Deltapath. frsip Unified Communications Core Configuration Guide version 1.0. For Use with XO Communications SIP Trunk services

Transcription

1 Deltapath frsip Unified Communications Core Configuration Guide version 1.0 For Use with XO Communications SIP Trunk services

2 Table of Contents Introduction 1 Executive Summary... 2 Software and Hardware Equipment Requirements for Testing Package Contents Types of PBX Primary and Backup synchronization Connecting frsip Appliance on Same Physical Location.. 5 Connecting frsip Appliance on a DR (Disaster Recovery site) or Geographical Cluster... 6 frsip APPLIANCE FRONT PANEL LED SIGNALING. 8 frsip APPLIANCE and Gateway CONSOLE ACCESS MANAGEMENT Setup Console Port Connection Parameter. 9 Console and Web Default Login And Password Change. 9 Help Menu.. 9 Change Default Account Login Password. 9 Setup Network Via Console Access Test Bed Configuration Files Lab Configuration (Diagram). 11 Network Configuration: Set WAN (IAD) & LAN IP. 12 Recommended settings of IP address.. 13 Phone System Dial Plan User configuration User Profile Add New User Extension.. 15 Extension Configuration Assigning DID Range to Extension Routing Handset configuration Trunk Group SIP Peer: Set NBS SIP Port.. 21 Music On Hold Analog Gateway System Extension. 23 Auto Attendant. 24 Creating IVR Menus.. 25 Call Center Call Queue Agent and Agent Group Creating the queue Creating Agent Group Extension Status Software Version Page 1 of 36

3 Executive Summary This report provides the test results found to date for the Deltapath Unified Communications SIP Trunking evaluation. The following is a summary that addressed the issues and limitations found while performing the test. Diversion header is not supported so that in the Call Forward which is originated from PSTN and forwarded back to another PSTN number, the 3 rd party can only see the 2 nd party calling ID instead of the 1 st party. Fax is only supported with T.38. Fax with G.711 pass-through is not supported. Call Transfer with REFER is not supported. Registration Method Static registration is utilized between the Deltapath IP phones and the XO call agent. XO SIP Service Packages Supported Pkg Codec DTMF Fax 1 G.711 RFC 2833 / in-band DTMF T.38 (G.711 pass-through NOT supported) 2 G.729a/G.711 RFC 2833 / in-band DTMF T.38 (G.711 pass-through NOT supported) Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 2

4 Software and Hardware Equipment Requirements for Testing Deltapath Software version : Polycom IP Phones Model no.: 430 Page 3 of 36

5 Package Contents Open the package and check the items included. The following items should be included: 1 x frsip Appliance Main Unit 2 x L-shape rack mount metals 16 x screws 2 x power cords 1 x console cable To install system hardware Disclaimer: The frsip main unit is around 15Kg. Do not attempt to rack mount the unit yourself. You must seek additional assistance to mount it on to your server rack or you may expose yourself to potential bodily injuries. Rack mount the frsip Unified Communication Platform, attach the 2 L-shape rack mount metals to the side of the front end of the appliance with the screws provided. Insert each power cord connector into the rear of the unit and connect each to an appropriately rated socket outlet. The frsip Unified Communication Platform unit is supplied with two power cords, BOTH power cords should be connected to the power supply outlet during normal operation. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 4

6 2 Types of PBX Primary and Backup synchronization Connecting frsip Appliance on Same Physical Location: If you only have one single frsip Unified Communication Platform, simply connect the RJ45 cable from the port you have configured the IP address to your network. When you are setting up cluster with two frsip appliances, refer to the following graph for how to connect your main and hot standby frsip appliance. Power on both appliances and connect your PC to the console port of the primary appliance and assign an IP address on Port 1(eth0). After assigning an IP to the primary box, connect the console to the backup server and assign an IP on Port 6(eth5). This IP address will be the main access of your whole frsip UC Platform system. Please refer to later chapter frsip Appliance Console Access Management for how to setup IP address via console access with your PC. Line connection description: Primary appliance port 1 to network: Main connection of the whole frsip UC Platform system. Hot standby appliance port 6 to network: Temporary network connection for hot standby unit which will be removed after forming cluster. Primary appliance port 2 or port 4 respectively to hot standby appliance Port 2 or port 4: The connection is for IP address failover use when primary server not in service. Primary appliance port 5 to hot standby appliance port 5: For data synchronization between primary and hot standby unit. Page 5 of 36

7 Connecting frsip Appliance on a DR (Disaster Recovery site) or Geographical Cluster: Setup different IP addresses on both your primary and backup server with your assigned IP s. This IP addresses will be the main access of your whole frsip UC Platform system. Please refer to later chapter frsip Appliance Console Access Management for how to setup IP address via console access with your PC. Connect port 1 of the appliance to their respective networks to be able to access each server with different IP s. Server Synchronization - Forming Cluster With Two frsip Appliances Access the frsip web interface to setup cluster once the network is ready. This helps you to synchronize data of your primary and backup server, so whenever you ve done changes on either of the server, the result will reflect to the others (e.g. leaving a voic , adding new extension, modifying IVR, etc.). Instant server failover is also available when the connection to the primary server has lost. To setup a server cluster, go to System -> Cluster Management from both primary and backup server. Then click Configure and Form A Cluster. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 6

8 On the Setup Wizard of Primary Server: What Role is this PBX? Master Is here a Hot Standby PBX for this Master? Yes What is the type of the cluster? Select Local Cluster only if you are using frsip Appliances and they are installed locally with HA ports (port 2 or 4) directly connected. Select Global Cluster if the servers are connecting via a DR site. For DR cluster: What is the IP Address of the Master PBX (for DR cluster)? Primary Server IP What is the IP address of the Backup PBX (for DR cluster)? Backup Server IP For Local Cluster: Cluster Synchronization Port: Select port 5, the ports which you have connected between the primary and hot standby unit. (for local cluster) IP Take Over: For instant server failover, select Port 1 or 3 to fail over primary server port 1 or 3 connection on port 2 or 4 of backup server respectively. If you do not need IP failover, select data backup only. (for local cluster only) What if primary is recovered after failure? Select Automatically fall back to Primary or Manually Switch back to Primary.. If the manual option is selected, go to System->Status and click the switch status button to change the status to Active and resume the service on Primary unit. Which Database Should We Use? Choose either Master or Hot Standby PBX dataset. Chosen dataset will write its data to the other server. Once you have confirmed the configuration, click Save Cluster Configuration On the Setup Wizard of Hot Standby Server: What Role is this PBX? Hot Standby PBX Which Database Should We Use? Choose the same setting as you have done on Primary. Once you have confirmed the configuration, click Save Cluster Configuration After filling in the Setup Wizard on both primary and hot standby server, click Perform Data Synchronization on either one of the server ONLY. When done, both servers main and hot standby server status should show as active. If you see an error on either one of them, repeat all of the above steps again. Once done, you may remove the network cable on the temporary network port from the hot standby unit. Then reboot all phones (if already configured) to make sure all equipments are registered to both primary and hot standby server. Page 7 of 36

9 frsip APPLIANCE FRONT PANEL LED SIGNALING From the front panel of frsip appliance, you can find the followings LED which serves as status indicators or alarms. LED SYS RAID CF SMS Cluster PSU TDM FAILOVER 1 FAILOVER 2 Status Off Flashing On Off Flashing On Off On Off Flashing On Off Flashing On On Off On Off ON ON Meaning System is not running or hardware initializing System is booting/shutting down System is operating properly RAID is not running or running with errors RAID is rebuilding / syncing Raid is operating properly Compact Flash is not inserted or has errors Compact Flash is operating properly No SIM card / signal detected Unable to register to network SIM card detected and registered to network Cluster is not activated Cluster is not in sync Cluster in active and in sync One of the power supply modules has failed or no power connected (An audible continuous beep will also sound at the same time. To acknowledge the alarm and turn off the audible alarm, press the red button at the back of the appliance.) All installed power supply modules are operating properly. One or more errors detected with configured frsip Gateways. (E.g. Loss network connectivity, PRI lost framing) No errors detected concerning the frsip Gateways This primary unit is not operating properly and therefore is now failed over to secondary connected on HA-! (Only available when backup cluster is installed) This primary unit is not operating properly and therefore is now failed over to secondary connected on HA-2 (Only available when backup cluster is installed) Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 8

10 frsip APPLIANCE and Gateway CONSOLE ACCESS MANAGEMENT Setup Console Port Connection Parameter frsip appliance can be managed by console access. To connect to frsip appliance console port, configure the hyper terminal of your PC with the following settings: Bps/Par/Bits: N1 Hardware Flow Control: No Software Flow Control: No Console and Web Default Login And Password Change The console default login is same as the server default login username and password. This is a permanent administrator account that cannot be deleted. You will see a login prompt when the after connecting your console access. Login with the following default account: Username: administrator Password: Help Menu Type help after login to get a list of frsip appliance supported console command. Change Default Account Login Password For security reason, we recommend you to change the password of the default administrator password upon your first login. Type password on the command line after login to change your password. Setup Network Via Console Access Type network on the command line to enter the network setup menu, then type in the associate commend number: 1. DNS 2. DHCP 3. IP Configuration from Port 1 to 4, HA port and Uplink port. 4. Virtual Network Interface 5. Static Routes Page 9 of 36

11 Below is a visual guide on console setting: Select which option you would what to change. For IP address, select which port and enter what is required and just follow the on screen instructions. Once you can access the web interface, go to Configuration/Network Settings to save your network settings in database and click Submit and Restart Later. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 10

12 Test Bed Configuration Files Deltapath basic Configuration for XO lab Lab Configuration (Diagram) The following diagram is the configuration used during lab testing. Note: Above lab setup only shows main lab network elements. Page 11 of 36

13 To get to the frsip web interface, you will need to input the IP address of the PBX to a web browser: (Mozilla Firefox recommended) Network Configuration: Set WAN (IAD) & LAN IP You could define your own DNS server IP or use the public DNS IP. frsip could also act as the DHCP server for the phone system. The setting is located at Configuration -> Network Configuration. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 12

14 Make sure the IP address not in parenthesis and the one in parenthesis are the same. The one not in parenethesis is the current IP address while the one in parenthesis is what is stored in databse. If they are not the same, once the PBX is rebooted or shutdown properly, it will always follow the IP address inside the parenthesis. Recommended settings of IP address: Port 1 (Uplink 1) - LAN IP address (Voice) Port 2 (HA-1) - Failover IP of port 1 Port 3 (Uplink 2) - LAN IP address (Data)[if required] Port 4 (HA-2) - Failover IP of port 3 Port 5 - Cluster synchronization port [for redundancy] Port 6 - WAN IP [if available] Page 13 of 36

15 Phone System Dial Plan Routing number and trunk are used for VoIP calls to the other SIP PBX. Instead of creating the extension one by one, use the wild-card extension pattern to route a range of numbers to other peers. To use any of the following pattern, under score ( _ ) must be added before the pattern. Pattern Description Z Single digit number range from 1 to 9 N Single digit number range from 2 to 9 X Single digit number range from 0 to 9. Single to many digit number! None to many digit number Example pattern Description _100X Any 4 digit number starting with 100 _889X! At least 4 digit number starting with 899 _! Any number of digits number (Highly not recommended) _. At least one digit number. _ZN 2 digits number with the first digit cannot be 0 and the second digit cannot be 0 or 1. User configuration: User Profile Administrators can predefine various templates and simply attach a new or existing user to a particular template. If there is a need to make changes to a type/group of users, it can easily be done by Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 14

16 changing the settings inside that particular profile. You must create at least one profile before create any user in the system. Each user profile covers all aspects of account settings from permission group to voic preferences as well as default extension routing. To create a new profile, go to User -> User Profile Settings -> Add New User Profile. For example, you may want to have a Standard User profile which is set to have a maximum of 100 voic messages and by default ring for 30 seconds before sending a call to voic . Then another profile Meeting Room for phones that will ring by default for 60 seconds and do not send the call to voic . When you create a user, you can simply define extension to follow Standard User profile or Meeting Room profile and only enter the user s full name, , and MAC address. The user will automatically be created by frsip with the exact same settings as the selected User profile. Add New User Extension After creating a profile, go to User -> User Settings -> Add New User and fill in the required fields listed below User Extension number: Enter the extension number assigned to this user s phone. Typically a 4 digit number. User Profile: Select User Profile from dropdown menu. Click the HELP Icon to the right of User Profile menus box to view the details of that profile. All fields will be configured using User Profile Settings unless you indicate Customize on a particular field. Company or department: Select Company/Department from dropdown menu. If no company is available to select, create a new group in Group -> Group Settings -> Add New Group. If Call Accounting module is enabled in your system, please refer to Call Accounting section for details of how to create a new company/group. First/Last Name, Employee ID: Information of the user. User User can receive different kinds of system depends on the system setting, including password information, voic message, and receptionist message(crm module only). Mobile: Enter the user s mobile number. Short Dial for mobile: Whenever the short dial extension number is called, the call will be redirected to this user's mobile. Equipment: Select the provisioning type of the user equipment: Determine by MAC address: Enter the MAC address of your phone device (for Polycom, Cisco, or Grandstream GXP2000 only). Press tab after entering the MAC address and the brand of the phone Page 15 of 36

17 model will be detected. Then choose the Model, Line Number, and PBX Registration IP that the user s phone will register to. Own CPE: Any SIP phone device that cannot be auto provisioned by frsip UC Platform. Choose the phone model you are using from x-lite, eyebeam, Polycom Kirk, Polycom HDX/PVX, or others. System will provide you a SIP password for your phone menu configuration to register to our system. Make sure the allowed codec settings only includes those supported by your phone device when you are using own CPE. Otherwise it might result in no audio when making calls. Analogue Line: For analogue devices like fax, credit card machines, or analogue phones. This will only be available when you have added a frsip analogue gateway. Check this radio button and select an available port. Then you may patch your analogue device to the selected port. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 16

18 Extension Configuration To customize each extension number ringing plan, go to Numbering Plan -> Number Settings -> List number -> Extension. The path will display all the extension numbers defined in the server. From there, select the number you wish to modify with its dialing pattern. Page 17 of 36

19 Assigning DID Range to Extension Routing To add new DID range, go to Configuration -> DID Range Settings > Add New DID Range and fill in the form according to your DID range. You will need the following information when setting up your DID range: Your internal extension range. The DID range that the telco provided for you. Number of digits your vendor sends to you on inbound calls Caller ID you are expected to send to vendor when making outbound calls Example of setting up DID range: Range of DID numbers: to Internal Extension Range: 1000 to 1019 Number of digits vendor sends in: 8 (means vendor will send in all numbers XX) Number of digits required by vendor when sending outbound caller ID: 8 (means we need to send XX to vendor) Actual DID Number Range For Dial-In: This is the actual full international number that you want to inform your internal users. The number range you enter here will be automatically used in generating a system personalized informing the user of his/her password as well as phone number. It is important to assign a unique context to DID Range Context if the extension range has the same number of digit as DID number range to avoid call looping. Then assign the incoming SIP peer (e.g. frsip gateway) with a permission group that only includes the DID range context. (Please refer to User/Peers Call Restriction). Extension range and DID range must be consistent. E.g is a valid range of while is invalid. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 18

20 Handset configuration Deltapath being a Polycom arena partner has been fully certified by Polycom to have its endpoints connecting to frsip. Each release of frsip is bundled with a specific set of Polycom firmware which are validated by Deltapath. Changes to frsip s provisioning mechanism and SIP stack are often made in order to optimize its performance for a particular version of firmware. Firmware files are automatically pushed to all Polycom endpoints provisioned by frsip. Global and individual configurations are permitted with Polycom phones under Equipment -> Polycom Settings. To make changes in the Polycom Settings page click the Modify button at the bottom of the screen. Page 19 of 36

21 Trunk Group By creating SIP peers you can have the other peers to pass a call to your frsip. If you need to pass call to other SIP peers, you need to create a routing number associate with a trunk. Go to Numbering Plan - > Trunk Settings -> Add New Trunk and fill in the trunk form: Name: Name of the trunk Rewrite Caller ID for local users: The caller ID number that the other SIP peer will receive from internal users. Auto Resolve: Resolve caller ID from DID range setting. Default Caller ID: The default caller ID defined in General Setting will be used. Set Manually: Edit caller ID by stripping and adding prefix to your original extension. Rewrite Caller ID For Non Local User: The caller ID number that will be used if the caller to this trunk is not one of the frsip extension or if it is a forwarded call. Append Prefix to Extension Dialed: Add number(s) in front of number you have dialed. Number of Digits to Delete From Original Extension Dial: X number of digits will be deleted from the prefix of the number you have dialed. SIP Peer: Call will be passed to the selected SIP Peer/SIP Peer Group. Preferred Codec: System will force to use this codec when call to this trunk. Recommend Disabled unless you know every party supports this codec. Force DTMF Mode: System will force to use this DTMF mode when call to this trunk. Recommend Disabled. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 20

22 SIP Peer: Set NBS SIP Port The frsip UC Platform allows you to peer with other frsip UC Platforms or just about any SIP based PBX with only a few clicks. Besides simple peering, you may also enable auto fail-over SIP trunks. Once enabled, a SIP Peer status will be checked via SIP PING on a regular basis. Page 21 of 36

23 Music On Hold The Music on Hold feature provides you with the capability to provide the music selections (MP3 format) of your choice for callers calling in or placing calls on hold. Music loaded on the Global group (default) will be used unless you ve loaded a music file for another group. Analog Gateway Once you have all the configuration details entered, you can go to Configuration -> PRI/BRI/FXO/FXS Gateways -> List Gateway and click on the PROVISION icon To add a real fax line or an analogue phone line, record the port number you selected when creating analogue user and connect your analogue device into your frsip FX series gateway patch panel port. When you are finished check devices for dial tone. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 22

24 System Extension To define system extensions for dial-in, you can go to Numbering Plan -> Number Settings -> System Extension. Remote Dial In (AKA DISA): The DISA main number is for users to make toll calls via the system when they are not in the office. Recommend to use a number within the DID range for users to use the feature. Page 23 of 36

25 Voice Mail Direct Access: For users to access their voic box internally, which will be loaded to the phone message button that is auto provisioned by frsip. Only user password will be asked when calling this extension. Note that you must reboot all phones after changing this extension. Remote Voice Mail Center: Same as voice mail direct access. This extension will serve as an external access point for users to call in when they are not in the office. User ID and password will be asked for authentication. Recommend to use extension within DID range. Conference: The main conference center for users to call in to access conference bridges. Recommend to use extension within DID range. User Management Center: A centralized access point for all system services, including user roaming, personal password change, conference room password change, call center agent sign in/out, call center agent password change, DISA access point, and voic center. Call Parking: Allows user to put a call on hold at one extension and continue the conversation from any other extension. Usage of Call parking will be discussed in later chapter in frsip feature guide. Digital Recorder: This feature will let your phone act as a recorder. If you assign an extension number to digital recorder, you can simply dial that extension number and you will be able to hear a beep and start recording the conversation. Auto Attendant The Interactive Voice Response (IVR) module allows the system to work as an auto attendant to incoming calls. It consists of voice menu and IVR prompt settings, which allow voice recordings to be made and then applied to the voice menu(s) desired. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 24

26 Voice menu title: Name of the voice menu. Alphanumeric characters only Include Permission Group: The access right you grant the voice menus will determine what extension the caller can enter. Number of times to play this menu: : Enter Number of times this Voice Menu Title should play (before exit). Timeout for waiting user input (seconds): : Enter number of seconds allowed for user voice input (before timing out). Timeout for entering user input (seconds):: Enter number of seconds allowed for user to key input (before timing out). Prompt Settings: : Select a voice prompt to play for this menu. The voice prompt you have recorded will be listed here. Action before exit: : Select the IVR routing after call in to this menu from the followings: Creating IVR Menus Go to IVR -> Voice Menu -> Add New Voice Menu and fill in the followings: Voice menu title: Name of the voice menu. Alphanumeric characters only Include Permission Group: The access right you grant the voice menus will determine what extension the caller can enter. Number of times to play this menu: Enter Number of times this Voice Menu Title should play (before exit). Page 25 of 36

27 Timeout for waiting user input (seconds): Enter number of seconds allowed for user voice input (before timing out). Timeout for entering user input (seconds): Enter number of seconds allowed for user to key input (before timing out). Prompt Settings: Select a voice prompt to play for this menu. The voice prompt you have recorded will be listed here. Call Center Call center on frsip is used for the purpose of receiving and transmitting a large volume of requests, which contains by 4 parts: Call Queue - Store incoming calls and send to next available agent. Agent - Need to login a phone to answer calls sent from the queue. Call Center Monitoring - Use frsip switchboard to monitor all queue and agents status Call Center Report - Generate a statistical report for all queues and agents To following shows the flow of setting up call center: Create Call Queue Create Agent and assign them to desk phone Create Agent Group and associate the group with agent(s) and queue(s) Create a virtual number and select call queue in the routing Agents call to UMC to login Call comes in and sends to agent from queue. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 26

28 Call Queue To create a call queue, go to Call Center -> Queue then click the Add button on the top left corner of the table. The queue screen will appear on screen with the following fields: The below table shows the value you can set for a queue: Field Name Description Valid Value Queue Name Name that represent this queue.. Number of Number of queue slot to show on status screen Queue Slot Used for service level statistics. A call will be consider as Service Level successful if agent can pick it up within a number of seconds entered. After a successful call, how long to wait before sending another Wrap Up Time call to the agent again Maximum Maximum number of people waiting in the queue Caller Strategy Group Call Recording Agent Group Priority Alphanumeric Numeric only Numeric only Numeric only Numeric only. 0 for unlimited. Method to use when distribute calls to agents: Ring All - Ring all available channels until one answers All Least Recent - Ring interface which was least recently called by this queue Least Recent Fewest Calls - Ring the one with fewest completed calls from this queue Fewest Calls Round Robin(evenly distributed) - Take turns ringing each available agent. Round Robin Round Robin(prioritized) Always ring the agents in the preset sequence. Only ring the second agent if the first one is busy. Random Random Randomly ring an agent Assign which group this queue belongs to. Only the ACL user Select from list assigned with this group can monitor this queue. Record all conversations going through this queue. Note that Yes/No you need to have call recording module to have this feature Caller from this queue will be distributed to the included Agent Group according to strategy. Queue with higher priority will send calls to an available agent first. Low priority queue will not send out any calls until the higher priority queue is empty. Select available group from left and move to right Drag and drop the current queue to a priority level. Page 27 of 36

29 By Default only priority Level 0 will be listed. Click Append a new level to add next level priority or highlight and click Remove the selected level to remove one leve. LCD Display Current Queue represent the one that you are creating, drag and drop it to different level s folder to assign priority. The caller ID name and number that the agent will see on the phone display screen with the following options Display Name: Show caller name Show queue name Show caller name and queue name Show custom name Show caller name and custom name Display Number: Show caller number Show custom number Show caller number and custom number Display Name: Alpha Numeric Display Number: Numeric characters only Announce Options Announce Position Yes: Announce the current position in queue to caller No: No position announcement Limit: Announce the position to caller only if the current position is within the position limit/more value Yes No Limit Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 28

30 Announce Position Limit/More More: Announce the position to caller if the current position is within the position limit/more value. Otherwise it will announce There are more than X number of persons waiting in this queue Works together with the Announce Position field to control and position announcement More Numeric characters only Yes Announce Hold Time Announce the estimated waiting time to caller No Hold Time/Position Announce Frequency Round Up Hold Time in Seconds Only available if Announce Hold Time is checked. Enter X number of seconds between each time/position announcement. Works together with Announce Hold Time. Round up the announcing time to caller in seconds Once Numeric characters only Numeric characters only Voice Mailbox Voice mailbox to use after caller pressing the exit key Select from list Minimum Wait Before Making Hold Time/Position Announcement Periodic Announcement Message Frequency Whenever your position in queue has changed, the system will update you with time/position announcement. Enter a Numeric minimum wait time in seconds so the system won t do the characters only announcement in a short while. How often to make any periodic announcement in X number of seconds. Enter 0 to turn off. Numeric characters only Prompt Options English (US) Language Language preference of your call center. English (British) Chinese Page 29 of 36

31 Prompts settings Other Options Allow a Caller To Wait While No Agents are Logged In Wait Time Before Trying Another Agent For each of the prompt settings, select System Default to use the pre-recorded system voice prompts. Otherwise, you can add a new prompt at Configuration -> Voice Prompt Settings and add a new prompt from there. Then come back to this page and select it from list yes - callers can join a queue with no agent or only unavailable agent no - callers cannot join a queue with no agent strict - callers cannot join a queue with no agent or only unavailable agents loose - same as strict, but paused queue agents do not count as unavailable Time to let phones ring before we consider this a timeout and send to the next available agent Disconnect Callers In Queue All callers waiting in queue will be disconnected when all When All Agents agents logged out Are logged out Report Callers Hold Time To Agent Include estimated hold time in position announcements. Either yes, no, or only once. Hold time will be announced as the estimated time, or 'less than 2 minutes' when appropriate Select from list Yes No Strict Loose Numeric only Once Only Exit Voice Mail Key to press on phone pad to exit the queue and go to the 0 to 9, * or # selected voic box Exit Digit Agent Ring Time to wait in seconds before redialing another call to Numeric only Duration agent Auto Pause Pause an agent if the agent missed a call Yes/No Yes No Yes No Agent and Agent Group After creating a queue, agent group(s) and agent(s) are needed so queue can distribute calls to them. Agent is an account that can login to any phone via the User Management Center or frsip switchboard. When a call comes in, the queue will distribute the call to agent group(s) and ring to agent phone(s) according to the strategy of the queue. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 30

32 To create Agent Group, go to Call Center -> Agent -> Agent Group. Click Add on the top left corner of the table and fill in the following fields on the popup window: Field Name Description Valid Value Name Name of the agent group Alphanumeric The selected agents will receive calls when a queue Select agents to the Agent Users distribute a call to this agent group right side list Queue that this agent group is responsible for. Select a queue from Attach a new queue and a new queue tab will appears on the bottom. Your current agent Queue group will show up as ---This Agent Group---. Press Select from list the up and down arrow on the right to change priority of distributing calls from queue to agent group. To create Agent, go to Call Center -> Agent -> Agent User then click Add on the top left corner of the table and fill in the following fields on the popup window: Field Name Description Valid Value Agent ID For login to phone via UMC Numeric only Password /Confirm First/Last Name Group ACL Group Call Recordings Agent s Seat Agent Group Your agent id pin will be asked when log in to a Numeric only phone via UMC Name of the agent Alphanumeric Group that this agent belongs to. Only the ACL user Select from list assigned with this group can monitor this agent The access right that this agent has when login to Select from list frsip switchboard. Turn on call recording to record all calls answered by this agent. Note that you need to have call recording Yes / No module to have this feature Select a phone (user extension) from list. Agent only Select from list allows to login to selected phones Groups that this agent belongs to. Identical to agent Select from list users list in Agent Group s Agent Users setting. Page 31 of 36

33 Creating the queue Please refer to the data discussed above about call queue. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 32

34 Creating Agent Group Please refer to the data above about Agent and Agent Group Page 33 of 36

35 To enable calls to pass through the call center, a dial in number or a hotline must be created, either via a virtual number with DID or an extension number with DID. Once the number is created, the dialing plan must be pointed to ring to a QUEUE. Copyright 3/22/2013. XO Communications, LLC. All rights reserved. XO, the XO design logo, and all related marks are trademarks of XO Communications, LLC. All other trademarks are property of their respective owners. 34

36 Extension Status To monitor real time phone status, go to the System -> Status page. This page displays the actual status off all registered phones in the server. Software Version Page 35 of 36