Avaya IP Office 500v2 Quick Setup
Scenario
Version 11.1
Licenses - Essential Edition IP Endpoint (J Series) SIP Trunks
Install System SD card in System SD card slot
Install all base cards and trunk cards
Connect power to system, wait till the system has fully powered on
(When powering on you will see red lights solid, red lights flashing, heartbeat light on first port of each card flashing every 30 seconds)
System Default address are below (however if the system is on a network with DHCP then the system will obtain an IP address from the DHCP server
LAN1 192.168.42.1
LAN2 192.168.43.1
If the system is not on a network with DHCP and you have connected your PC directly to the LAN1 port. Configure your PC/Laptop/Notebook to be on same subnet as IPO LAN1
Example
IP Address: 192.168.42.10
Mask: 255.255.255.0
Gateway: 192.168.42.1
Open IP Office Manager
Select File, Open Configuration
Select system using broadcast address 255.255.255.255
Or enter IP address of system and refresh and select system
Select OK
Setup new passwords
(Write these down)
Enter a password for Administrator login
Enter a password for security login
Enter a System password
The default passwords on previous version were the below (earlier versions did not ask for these passwords to be changed on setup)
Username: Administrator
Password: Administrator
Username: security
Password: securitypwd
Select OK
Enter Administrator for Service Username and the new password created
Set System Mode as IP Office Standard Edition (if using PLDS licensing)
(Set System Mode as IP Office Subscription (If using Subscription Licensing, see next page)
Set System Name and Locale
Set LAN1 Interface network details
Set DHCP Mode to Disabled
Set DNS Server (primary DNS, the secondary DNS can be configured later)
Select Save
If using Subscription licensing
Set System Mode as IP Office Subscription
Set System Name and Locale
Set Customer ID as per email received for Subscription Licensing
Set License Server as per email received for Subscription Licensing
Set LAN1 Interface network details
Set DHCP Mode to Disabled
Set DNS Server (primary DNS, the secondary DNS can be configured later)
Select Save
In the System tab
Confirm the Name and Locale are correct
Configre the TFTP and HTTP Server as the LAN1 IP Address
Set Messaging Server as Avaya Spaces
Set Time Setting Config Source as NTP
Enter a Time Server Address as 0.au.pool.ntp.org
Set the Time Zone as required
Select System from Navigation Pane
In the LAN1 / LAN Settings tab
Confirm the IP Address and IP Mask are correct
Confirm the DHCP Mode is Disabled
Select System from Navigation Pane
In the LAN1 / VoIP tab
Tick SIP Trunks Enable
Tick SIP Registrar Enable
Tick Enable RTCP Monitoring on Port 5005
Set Keepalive Scope RTP-RTCP
Set Initial Keepalive Enabled
Set Periodic Timeout 5
NOTE: you can enable TLS if required
When TLS is enabled
when connecting a J Series handset, you can enter the LAN1 IP Address of the system when prompted for the provisioning server
Example: https://192.168.86.200
If TLS is disabled
when connecting a J Series handset, you can enter the LAN1 IP Address of the system when prompted for the provisioning server but you will need to first change the https to http
Example: http://192.168.86.200
Select System from Navigation Pane
In the DNS tab
Confirm the DNS Primary IP Address is correct
Enter the DNS Secondary IP Address
Select System from Navigation Pane
In the Telephony / Telephony tab
Set the Hold Timeout (secs) to 120
Set the Default Currency to AUD
Tick DSS Status
Un-Tick Inhibit Off-Switch Forward/Transfer
Un-Tick Drop External Only Impromptu Conference
Select System from Navigation Pane
In the Telephony / Tones & Music tab
Set the Hold Music System Source as External if using an external music on hold player connected to the audio port on the back of the system
If using an internal WAV file, then leave as WAV file
Return to this page and carry out the below
Open Manager again and login to the system
Select File, Advanced, Embedded File Management
Login to the system
Select System SD
Select SYSTEM
Select PRIMARY
Select File, Upload File, select the holdmusic.wav and upload. Check the Files to confirm the file has uploaded
To test hold music dial *341# from a handset
NOTE: The file must be named holdmusic.wav and must be in the correct format (right)
Select System from Navigation Pane
Enter the SMTP settings (these will allow voicemail to email and System Events/Alarms to be sent)
Select System from Navigation Pane
In the VoIP / VoIP Security tab
Set the Default Extension Password (numbers only)
As default it must be 10 digits but can be adjusted in security settings
Set the Media as required
Disabled will mean traffic is RTP and not secure
Preferred will mean traffic is SRTP if possible and secure
Select Line from Navigation Pane
Right click and add new SIP Line
(or if you have a SIP Line template, right click and select New From Template, Open From File, change the File type in the bottom right to Template File.xml and locate and select the SIP Template to import)
In the SIP Line tab
Set the Line Number 17
Enter the ITSP Domain Name supplied by the SIP Provider
In the Transport tab
Enter the ITSP Proxy Address supplied by the SIP Provider (this can be the Domain Name and IP address or if left blank it will use the ITSP Domain Name
Set the Protocol and Ports (set any explicit DNS Servers) supplied by the SIP Provider
In VoIP tab
Set the Codec Select to Custom
Configure the Selected Codecs to match the supported Codecs of the SIP Provider
Tick Re-Invite Supported
Tick PRACK/100rel Supported
Set the Fax Transport Support to match the supported Fax Transport of the SIP Provider
Set the DTMF Support to match the supported DTMF of the SIP Provider
In SIP Credentials tab
Select Add
Enter the SIP Credentials supplied by the SIP Provider
Select OK
In the Call Details tab
Select Add
Set the Incoming Group as 17
Set the Outgoing Group as 18
Set the Credentials to the one configured
Set the Max Sessions to match the SIP trunk provided by the SIP Provider (this would usually also match the total SIP trunk license)
Set Local URI Display and Content to Auto
Set Contact Display and Content to Auto
Tick P Asserted ID and set Display and Content to Auto
Leave all other settings
Select OK
In the Call Details tab
Select Add
Set the Incoming Group as 18
Set the Outgoing Group as 17
Set the Credentials to the one configured
Set the Max Sessions to match the SIP trunk provided by the SIP Provider (this would usually also match the total SIP trunk license)
Set Local URI Display and Content to Use Internal Data
Set Contact Display and Content to Use Internal Data
Tick P Asserted ID and set Display and Content to Use Internal Data
Leave all other settings
Select OK
After you have completed the configuration and saved to the system
Select File, Advanced, System Status
Login to System Status and check Trunks and Line 17 to confirm it is registered
You can also monitor the trunk in real time while testing inbound and outbound calls
Select Users from Navigation Pane
Delete the user Remote Manager
If you have Digital and/or Analog extension cards, the users will be automatically created
Edit the Users as required
If using JSeries SIP handset, you will need to create a new user for the phones
Right click on User in the Navigation Pane and select New
In the User tab
Enter the Name
Enter the Password
Enter the Full Name
Enter or Update the Extension number (if required)
(if updating a Digital or Analog extension in user, you will then have to update the Extension number in the Navigation Pane / Extensions to match the users new extension number)
(If creating a user that will be using a JSeries SIP handset, once the user has been configured and you click OK to save, you will be asked to create an Extension, you will need to select SIP Extension and configure the Login Code to match the Supervisor Login Code of the User)
Enter the Email Address
In the Voicemail tab
Enter the Voicemail Code
Tick Voicemail On if the user requires voicemail
Confirm the email address
Select the Voicemail Email delivery option as required
In the Telephony / Supervisor Settings tab
Enter the Login Code (numbers only)
In Telephony / Multi-Line Options tab
Tick Reserve Last CA
In the SIP Name and Contact, Enter the telephone number that the user will display when making outbound calls
Select OK to create the User
If the User created has no Extension matching the Users extension number, then a window will ask if you want to create a VoIP extension with the number
Select SIP Extension and enter the Phone Password to match the User Login Code (numbers only)
Select Extension from Navigation Pane
Confirm the SIP Extension has been created and the Base Extension matches the User Extension
It is recommended to disable Allow Direct Media Path on the SIP Extension in the VoIP tab
Select Groups from Navigation Pane
In the Group tab
Configure the details as required in the Group tab (additional group can be configured and used in the Overflow tab to route calls to other users)
Set the Group No Answer Time (sec) as required, any calls inbound to the group not answer will be sent to the group voicemail once this timer has been reached
If the customer would like to Night mode to be automatic, then a Time Profile would need to be created Time Profiles in the Navigation Pane and then the Time Profile would be set against the groups Day Service Time Profile
When a group is in Night mode calls route to the Night Service Destination, if set to None; callers will go straight to the Main group voicemail
In the Voicemail tab
Enter the Login Code (numbers only)
Enter the email address
Select the Voicemail Email delivery option as required
Queue announcements can be enabled and configured to play to callers in the queue of the group
Timers can be adjusted as required
A short code of *91N# can be used to record the first announcement greeting
A short code of *92N# can be used to record the second announcement greeting
(N is replaced with the group number, for example *91200#)
In the SIP tab
In the SIP Name and Contact, Enter the telephone number that the user will display when making outbound calls
Group Voicemail Access
If a group will have voicemail
You will need to create a short code to access the group voicemail to retrieve the voicemails left and record the greeting
select Short Code from Navigation Pane, right click, and select New
Create a new shortcode (example below) to allow access to the Main group voicemail
If you need the group voicemail to indicate on a user’s handset and the handset to retrieve the voicemails left
In the User, select Source Number tab
Add in a Source Number (example right)
where you have the letter (capital H) followed by the group name
Select RAS from the Navigation Pane
Delete the entry DialIn
Select Incoming Call Route from Navigation Pane
Delete the entry for Destination DialIn
Update the default entry with Incoming Number blank
Set the Line Group ID to 17 to match the SIP trunk line ID
Set the Destination as required
Add any additional incoming numbers / DDI numbers and set the destination as required
select Time Profiles from Navigation Pane
If required created a Time Profile to be applied to the group for automatic night switch
Remember to return to the group programming to apply the time profile created
Delete the entry for Destination DialIn (this will have a blank entry if the Remote Manager user has already been deleted)
Select User Rights from the Navigation Pane
Delete the default User Rights
If required create a new User Right and assign the Users as members of the User Right
Button Programming can be configured and applied to the User Right
Users can then be selected as Members of the User Right and the button programming will apply to all members of the User Right
If required, an Auto Attendant can be created and set as a destination for a number in the Incoming Call Routes
Right click and select New
Enter a name of the Auto Attendant
Set the Maximum Inactivity (the call will be redirected to the fallback destination in the incoming call route assigned to the auto attendant if the caller has not selected an option after hearing the greeting and this timer has been reached)
To record the greeting of the auto attendant, dial the shortcode assigned to the Menu Option (*8401 in the example below)
In the Incoming Call Routes, select the telephone number and assign the Auto Attendant as the destination with a fallback destination as a group or user
Select ARS in the Navigation pane
In the 50 Main ARS
Update the Codes
The code right will allow calls outbound via the SIP trunk
Note the Code is N; (N followed by the semicolon)
The code right will allow calls to local number beginning with 3 outbound via the SIP trunk
Note the Code is 9XXXXXXX so any local number dialed that is 8 digits long beginning with 9 will be prefixed with the telephone number 03 and sent to line
For example 0398722999
Add in all local number codes required
Select Add
Once the license has uploaded successfully a window will advise this
Once the configuration has been completed
Select File, save configuration, select Immediate
This will send the configuration back to the system and reboot the system
It is always recommended to always save a config offline as a backup
Select File, close configuration
Wait till the system reboots and is accessible again
Select File, Login
Select the system and login
Check and confirm the system configuration is correct and licenses are correct
Power on the J Series phone
Select No when asked if you want to activate Auto Provisioning
Wait till the J Series phone has obtained IP address details from the DHCP server on the network
(If required you can select Admin and login with code 27238 and select Select IP Address and Ethernet IPv4, Disable Use DHCP and enter manual IP address settings to be used on the phone)
Select OK when prompted to Enter Provisioning Details
if TLS is disabled, set the address to http:// then enter the LAN1 IP address of the system
Example http://192.168.86.200
If TLS is enabled, then leave the address https:// then enter the LAN1 IP address of the system
Example https://192.168.86.200
Select Save, The J Series phone will reboot
You will then be asked to login with the extension and login code (enter the details required and select Enter) If required the phone will upgrade firmware then login as the extension and user (if the user matches the extension number and login code)
In Manager, select File, Advanced, System Status
Login to the system
Select Extensions and confirm the phones are showing correctly against the extensions
Select Trunks and the SIP trunk line 17. Confirm the SIP trunk is registered