to start provisioning. Provision Server Address Enter the IP address/Domain of Provision Server required by your provider. Port The port of Provision Server. Packet Format Select the packet transmitting format required by provision server. Connect Provision Server During Start Up OR SERIES will connect to Provision Server when it power on or reboot. Connect Provision Server Periodically It is to adjust the parameters for OR SERIES to connect to provision server periodically. Suspend Service It is to adjust the parameters for OR SERIES to do auto provision task.
SIP OPERATION MANUALTerminal Adapter 75 Item Description Binding Server for Trigger Tick the check box to trigger of a connection between server and OR SERIES. Server will bind a port for the gateway to send provision request. Binding Port The binding port number of the server is used to tell OR SERIES the path of binding server. Binding Interval It to set the desired Interval at which OR SERIES will keep the binding.
SIP OPERATION MANUALTerminal Adapter 76 Transit Call Control This is to control outgoing call and incoming call through FXO. Transit Call Control is effective when it cooperates with Long-Distance Control Table. Long-Distance Exception Table is for an exception and it will not be restricted by Transit Call Control and Long-Distance Control Table. You have to enable both of Inbound/Outbound Call Control and PIN Code. Transit Call Control is active in one-stage dialing. Item Description Inbound Call Control Tick the inbound PIN code when users make phone calls from a PSTN to FXO and then using a VoIP － only effective for incoming calls calling from a PSTN trunk. Outbound Call Control Tick the outbound PIN code when users utilize FXO interface to divert to a PSTN －only effective for outgoing calls being diverted to a PSTN Trunk. PIN Code Enter the PIN code (4-6 digits or leave blank. A blank indicates no PIN code is required at this level. Generally, the PIN at level 5 can remain blank to simplify the phone number.) Enable Tick the check box to enable the PIN code at each level. Privileges The level is divided into 0~5 (The levels are in descending order; 0 stands for the highest authority and 5 stands for the lowest.) The dialing principle to PIN Code is below: * inbound call control PIN code* outbound call control PID code* phone number Using * to separate PIN code and the phone number is based on actual settings.
SIP OPERATION MANUALTerminal Adapter 77Long-Distance Control Table This table controls the level of authority of an outgoing (transit out) call that is dialed through FXO and diverted to PSTN, as below. This table is used to prohibit dialing any numbers started with specified prefixes. Digit strings in this table are prefixes that the gateway will check on dialed numbers in transit out calls. It is Downward Restriction — If the users at a higher level cannot dial a number with a certain prefix, then users at lowers level also cannot dial a number with the same prefix. For example, Level 1 is set to prohibit dialing any number with prefix 0, then any level below 1 (including Levels 2 to 5) is also prohibited. Since Level 0 is not restricted to any prefix, therefore at level 0 users can dial a number with the prefix 0. Long Distance Exception Table This table handles any exceptions to the long-distance call table. According to the Long Distance Control Table, users at Level 0 are prohibited from dialing a number with the prefix 0204. But, if the number 020488988 is set in the Exception Table as above, then users could then dial this number. It is Upward Opening —If the users at a lower level can dial a number with a certain prefix, then the users at higher levels can also dial a number with the same prefix.
SIP OPERATION MANUALTerminal Adapter 78 Status and Tools This section shows the status of OR SERIES. There are Current Status, RTP Packet Summary, System Information, Ping Test and STUN Inquiry. Current Status Port Status: It includes if each port registers to Proxy successfully, the lasted dialed number, how many calls each port had since OR SERIES is start, etc. Server Registration Status: It shows the registration status of DDNS, Phone Book Manager, STUN and UPnP. RTP Packet Summary Display the information of the final call. Press Refresh button to get the latest RTP Packet Summary.
SIP OPERATION MANUALTerminal Adapter 79System Information WAN Port Information: It shows IP address, subnet mask, default gateway and DNS server. If you use PPPoE to obtain IP, you can know if the IP is obtained through this. If IP address, subnet mask, default gateway is blank, it means that OR SERIES does not obtain IP. LAN Port Information: It shows LAN port IP, subnet mask, and the status of DHCP server. Hardware: It shows the hardware platform.
SIP OPERATION MANUALTerminal Adapter 80 Ping Test Use Ping to identify if the remote peer is reachable. Fill in remote IP address and click Test will start the test. STUN Inquiry It is to know what NAT type of the router when OR SERIES is behind NAT.
SIP OPERATION MANUALTerminal Adapter 81System Settings This section provides system settings such as NTP, Login Account, Backup/Restore, System Operation, Software Upgrade and Logout. NTP It is to set the Time Zone where OR SERIES resides. You can set the Time Server where OR SERIES should sync up during start up. Login Account There are two sections in this page: Login Settings and Accessing Services. Login Setting: There are two levels to enter Web. Administrator is able to change all settings. Web UI only changes some settings. Access Services: It is to allow users to access OR SERIES not only from Web but also from Telnet. Login Settings Note: Enter new Login ID and password for two levels.
SIP OPERATION MANUALTerminal Adapter 82 Access Services Note: When “Enable Web UI” is unticked, you cannot access from Web. Item Description Port of Web Access from WAN Http port for WAN. To make this setting, the LAN Port must be used. It cannot be made using the WAN Port. Always use port 80 when connecting to LAN port. A setting of zero is to disable http port for WAN. Web UI auto logout If OR SERIES is inactive for the period defined in this filed, Web UI will auto logout to keep OR SERIES secure. Enable Web UI Untick the check box to disable WEB access from WAN or LAN while necessary. Enable Telnet Service Untick the check box to disable Telnet access from WAN or LAN while necessary. Backup/Restore You can backup settings to a file and restore settings from that file. Backup Configurations Item Description Configuration File It is to backup the all settings. Configuration Template File It is to backup the settings as template file for editing.
SIP OPERATION MANUALTerminal Adapter 83Restore Configurations You can backup settings to a file and restore settings from that file. You also can restore all settings back to default by selecting Restore Default Configurations and click Restore. Note: You have to save settings and restart, and all settings will take effect. System Operations Some settings are effective by Restart. Remember to save all settings by Save Settings before to restart. Item Description Save Settings Save settings after completing configuration. Restart The new settings will take effect after OR SERIES is restarted. Please select it and click the Accept button.
SIP OPERATION MANUALTerminal Adapter 84 Software Upgrade OR SERIES provides software upgrade function for a remote end. Your provider gives all parameters. Item Description Upgrade Server Choose the server type of your provider. Software Upgrade Server IP Enter the software upgrade server IP address. Software Upgrade Server Port Enter the port that server uses. TFTP is 69, and FTP is 21. User Name/ Password The account/password is to login the upgrade server. Directory The location of Directory for Upgrade Server. Logout OR SERIES only allows one user to login at a time, so whenever a change is made, please save the settings, restart OR SERIES, or logout to avoid the situation where other users cannot login to change settings.
SIP OPERATION MANUALTerminal Adapter 855. TCP/IP Setting Follow the description if you have problems in how to assign a static IP Address in your PC. Using Windows XP for example Go to Start -> Click on Control Panel -> Double-click on Network and Dial-up Connection -> Click on Open Local Area Connection ->
SIP OPERATION MANUALTerminal Adapter 86 Click Properties. Highlight Internet Protocol (TCP/IP) and then click Properties.
SIP OPERATION MANUALTerminal Adapter 87Select 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.
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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