NETWORKING LIKE NEVER BEFORE
EACOM Business VoIP Solution Training 04/2009
Confidential and Proprietary, Eacome Inc.
EVS3080 IP PBX
SD Card Slot
FXO Ports
PBX LAN port
PBX WAN port
FXO Ports LED, RED means online
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Connect the IP PBX to Your LAN
Router
FXO Ports Ethernet Switch
FXS Ports
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Access the PBX’s WEB GUI vThe default IP address of WAN is 192.168.1.100 vIf the PBX’s IP address is different from your LAN IP address, you need to manually change IP address of your PC to 192.168.1.101, then connect both PBX and PC to the same LAN
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EVS3080 Built-in Web Server Provisioning Interface
ONLY Firefox web browser is fully supported now
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How to Change Configuration of PBX
When any changes made, you need to save the change and apply the changes so it can be effective immediately
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Setup the IP PBX vSetup Users vSetup out bound call vSetup inbound call vSetup switch board vSetup other features 4Audio Conference 4Call queue 4Other PBX Features
vUpdate PBX Firmware
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Setup Users vThe default users are from 2000 – 2049 vYou can bulk add users or add individual user vSetup user specific settings 4SIP password 4Voicemail password – usually the extension number 4DTMF mode – out band RFC2833 or in band audio 4Enable Re-invite for the users in the same LAN with IP PBX, disable Re-invite for the remote users 4Choose dial plan for users (after the outbound call setup) 4User defined call forward, find-me, follow-me services
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User Management
Select one or more users to configure
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Click EDIT to change configurations or DELETE to delete users
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User Configurations
Each configuration item has online tip notification to assist user
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Bulk Add Users
Specify the starting extension and number of continuous extensions to be added
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User Follow me Configuration
Configure follow me features on per extension basis. Can ring number one by one or simultaneously
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Setup Outbound Call vConfigure the FXO hardware in system-hardware to your regional setting, e.g. if you are in US choose country to US vConfigure the FXO trunk 4each trunk has several FXO ports, the ports in each trunk share the same calling rule 4Setup busy detection ON and polarity switch ON so PBX can correctly hang up the call 4Setup gain for each FXO port
vSetup outgoing calling rule for each trunk vDial Plan 4 Prefix based number routing 4 Dialing out number policies 4 User group based calling rights control
vSIP Trunking 4 Can setup unlimited SIP trunk to the VoIP service providers.
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FXO Hardware Parameter Setup
FXO port information, you can also change the country settings to match your local network
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FXO Hang-up Condition Detection Setup
FXO related information can be setup in trunk menu. FXO support busy detection or polarity reversal to determine the hang up condition
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Outbound Call Control – Define FXO trunk
Give trunk a name and select the FXO ports should be included in the trunk
Define the trunk features, most importantly you should decide how FXO decides the hang up condition, use polarity reversal or busy tone detection. If busy detection is enabled, the busy tone frequency and cadence has to be set to the country standard where the EVS3080 is deployed. Confidential and Proprietary, Eacome Inc.
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Outbound Call Control – Define Outgoing Calling Rule
For each SIP or FXO trunk, you need to specify a calling rule which defines what calling prefix should be used to access the trunk
You can also specify a fall over trunk when the trunk is unavailable due to busy or other conditions, the call can be made through fall over trunk
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Outbound Call Control – Define Dialplan
A dialplan is the set of calling rules and special features that user can use. Each user can have its own dialplan setup.
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Setup SIP Trunk
Click New SIP trunk to add outgoing sip trunk
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Setup SIP Trunk Setup
Service providers’s domain name or IP Address, default port is 5060 or use :xxxx to specify PBX’ B’s SIP registration name and password
SIP trunk options
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SIP Trunk Status
SIP trunk registration status
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Inbound Call Control v Multiple Inbound Trunking Interface Options 4 Integrated PSTN analog FXO gateway, configurable 2/4/8 FXO ports 4 Integrated unlimited VoIP trunking interface seamlessly connected with multiple VoIP service provider, with call in/call out. 4 Integrated SIP peering capability enable distributed EVS3080 deployment 4 Enable enterprise with multiple local numbers to more conveniently connected to customers
v Powerful Number Based Call Routing 4 Redirect 4 Redirect 4 Redirect 4 Redirect 4 Redirect 4 Redirect
to to to to to to
auto attendant extension number voice mail call queue audio conference room voice mail recording
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Inbound Call Control vEach incoming call rules apply to each FXO/SIP trunk 4Call can be directed to different extensions 4Call can be directed to auto attendant or other voice menus 4Call can be directed to conference room or call queue
vTime based rule can be defined for each incoming call rule
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Inbound Call Control
You can also define callee or caller number based matching
Define how call should be directed, can be call queue, conference, voicemail or other special routine
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Switchboard Setup vAuto attendant 4Upload voice greetings or record live through phone 4Can specify the time triggered auto attendant 4Can be a number or voice prompts or customizable IVR
vOperators 4Can specify multiple operator numbers 4Ring together or sequentially ring within operator groups 4Support blind or non-blind call transfer 4If the transfer failed, the call can be directed to operator again
vCalling Status Monitoring and Control 4Can install web based calling status monitoring and control application
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Switchboard Setup
Use switch board options to define voice menu, ring group, upload voice prompt files, define user directory and time based routing rules
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GUI Based IVR Menu
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PBX Feature vCall Forward 4 Individual setup of the call forward on conditions
vCall transfer 4 A calls B, B push ‘TRANSFER’ button, dial C’s number and push ‘SEND’ or ‘#’, A and C can talk
vCall Park 4 A calls B, B transfer the call to 700 and hang up. B goes to another location and dial the 700 to retrieve the call
vCall Pickup 4 A calls B, B and C are in the same user group, C can get the call by dial “*8”
vCall Hold 4 A calls B, B push ‘HOLD’ button and dial C’s number, B can switch between the call to B and C by push ‘HOLD’ 4 A can push ‘CONFERENCE’ to start conference call
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Configure the Call Park Features
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Calling Queue– Automatic Call Distribution vCustomizable queue number vPriority based operator routing, Ring Together or Roundrobin vConfigurable Voice Greetings with prompt files upgrade capability vCan have customized IVR for each queue
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Calling Queue Setup
Define the extension of the call queue, ring strategy, background music
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Other Functions vConfigurable Music-on-hold 4Allows selection among multiple music files uploaded from PC 4Audio convert tools for converting mp3 files to music RAW files
vUnified Messaging 4Dedicated voice mail for each individual user 4USB flash based disk or hard disk based massive storage 4Support message waiting indication message 4Automatically forward voice mail to user specified email account
vIP Broadcasting 4Dial 1800, all the users in group 0 will hear the broadcast message
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Music-on-hold Setup
Use tftp or http to upload music files to the PBX
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Audio Conference Features v Two built in conference room with 10 participants each v Customizable access number, VoIP or PSTN v 5 Ways to dial into conference room 4 Call from PSTN or VoIP directly to the conference room 4 User dial the conference room extension 4 Transferred from AA or operator 4 Dial from other PBX 4 Customizable IVR
v Conference room PIN management 4 Conference moderator has special PIN 4 Normal participants have their own PIN
v Advanced Conference Control 4 Real time conference status monitoring 4 Participants control from web or DTMF, invite participants, convert to the moderator
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Audio Conferencing Setup
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Update PBX Firmware
Put image file in tftp or http server directory, specify the sever name or URL, then upload. After uploading, reboot the PBX
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SIP Server and NAT/Firewall Traversal vField-proven SIP Servers 4SIP Registration 4SIP Proxy 4SIP B2B UA
vNAT/Firewall Traversal
EVS
308
0
EVS3080 forward voice media flow in between
4Built-in Session Border Controller (RTP Relay) 4SIP regular refreshing registration 4Media flow relay AD SL
Cab le
Voice media flow is blocked by firewall
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Typical PBX Network Setup Scenario – DMZ Mode EVS3080
Linksys W54GT Ethernet Switch
Router forward all request initiated by Internet user to the Router’s public IP to PBX DMZ
WAN
EVS3080B
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Setup PBX in Router’s DMZ v Purpose 4 PBX it 4 PBX 4 PBX PBX
can expose to the Internet so it is easy for other phones or PBX to find can forward media between the PBX/Phones behind the NAT/Firewall only sees only voice traffic and other traffic still go through router, so can perform better
v Guidelines 4 Go register a dynamic domain name (DDNS) if router uses dynamic IP address, e.g. dyndns.org, www.3322.org 4 Setup router’s DDNS according to specific router instructions 4 Put PBX in the router’s LAN and assign a static IP address to PBX 4 Enables router’s DMZ, pointing to PBX’s IP address 4 Enables router’s port range forwarding to reserve SIP and RTP ports that PBC will use 4 Setup DMZ support in PBX, tell PBX to use router’s public IP in SIP signaling 4 Try to register or access PBX web from other broadband link
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Setup DDNS in Router
Choose the DDNS service provider name, DynDns is preferred
Fill in the domain name, user name and password you got when you registered the domain with DDNS service provider
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Setup DMZ in Router
Assign a static IP for the PBX and enable router’s DMZ to point to PBX IP
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Assign Static IP to WAN in PBX
Please use different subnet for LAN a WAN
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Enable DMZ support in PBX
User router’s DDNS domain name for external IP address
User router’s static IP address external IP address
Specifies WAN’s subnet information
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User Setup in DMZ mode
In DMZ mode, for the extension that registers to the PBX from Internet, please disable “Can Re-invite” option in the user property, otherwise one way voice can happen
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EVS3080 Deployment Scenario 1 – Trust Host Based Trunking 1. 2. 3. 4.
PBX trust each other and does not require authentication – no need to register to other PBXs Each PBX has its network ID – something like area code in PSTN Each PBX defines prefix dialing to interconnect with other PBX Each PBX has its own connection to PSTN, flexing routing rules can be set to land the PSTN calls according to best rate among the PBXs
EVS3080 A 81xxx
EVS3080 83xxx
EVS3080 84xxx
EVS3080 85xxx
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EVS3080 Deployment Scenario 2 – Authentication Based Trunking 1. 2. 3.
A and B are independent of each other and have their own PSTN trunking interface A and B has unified extension plan, all the extensions can be accessed from A and B A and B has different local PSTN access number but unified auto attendant
EVS3080 A
1. 2. 3. 4. 5.
EVS3080A and EVS3080B has independent extension group, A group is 3xxx, B group is 2xxx EVS3080A log in EVS3080B with 2xxx account, EVS3080B register to EVS3080A with 3xxx account Adding VOIP outgoing dialing rule, in EVS3080A add outgoing SIP account and dialing 2xxx to reach B, same setup in EVS3080B EVS3080A and EVS3080B dial out to PSTN with prefix +9xxx EVS3080A can dial out to EVS3080B’s PSTN with prefix +39xxx, EVS3080B dial out to EVS3080A’s PSTN with prefix +29xxx
EVS3080 B
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EVS3080 Deployment Scenario 2 – Centralized Management
EVS3080 A
EVS3080 B 1. 2. 3. 4.
All the SIP phones in group B are registered with EVS3080A, EVS3080B only provides local PSTN access and trunking interface No special dialing rules are needed for the calling extensions EVS3080A registers with EVS3080B to provide access to EVS3080B’s PSTN trunking interface Prefix based dialing for PSTN dialing out, +9xxx for EVS3080A local access, +8xxx for EVS3080B local access
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