VoIP SIP IP PBX IP PBX with 4-FXO and SIP Proxy Server UTG7204-ON User Manual Copyright EUSSO Technologies, Inc. All rights reserved.
Table of Content 1. INTRODUCTION ........................................................................................................................................................... 4 1.1. OVERVIEW .................................................................................................................................................................. 4 2. FEATURES................................................................................................................................
8.4.3 Fallback Settings............................................................................................................................................... 40 8.5 SWITCHBOARD (AUTO ATTENDANT SETTINGS) ............................................................................................................ 42 8.5.1 Prompt Message............................................................................................................................................... 42 8.5.
1. Introduction 1.1. Overview The UTG7204-ON is an embedded Voice over IP (VoIP) PBX Server with Session Initiation Protocol (SIP) to provide IP extension phone connections for global virtual office of small-to-medium business (SMB) companys.
2. Features The UTG7204-ON IP PBX equipped with RJ45 & RJ11 connectors is featuring as the following: ¾ SIP Server supports 50 user registrations and 20 concurrent calls ¾ SIP v1 (RFC2543), v2 (RFC3261) with MD5 authentication (RFC2069 and RFC 2617) ¾ RJ45 x 2 for Ethernet WAN and LAN ports + RJ11 x 4 for FXO ports + Life Line FXS port ¾ Supports ITU-T G.711a, G.711u, GSM/MS-GSM, G.
4. Packing Content Inside the package you should find: (1) One UTG7204-ON IP PBX (2) One AC100~240V to 12VDC/1A Power Adaptor (3) One Cat 5 Ethernet Cable (4) One User Manual CD 5. LED Indicators & Interface Connectors LED Indicators On the front panel of UTG7204-ON, there are 12 LED indicators as the following table LED Status POWER ON Power is Normal.
Interface Connectors 1. DC Power 12Volt DC / 1A Power Adaptor with 100~240V AC power input 2. FXO ports 4 FXO ports are for connection to PSTN lines, and numbered 1, 2, 3 and 4 from left to right. 3. Life Line port Life Line FXS port connects to an analog telephone. When power is down, the Life Line will switch to FXO port 1 for PSTN line 1. 4. WAN port Connect to a broadband ADSL/Cable modem or a WAN router. 5.
6. Installations FXO WAN ADSL modem LAN Ethernet Switch Hub FXS PSTN IP Phones Internet Life Line POTS 7. Reset to Factory Default IP PBX UTG7204-ON can be reset back to factory default when IP address is not accessible for web configurations. The procedures are as follows: 1. Power off. 2. Press the RESET button and hold continuously, then power on. 3. Hold the RESET button until all the LED indicators start flashing for three times.
8. IP PBX Configurations by Web Browser You may enter the IP address from PC Web browser to configure UTG7204-ON. For example, enter http://192.168.1.1 from IE web browser to display login page as follows. supports auto-MDIX function for LAN port. Note that UTG7204-ON A beginning user of UTG7204-ON may click Setup Wizard to configure the IP PBX step by step for web configurations. 1). Please enter the default IP address http://192.168.1.1 from PC Web browser.
4). UTG7204-ON provides 50 SIP user ID number accounts which can be configured by Web browser. The preset user ID numbers are from 2001~2010 with same password 123456. The SIP service port is default at 5060.
Setup Wizard Setup Wizard provides two fast setup modes, Router & Non-Router Modes, for sequential configurations; (1) IP-PBX used as a Router (Router Mode) ¾ LAN Setting ¾ WAN Interface Setting ¾ Dynamic DNS Setting ¾ Incoming Call from VoIP ¾ Add TrustHost ¾ Outgoing Call via VoIP ¾ Call from FXO ¾ Outgoing Call via FXO ¾ Finish and reboot (2) IP-PBX used as a Non-Router (Non-Router Mode) ¾ WAN Interface Setting ¾ DMZ Mode ¾ Incoming Call from VoIP ¾ Add TrustHost ¾ Outgoing Ca
8.1 Network UTG7204-ON IP-PBX provides two RJ45 connectors for LAN and WAN ports at 10/100M Ethernet interfaces. The Network will display the current status for LAN, WAN, DHCP, DDNS, and VPN settings. 8.1.1 Network Status Network Status shows all the IP addresses for LAN, WAN, VPN server and VPN clients.
8.1.2 LAN Settings LAN Port can be used for IP-PBX to connect to a Notebook PC for configurations. The embedded DHCP Server will automatically assign IP address through the LAN port. MAC Address should be unique in the same network, and IP Address must be in the format of xxx.xxx.xxx.xxx and xxx is from 0 to 255, e.g. 192.168.1.1. segmentation. Subnet Mask is used for network Please make sure the mask is correct and all the VoIP devices are within the same network as UTG7204-ON. 8.1.
Static IP mode Dynamic IP Mode PPPoE Mode When UTG7204-ON IP-PBX connects to ADSL Modem with PPPOE link, you need to enter the account name and password for PPPOE.
IP address for the First and Second DNS servers. 8.1.4 DHCP Server The embedded DHCP server in NAT will automatically assign IP address to the network devices. DHCP Server Status: To show the current DHCP server status DHCP Server Start/Stop: To enable/disable DHCP Server Start/End IP Address: DHCP Server will assign an IP within the start/end IP address range, e.g. 192.168.1.100－192.168.1.200. Note that the start IP and end IP must be in the same 192.168.1.xxx network. Subnet Mask: Usually 255.255.
Service Provider: Choose DDNS server that updates your IP address. Host Name: Set your host name for DDNS update. User Name: The user name registered in DDNS server. This name is given by DDNS server. Password: The password registered in DDNS server. This password is given by DDNS server. DDNS Status: Current Status of Disable/Enable. DDNS Service: To Disable/Enabled DDNS service. Example for www.3322.
Example for www.dyndns.com DDNS Settings It shows the current registration of www.3322.org or www.dyndns.org for dynamic DNS service. 8.1.
VPN Server Configuration for PPTP (Point-to-Point Tunnel Protocol) VPN Server: You may enable or disable VPN Server Service. UserName: For remote user when connected to VPN server. Password: For remote user when connected to VPN server. Press Update to save the configurations, and press Reboot to activate the new configurations. VPN Client Configuration for OPENVPN OpenVPN is used when the IP-PBX register as a VPN client to a remote VPN server.
8.2 System The UTG7204-ON IP PBX System configurations can be set in this section. The settings include Server Ports Setting, Rate Setting, DMZ, TrustHost, Music on Hold, Hotlines, Admin Account, USB_Disk Setting, Voicemail Setting, and Time Zone. 8.2.1 Server Ports Setting The SIP port number is default at 5060 for VoIP Applications. Note that the SIP port number must be the same for all the IP-PBX and VoIP phones and TAs. The encrypted port can be used for VoIP encryption.
8.2.2 Rate Setting Rate setting is used to calculate the charges for each call. The IP PBX will generate a call record for the charges. All the charges are based on this rate table. The rate is based on the prefix to calculate the charges: Example: Prefix: Rate: 0 15 (cent) Time Unit： 60 (second) This means when calling a number with 0 prefix (e.g., 01010086), the rate is 15 cents for every 60 seconds, and the duration less than 60 seconds will be charged the same as 60 seconds.
Please follows the steps to configure the DMZ: (1) Click on DMZ (2) Set the DMZ IP or domain name URL (3) Enter the Network IP and Subnet Mask Example: 192.168.1.0/255.255.255.0 The step (3) is optional. 8.2.4 Trust Host The calls from Trusted Host will be accepted without authentication by IP PBX. This function can be useful for calls between two or more IP PBXs. If the port is set at 0, all the ports from this trusted address will not require authentications.
(1) Need to work with FXO ports, (2) Need to connect with UTG7204-ON, (3) Need to connect with another SIP Server. In summary, the TrustHost address can be set whenever the authentication is not needed. 8.2.5 Music ON Hold The IP PBX will play music when a call is on hold due to the following situations; (1) When the call is transferred to attendant and waiting for answer. (2) When the call is hold and waiting for answer. (3) When the call is transferred and waiting for answer.
8.2.6 Hot Lines Hot line function numbers can be entered into the list. Make sure each of the Hot line function number is unique, and not the same as any extension number or PSTN numbers. The Hotline list displays the function number of the PBX, and it will activate the function service when pressing the corresponding function numbers.
Detailed Descriptions for Hotline Function Numbers Hotline Numbers Functions Descriptions *8 Call pickup Press *8# to pickup calls within the same group 9 Operator Press 9# to call Operator. 112 Auto Attendant (AA) 700 Call Parking 701 Call Parking Press 112# to enter Auto Attendant IVR followed with the voice prompts. Press Hold + Transfer + 700# to park the incoming call. You may hang up the phone afterward. Pick up the phone and press 701# to take the waiting call.
Forward , Find me , Binding, and Password can be set up for each extension number. 1602 Extension functions Example: To set extension 202 busy forward to 203 Procedures: From 202 IP Phone, Dial 1602 -> 1 (set forward) -> 2 (busy forward)-> 203# -> 1 (submit) -> hang up. To play & to delete the voicemails for the current IP phone extension number.
Hotline Applications Example: A customer calls into Operator and wants to talk to the IP extensions 203, but the 203 user are not near the phone.
Submit to add the number 208 for Queue1. The number will be displayed in the Current Queue Agent List for Queue 1. Submit to add the number 203 for Queue2. The number will show in the Current Queue Agent List for Queue2 . 3) Add Voice prompts to Switchboard. Submit.
It will display voice prompts in the List. Press the ICON to Activate the Voice. The No. 1 of Status will show “1” to indicate activate status. 8.2.7 Admin Account 8.2.
When you need to store the voice mails, please do the steps as follows; (1) Insert USB disk (2) Select “Insert USB Disk” in the current status (3) Select “Enable” for USB voice mail recording. (4) Click Submit button. (5) Move to Upgrade&Reboot then press Reboot to restart . When the voice mails is not needed, please do the steps as follows; (1) Select “Remove USB Disk” in the current status (2) Select “Disable” for USB voice mail recording. (3) Click Submit button. (4) Pull out the USB disk. 8.2.
General settings : MailBox: The e-mail address of mailbox. User: The user name used by the e-mail address. Password: The password used by the e-mail address. SMTP Server: This is the e-mail address used by SMTP for e-mail server domain name. Use SSL Mode: Check to use SSL mode. Do not check if General settings is used. SSL for Gmail : The UTG7204-ON supports voicemail via Gmail server by using SSL mode. MailBox: The sending account in Gmail. This must be registered in advance in Gmail.
8.2.10 Time-Zone Setting System date and time can be set in this section. The time will be recorded for use of CDR, or the CDR may record wrong timing if the system date and time are not set. The Network Time Service can be enabled to synchronize with NTP server. You still need to set the TimeZone, Date, and Time. It will display the synchronized NTP time when you re-enter this setting. Press Submit to save the configurations, and press Reboot to activate the new configurations. 8.
8.3.2 Calls from VoIP The IP PBX can function as a terminal CPE to register into a SIP server or another IP PBX. For any external VoIP calls, for example, 1234 into this IP PBX, you may redirect to either Auto Attendant, Operator, Conference rooms, etc.
z Registry Address : The SIP Registration Server IP/URL:Port number provided from ITSP, other IP PBX, or SIP Server. z User Name : The user name provided from ITSP, other IP PBX, or SIP Server. z Password : The password provided from ITSP, other IP PBX, or SIP Server. z Redirect To: AA, Operators, Conference rooms, etc.
The list will display registered users with status after submit. You may also press the icon to modify the setting.
You may press the icon to delete this setting. 8.3.3 Access Number Access Number is used to supplement the function of incoming calls from VoIP SIP Server in the previous section. For some ITSP SIP Servers, one way voice connection may be missing due to protocol incompatibility. In this case, the Access Number can be set up to register into ITSP SIP Server and to redirect VoIP calls from ITSP into the IP PBX.
Access Number : The registered number to ITSP or SIP server. Redirect into : One of the hotline functional calls. Memo : For memo descriptions. The Access Number will be displayed after pressing Submit button.
8.4 Outgoing Call The IP PBX provides two kinds of outgoing calls; one is via FXO port to local PSTN lines, and the other is via VoIP calls. The settings are as follows: 8.4.1 Outgoing Calls via FXO ports The IP PBX is equipped with 4 FXO ports for connection to local PSTN lines. number can then make outgoing call to local PSTN numbers. All the extension You need to assign the group number to each of the FXO port to activate the FXO outgoing PSTN line.
Example: When one user press 056789 to apply the outgoing rule, the IP PBX will use one of the FXO port in the group 0, delete 0 (one digit), then add the prefix blank. PSTN line.
8.4.2 Outgoing Calls via VoIP The UTG7204-ON IP PBX can make a VoIP call to the user of another remote UTG7204-ON. The user of the remote IP PBX can be either extension number or its PSTN numbers. There are two ways to connect to the remote IP PBX: (1) The remote IP PBX provides an account name and password. The local IP PBX will make outgoing call authentication. (2) The remote IP PBX configures local IP PBX as TrustHost address.
8.4.3 Fallback Settings The fallback can be used to call out via different routing when the previous call fails. This reduces the calling failure rate and simplifies the dialing. z No: Fallback Sequence Number. If the old number matches more than one rule, the rule of fallback generated number with smaller sequence number will be dial first. If failed again, the rule of next sequence number will be dialed, and so on until call connects.
Example: You may also use the dial plan for fallback new number. If you want to dial 9123456 with fallback new number 0123456, then you may enter 0${EXTEN:1}. Given dialing number 9123456, ${EXTEN} represents 9123456, ${EXTEN:1} represents 123456, and 0${EXTEN:1} represents 0123456. z Match: Yes/No Yes indicates that old number is a prefix and the rule of new number will be applied to generate new number for any prefix-matched number.
8.5 SwitchBoard (Auto Attendant Settings) The IP PBX SwitchBoard provides an Auto Attendant to service user for calls. When receiving an incoming call, the auto attendant will play a welcome prompt message and transfer to another extension number. The settings include welcome prompt message, operator, and auto attend. 8.5.1 Prompt Message You may choose the prompt message for the auto attendant to play while receiving incoming calls.
8.5.3 Auto Attendant Auto Attendant will handle all the incoming calls when no one can answer in the company. There are two ways for attendant. The first one is Prompt with playing answering prompt messages, and the second one is Phone with direct transfer to the phone number. z Prompt : An answering prompt message will be played to answer the incoming call. Make sure the answering message is uploaded and chosen. The IP PBX provides many options for time durations.
z Number : If you select the Number way . The extension number will answer the call. z Time/Week/Date/Month: The time of these time field set (Time & Week & Date & Month), the extension user will answer the call. 8.6 Users Management This section describes the account opening, closing, and management. The extension number (also known as user name) must not exceed 32 digits. For easy management, it is recommended to assign different prefix number for different departments.
8.6.1 Add Single User – Single User Account Opening You may add single user account in this section by entering user name, password, groups, and call authority. Remember the length should not exceed 32. Note: when the number of registered users reach the limit, the function of add user will not work. 8.6.2 Add Group Users – Group Users Account Opening You may create group users accounts and assign group numbers with priority to all the accounts.
8.6.3 Bindings Extension binding is used to bind many extension numbers together as a group. of the extension, the other binding extension will ring as well. When calling to one This is also referred to as Group Ringing. (1) User-ID: To enter extension number. (2) Bind-ID: To enter the binding ID. The extension number with same binding ID will belong to the same binding group. (3) The binding list will display the extension numbers. Note that one extension number can belong to multi-binding groups.
8.6.4 Delete User Account There are two ways for user account deletion; one is for single account deletion, and the other is for group account deletions. For single account deletion, you need only to enter the extension number. For group account deletions, you need to enter a range of extension numbers. The IP PBX will delete all the user information for the extension within the range. Note that the information will not be recovered once deleted. 8.6.
z Inquiry : Choose one User-ID or User Name for search . z Condition (Fuzzy Query) : Enter the number or user name for search. This page will show the Modify User settings. 8.6.
z Inquiry : Choose one User-ID or User Name for search . z Condition(Fuzzy Query) : Enter the number or user name for search. This page shows the Modify User's Function setting. (1) Call Forward ¾ Unconditional Forward: Any incoming call to this extension will be forwarded directly to the set extension number. ¾ No-Response Forward: When no answer in one minute, the call will be transferred to the set extension number.
answer for the incoming call. If no answer again, the call will be transferred to the next extension number until the call is answered. The maximum is 6 extensions numbers. After that, the call The telephone binding is to bind the extension number with a PSTN number. For incoming call will be disconnected. (3) Telephone Binding to the extension number, the PSTN phone number will ring simultaneously. (4) Voice Mail You may enable the voice mail for extension numbers.
The list will show the current setting status for each user as follows; (1) Normal (2) Unconditional Transfer (3) Call Transfer; including busy, off-line, and no answer transfer. (4) Find Me (5) Telephone Binding You may modify and update all the user settings directly from the entry of the list. 8.6.8 On-Line User List The on-line users are for the current registered users. 8.7 Advanced Settings The advanced Settings cover some call waiting, voice conference rooms, and IVR upload process. 8.7.
Firstly, you need to set up auto configuration for the IP Phone. The following is a reference example for IP Phone setting. z Auto Configuration: Select the HTTP protocol. z HTTP Server: IP address or URL of the IP-PBX. z HTTP File Path: Configuration file path, it must be “/conf/”. Secondly, you need to configure for the IP PBX. z User-ID : The extension number of the telephone. z Server Address : The IP address or URL & port of the IP PBX.
z Phone's Mac Address : The Mac address of the IP Phone. 8.7.2 Queue Setting Call waiting is to queue all the incoming calls and to assign based on rules to the desired extension numbers. If all the available extensions are busy, the system will play the waiting music for the queuing incoming calls. Once available, the system will connect to the desired extensions.
The IP PBX provides 4 call waiting queues. The waiting queues can be set to a simple calling center, and the user may define individual waiting queues for its own purpose.
8.7.3 Conference Rooms The IP PBX provides two standard voice conference rooms. The details are as follow; (1) Conference Room Password When a user calls to the conference room, he will be required to enter the password, or the call will be denied. It is recommended to set the password for the conference room. Please refer to the conference room setting for password. (2) Maximum attendants of conference room If set at 0, it has no limitation. Please refer to the conference room settings.
(2) The extension is in monitoring status. (3) Kick out the extension number. The extension will be forced out in 5 minutes.) (4) IVR Control of Conference Room The user after entering the conference room may press * key for an IVR playback. The user may interact with the IVR messages. General User Extension: 1) Make oneself mute. The user voice will not be heard in the conference room. To let one’s voice heard in the conference, just repeat the same procedure as mute.
The conference room password is used to enter the conference room. The maximum capacity is for the maximum number of users in the room. 8.7.4 Network Parameters The network parameters allow to enable/disable IP network functions. Support Voice Priority Tag (TOS/DSCP) TOS (Type of Service) and DSCP (Differentiated Services Code Point) are used to set different priority to data packet and data flow, thus enabling QoS in IP communication.
DSCP field is a superset of TOS, its definition is backward-compatible with TOS, and its value can be from 0 to 63, with 0 for minimum priority, 63 for max priority. Support Tag-based VLAN 802.1P/802.1Q. VLAN 802.1P/802.1Q are for local area network standard recommended by IEEE, which can partition network users from different physical locations into several logical subnets. The UTG7204-ON IP-PBX supports this standard, and can add special VLAN Tag to data frames passing through. 8.7.
FXO Volume: Voice transmit gain and receive gain of the FXO port. Silence Threshold: Signal under this threshold will be regarded as silence. Tolerances Percentage: Number of percentage for the silence threshold. Busy Min: The minimum number for busy detection at unit of ms. Busy Max: The maximum number for busy detection at unit of ms. 8.7.7 Advance Settings ( Caller ID Settings) Reinvite is a unique function, and you may enable the function if the SIP server of your ISP supports such a function.
8.8 Call Detail Records Query The IP PBX keeps call records for 2 months. You may have two ways for record queries. query for the whole IP PBX, and the other is for the specific user. If you want to keep the call records, you may copy to USB disk before erased. 8.8.1 System Call Records You may query all the calls during the specified time frame and export to local storages.
By Time 8.8.2 User Call Records You may query all the calls for certain user during the specified time frame and export to local storages.
8.9 Upgrade & Reboot 8.9.1 Export & Import Before upgrading the UTG7204-ON IP-PBX or reset to factory defaults, you may export all the user information to local PC storage and import back after the upgrade or default settings are done. Export: Click on “DOWNLOAD” button and select “Save as New File”. Import: Select the desired file, and upload.
8.9.2 System Upgrade The UTG7204-ON supports two different ways for system upgrade; Web GUI Upgrade and SIP Server Upgrade. Before upgrading, please follows the steps: (1) Get the desired upgrade version. (2) Backup a copy of user information by import/export data files. (3) The power must be on while upgrading. Then you may upgrade per the web instruction procedures. 8.9.3 Reboot Reboot is used to enable the changes. Note that for Factory Default Setting, please refer to the chapter.
9. Applications Applications of IP PBX under Firewall with DMZ This will protect corporate network security while allowing UTG7204-ON to work as IP-PBX for VoIP applications. When UTG7204-ON IP-PBX is operating under the corporate firewall, remember to enable and open the following service port numbers for VoIP applications. ¾ TCP Port：80, 1723 for Http, and PPTP. ¾ UDP Port：5060, 1194, 10000-20000 for SIP, OPENVPN, and Voice RTP Range.
Applications of IP PBX with ADSL This UTG7204-ON supports PPPOE to work with ADSL and to integrate IP-PBX into the corporate network for VoIP applications.
