Limit Number of Concurrent Calls.
(Provide busy for second call)
First Option
Configure the phone as below:
Configure only 2 call appearances for the phone
Restrict Last Appearance? Y
This configuration will allow the phone to make transfer and conference
A second call to the phone will get busy.
Note that for the ADA phones you will need to configure only one call appearance. The ADA phone
will be able to make transfer and conference with just one call appearance
Option 2
The Limit Number of Concurrent Calls (LNCC) feature is used to control the number of concurrent
incoming calls, and to change Multiple Call Appearance phone to a Single Call Appearance
phone. If a user is active on a call and receives an incoming call, if the LimitInCalls feature is
enabled, the caller gets the busy tone.
This feature must be activated on the Avaya Aura ® Communication Manager.
Hot line
This feature will work for analog phones.
It will dial a destination configured as soon as the handset is picked up.
Create abbreviated dial group
Assign the group to the user and to the hot line
Supported also for J100 phones
Use 46xx settings file with below parameter
SET HOTLINE 3127
The destination number will be dialed immediately.
I am now looking for a timer is possible??
3127 is the destination number that will be dialed
PCOL – PVR - private line. (Works for H323 currently)
Add button of type PCOL
FYI, adding support for the PCOL feature on SIP endpoints in CM/SM/SMGR is not a small
undertaking. We could add it as an ADA-only feature, such that real SIP endpoints do not need to
support it, but that of course leads to a feature debt if/when the customer migrates from CS1k
endpoints to SIP endpoints.
Before we start we do need to make sure the PCOL feature is close enough to the PVR feature, so we
need Eyal or Shlomi to test out the H.323 version of the feature.
Log call options
It is possible to log only calls to main number and not bridged lines for J100 and possibly other
phone types
Team button
Team button feature works for both H.323 and SIP endpoints, which incorporates BI functionality,
but also functions as a call pickup button when a call arrives.
Create a team group, assign users to it and assign the team button key to the users.
Different ringtone for internal and external calls
See feature description and implementation book - Defining Distinctive Ringing.
Basically need to configure the number of rings for internal and external under:
The Feature-Related System Parameters screen displays the Distinctive Audible
Alerting field only when the Tenant Partitioning field on the System Parameters
Customer Options screen is set to n. If the Tenant Partitioning field is set to y, you
must change the Distinctive Audible Alerting area on the Tenant screen.
Change tenant x
In addition, you may modify the behavior for transferred calls
Change the station ring pattern association – only for H.323 phones??
Personalized ringing pattern
Intercom Group
Intercom group can be created between 2 phones and more
You can create up to 256 intercom groups per standard and up to 1024 intercom groups with
SA9035 - Increased Intercom Groups on one server that runs Communication Manager.
• Each group can contain up to 32 extensions in it.
• You can assign the same extension to different groups.
• Intercom calls are possible only between extensions in the same group
Create an intercom group
Use sat
add intercom-group x
add to the list any phone number you would like.
Each phone in the list will get a DC number.
This number will be used to call it.
Assign the intercom group to a phone key
Use sat
Change station xxxxx
Add a key of type auto-icom and assign to it the intercom group number and the DC number
Repeat above step for the other phone.
Specify if intercom call will be answered automatically by changing station config page 2
Auto Answer=icom
Press the Do Not Disturb button or the Send All Calls button on your telephone when you
don’t want someone in your intercom group to listen in on a call. Auto Answer ICOM does
not work when the Do Not Disturb button or the Send All Calls button is pressed on the
telephone.
You can modify the intercom behavior on system features page 19
Call Pickup on Intercom Calls? y
Notes
An extension with Data Privacy or Data Restriction active cannot originate an intercom
call. The user receives an intercept tone.
Group-page
This option will dial many phones simultaneity and auto answer on all of them.
Up to 32 groups
Up to 32 members in each group
add group-page x
assign a number and a name to the group.
The number should be in the dial plan EXT range.
Add the extensions to call when this number is dialed.
To use – simply dial the group-page extension number.
Answer Group
An answer group contains up to 20 sip phones or 80 non SIP phones who act as a coverage point for
another user.
For example, if several secretaries are responsible for answering a department’s redirected
calls, all the secretaries could be assigned to an answer group.
The answer group is assigned a group number, and that group number appears in the department’s
coverage path.
All telephones in an answer group ring (alert) simultaneously.
Any member of the group can answer the call
Add coverage answer-group x
A station cannot be directed
Configuration steps needed:
1. Add the coverage answer group
2. Add coverage path
3. Add virtual station – this will be number called
Coverage Remote
Type: change coverage remote 1
This brings you to the remote call coverage table.
1. In one of the blank fields of the table insert the phone number that you wish to be
called externally.
Take note of the field number that you place the external extension. For example.
You choose field 1.
2. Next add a coverage path.
Type: add coverage path <#>
The number can be any number that is free.
At the bottom of the coverage path page you get to see a section called coverage
points. This is where that field number of the 1 that you placed in the coverage remote
table comes into hand.
Place a r and the number <r1> of the remote number in the coverage remote table in
point 1 of the coverage points.
3. Now do a change station <station extension>.
On the change station page, In the coverage path 1 add the coverage path that you
created.
Now make a call to the extension you added the coverage path and watch the call be
automatically routed to the external number you put in the display coverage remote table.
Pick up group
Create a pick up group
Use sat
add pickup-group x
Assign a name to the group
Add any extension to the group
Assign a pickup key to the phone
Use sat
Change station xxxx
Add key of type call-pkup.
Use key number 5 to see the feature in the main phone screen
You can modify the pickup behavior on system features page 19:
Call Pickup on Intercom Calls? y
Call Pickup Alerting? n
Temporary Bridged Appearance on Call Pickup? y
Directed Call Pickup? n
Extended Group Call Pickup: none
Enhanced Call Pickup Alerting? N
Virtual station
Virtual station is used as a phantom phone or CDP\CDN
The station can be forward to a coverage path
Hide caller CLID/ block outgoing CLID
Define a code for the feature as below:
To use simoply dial the feature code followed by the number you are calling E.G
*4390508503129
Bridge Appearance
Bridge appearance allow the same number to dial at more than one station
It can be used for Boss\Secretary
Bridge appearance will not forward. E.G if 3188 Ext is showing on station 3307 and 3307 is
forwarding the call or forward all calls 3188 will not forward.
To forward bridge appearance key use feature access code “Call Forwarding Activation Busy/DA ALL”
Press the bridge appearance key + forward all code+ extension to forward to.
You should hear a special confirmation tone. (BIP, BIP, BIP and silence)
Use sat
change station xxxx
assign key of type brdg-appr
B filed is the key number to bridge
E field is the extension number to bride
If the phone has more than one appearance then you might need to bridge all of them.
You can modify the bridged number behavior in station configuration page 2
Per Button Ring Control? n – Yes will allow to define ringing options per line:
r=Ringing – default
a=abbreviated ringing – short ring
d=delayed ring with x seconds
n=no ring
Bridged Idle Line Preference? Y=if you want the user to lift the telephone receiver, then
press the lighted bridged appearance button to connect to the call.
Bridged Call Alerting? N=Flash,y=Ring
To allow transfer and conference to Bridged calls change station setting as below H323 only:
Bridged Call Alerting? y
Restrict Last Appearance? N
Conf/Trans on Primary Appearance? y
To allow transfer from SIP station must add at least 2 Bridge appearances:
Data privacy for bridge appearance
To block entering into existing call on bridge appearance configure as below
Example
Station A has bridge appearance for station B
Station B answer call and station A will not be able to enter into the call
How it works
To enable this feature you will need to configure for the stations using bridge appearance a special
key called exclusion.
Once a phone answer a bridged line the key will lit automatically
The other phone with same number will not be able to bridge into the call.
If the user who answered the call will want to allow bridging into the call he can press the key
exclusion.
Once the key is pressed the call can be bridged into.
Under system parameters features modify below to yes
And
For each COS enable the feature so exclusion will go on automatically.
Under the station configuration verify Data Restriction =n
If you set this to yes it will also disable the option to bridge into a call BUT also music on hold will not
be allowed for that station or any other system tones
For each station add the key “exclusion”
Busy Indication (BI)
The CM busy indicator (BI) feature will allow to monitor EXT status and call that EXT.
Monitoring: has two states: Lit means the other party is off-hook (busy), dark means the
other party is on-hook (idle).
You can press the button to call the other party, like an abbreviated dial (AD) button.
You may use it while transferring calls on H323 phones
Enhancement to BI feature a little bit in 8.0.1 for Special App SA9138.
To enable simply add a key Busy indication and the EXT number you want to monitor.
To make the button viewable on the main phone screen add it via SMGR and mark it as
favorite
Share Control
If using both soft client and hard phone and want to be able to pick up an existing call make sure that
on ASM configuration below is enabled:
Also make sure that the SIP trunk group between CM and ASM is configured as private type:
Also:
EXT must be configured in AAR table as UNKU
Config server MUST be ASM in 46xxsettings file
Ring again
For ring again on busy extension configure key of type:
auto-cback
additional parameter that might affect:
Call forward
Cos configuration must be set properly to enable call forward
Call forward all calls – set to yes to enable
Restrict call Fwd-Off Net – set to no to allow call forward off the CM
Call forwarding Busy/Da – set to yes to enable call forward on busy or no answer.
Console permissions - set to yes to allow the user to call forward any station. DO NOT use.
Assign that cos to the user station.
Note that phone IP-network-region MUST be configured with the phone domain or feature
activation will fail
Feature access code
Configure the codes to use
This feature should work from Windows communicator or SIP desk phone
To use simply dial the feature access code and use the DTMP keypad to enter the destination.
No need to press # key at the end. If the number is valid dial plan number you should here 3 beeps.
System Parameters coverage forwarding
Set the number of ring before forwarding on busy or no answer.
Notes
To Call Forward off net to ARS or AAR, the AAR\ARS MUST be configured as feature access code and
not directly to the AAR\ARS table.
Call forward will forward the calls from out of the CM if the EXT is configured in the uniform dial plan
to goto CS1K first for dual forking.
Call forward of net. (External call forward)
Notes
To Call Forward off net to ARS or AAR, the AAR\ARS MUST be configured as feature access code and
not directly to the AAR\ARS table.
Cos configuration – must set the Restrict call Fwd-off Net to N
Tenant configuration
Check the station TN – most likely it will be 1
Go to tenant 1 and verify that the COS used also have the Restrict call Fwd-off Net to N
Second call forward
This is system parameter that will allow call forward to forward again if the target is also forwarding.
System parameters features page 16
Chained Call Forwarding = y will allow second call forward
Call forward for off PBX phone from remote recognized phone
If a phone has EC500 configured it can dial a phone number configured in the off PBX feature table.
If the phone calling that number is not recognized in the CM the call will fail.
In the above example a call from EC500 to that number will be answered by CM.
CM will play constant dial tone.
Enter the destination EXT.
A confirmation tone should be heard.
Call Forward Enhanced
Key configured as cfwd-enh
It will allow the user to define a different forward number to:
Internal
External
Internal busy
External busy
Internal no answer
External no answer
Call display key
call-disp key will show call information like call time
Send All Calls
This feature will not ring the station, it will forward all calls to the configured extension in
the key or if not configured it will forward to the coverage path
send-calls
Must have a coverage path configured for that station for the key to work
COS of station must have “Console permissions ”=n
Transfer to Voice
This feature enables the user to transfer an existing call directly to the voice mail.
Must define the feature code in:
Change feature-access-codes
Page 4
Transfer to Voice Mail Access Code: *99
Voice Mail features and call forward
To forward calls to voice mail need to create:
1. Hunt group
2. Create coverage path to use this hunt group
3. Use this coverage path on the extension
Coverage path to Call Pilot Voice mail
Create coverage path remote
Change coverage remote 1
Add you call pilot number here
Create coverage path to use the remote call pilot number
add coverage path 1
Under point one add your remote coverage
Add the coverage path to the stations
Speed Call List and system call list
Step 1 – Verify feature access codes are configured for the lists:
use sat
change feature-access-codes
Abbreviated Dialing List1 Access Code: *10
Abbreviated Dialing List2 Access Code: *11
Abbreviated Dialing List3 Access Code: *12
Abbreviated Dial - Prgm Group List Access Code: *13
For SIP phones there is no button to add. It must use the FAC
For H323 you can add a button of type: abrv-dial and assign the list number
Step 2 - Add a list.
3 list types available
System call list
Group call list
Enhanced call list
use sat
add abbreviated-dialing system
add abbreviated-dialing group x
add abbreviated-dialing enhanced x
change the list size assign numbers to the raws and save.
Step 3 – assign the list to the station
use sat
change station xxxxx
page 4
ABBREVIATED DIALING
List1: system List2: List3:
Add on Module
Very simple to add
use sat
change station xxxx
Note for J169 add on module and general buttons on the J159, J169 and J179
SMGR allows to mark only 9 buttons as favorite, so ADL keys configured on the expansion module
are not actually showing.
Try below 46XX settings parameter:
SMGR_AUTO_FAVORITE 1
Specifies whether all the features and supported autodials configured through System Manager are available on
Avaya J159 IP Phone and Avaya J169/J179 IP Phone irrespective of adding it to favorite or not.
Value operation:
• 0: Do not auto-favorite
1: Auto-favorite
TFD – time force disconnect – trunk disconnect timer
The Outgoing Trunk Disconnect Timer (minutes) field on the Class of Restriction screen
provides the capability to disconnect an outgoing trunk automatically after an administrable
amount of time. This field defaults to blank (outgoing trunk calls are only disconnected when
dropped by one or all parties), or you can enter a timer value in number of minutes to apply to
outgoing trunk calls if the initiating party belongs to this COR.
Time in minutes 2-999
Speed Dial (ADL Key)
Assign a key of type autodial to the station
To modify a key content on SIP Phone :
1. press Features
2. Press on SysNum
3. Select the key you would like to modify and press “Edit”
To Modify key content for H323 phone
Use the utility server as below
First time setup by admin:
In Browser, go to https://<Utility Server IP Addr>:8443/MyPhoneAdmin
Login with the user/pwd used in ssh (in our lab admin/admin01)
Go to MyPhone AES, CM and SES Access
Login and Password are for the CM login and they have to be configured in CM before you change
them here (in our lab admin/admin01).
Note that the user must be in the format <CM user>@<CM IP add>
Save and then Test to make sure the connection is established.
MyPhone application will need to be restarted for the new values to be effective.
End User Usage
https://<Utility Server IP Addr>:8443/MyPhone/MyPhone
Login with the extension and Security code configured for it and select Features
In order to configure the number for the autodial button, press the appropriate Change Button <no>
Phone – time of day per location/group
If CM is serving a few time zones you might need to setup locations and groups for the phones.
For H.323 phones the time of day is taken from location settings in the CM.
The offset is relative to the CM time of day and not GMT, so if CM is in Israel and on time zone IST
and the phone is in USA Los Angeles use -10:00 on the offset to add 10 hours
For SIP phones the time of day is taken from the 46xxsettings file and the ASM clock.
The SIP phone group number should be configured manually on the phone menu.
Another option is to create in the utility server a few folders and for each folder create a different
46xxsettings file.
On the DHCP server option 242 use HTTPSRVR=http://149.49.103.86/USA
Dialing rules for 96x1 and H175 phones
The dialing rules for these phones come from the 46xxsettings file.
By default it will add 9 and 1 to external numbers.
You MUST modify the 46xxsettings file as below to cancel\change this
Split phone screen option
H323 Phones
This only works for H323 phones:
SET PHNSCRCOLUMNS 1
SIP Phones
For 96x1 you may set this in the CM template->Profile Settings->Phone Screen->Half
For J169 it should be same using the CM template, SMGR configuration but currently SMGR RLS
8.0.1 there is a bug and the option is not showing.
Manually Configure the phone menu to use split screen
9608 Phone Menu->Options & Settings->Screen & Sound Options->Phone Screen Width-> select Half
9641 Phone Menu->Options & Settings->Screen & Sound Options->Show Quick Touch Panel-> Select
2
Assign the key you want to see on the main screen as favorite
SMGR->User Management->Manage Users-><Select your user>->Communication Profile->Endpoint
Editor
Once this is done and phone will load new configuration, (Might take a few minutes or use “reload
config”) the phone screen should look like that:
9608
9641
