CME Additional Features - Part 1

Do Not Disturb (DND)

• This features is used to disable phone ringing (put phone in silent mode). But still calling party info is displayed and the call can be answered.

• DND softkey is used to toggle this feature on/off

• DND state can be toggled during idle and ringing phone states. One exception for ringing state: In case DND was ON in idle state and incoming call reached (phone in ringing state), DND state can't be toggled OFF while ringing.

• Supported for SCCP/SIP phones

• For SCCP phones, turning ON DND in ringing state will apply to the current call ONLY. For SIP phones, turning ON DND in ringing state will apply to the current and future calls. You should explicitly turn it OFF.

• In case CFNA is configured, once DND is turned ON, the call will be forwarded to destination number.

• In case CFA and DND ON applied simultaneously, CFA will take precedence.

Configuration Template

voice register template template-tag

 softkeys idle {[Cfwdall] [DND] [Gpickup] [Newcall] [Pickup] [Redial]}

 softkeys ringIn [Answer] [DND]

!

voice register pool phone-tag

 dnd

 template template-tag

Unlike SIP phones, SCCP phones don't require any configuration for DND. As a workaround to disable DND is to remove the softkey from ringing and idle states.

Note: For internal calls within CME, Ring out DND message will be displayed on calling phone to indicate the called phone has DND ON.

Feature Access Code

We have seen this feature in many topics so far. To make it simple, it enables phone users to dial special codes in order to turn on/off some CME features such CF, Call-Park, Call-Pick, etc.

Its mainly used for analog phones and IP phones without display (no softkeys). However, it can be used as well with IP Phones with display.

There are predefined standard codes and CME admin can change them to create custom codes.

telephony-service

 fac {standard | custom {alias alias-tag custom-fac to existing-fac [extra-digits]} | feature custom-fac}}

Handset Auto-Answer

This feature is used to connect incoming calls directly to lines automatically when handset key is activated. Once handset key is active, a yellow light is lit as indication.

ephone phone-tag

 headset auto-answer line line-number

Intercom

• Intercom is used to provide dedicated path between two phones. Its made of pair of intercom DNs.

• When an intercom button is pressed, a call is speed-dialed to the directory that is the other half of the dedicated pair. The called phone automatically answers the call in speakerphone mode with mute activated. The called phone can deactivate mute button to start two way conversation.

• In case standard DN dials intercom DN, both lines will be connected as intercoms. To prevent those unauthorized phones accessing an intercom-line,  you can assign the intercom DN a number that includes an alphabetic character.  No one can dial the alphabetic character from a normal phone, but the phone at the other end of the intercom can be configured to dial the number that contains the alphabetic character through the CME router. Here is an example.

• Another type of intercom is called Whisper Intercom. Standard intercom allows intercom to idle phones. In case the called phone is busy standard intercom won't be connected. Whisper intercom allows an intercom call to a busy or idle called phone.

• SIP phones have one type of intercom introduced in CME 8.8. In the beginning standard intercom is initiated. In case the called-party is engaged, it will be automatically converted to wisp here intercom

Configuration Template

ephone-dn dn-tag

 number number

 name name

 intercom extension-number [[barge-in [no-mute] | no-auto-answer | no-mute] [label label]] |label label]

!

ephone-dn dn-tag

 number number

 name name

 whisper-intercom [label string | speed-dial number [label string]]

!

ephone phone-tag

 button button-number:dn-tag [[button-number:dn-tag] ...]

!

voice register dn dn-tag

 number number

 auto-answer

 intercom [speed-dial digit-string] [label label-text]

!

 voice register pool pool-tag

  id mac address

  type phone-type

  number tag dn dn-tag

Restrictions

• Whisper intercom can be connected ONLY to a whisper intercom. It can't be connected to standard intercom

• Intercom feature can't be mixed between SIP and SCCP phones

• Intercom functionality can't be used with Shared-Lines

• Intercom DNs can't be dual or octo lines

• Intercom lines support Many-to-One deployment where many source intercom DNs can point to same destination intercom DN. But only one intercom call can be connected at a time.
• Intercom DN can be standard or whisper but not both at same time

Private Lines

Using trunk command under e-phone will enable PLAR for dialing the configured number (this number can be destination on PSTN or VoIP dial-peer). Also, connection plar-opx command can be used under FXO port for same purpose. Those two commands can be as combination or independently.

router#show running-config

voice-port 1/0/0

   connection plar-opx 1082                                             

dial-peer voice 82 pots

   destination-pattern 82                                               

   port 1/0/0

ephone-dn 10

   number 1010

   name manager

ephone-dn 11

   number 1082                                                          

   name private-line

   trunk 82                                                             

ephone 1

   button 1:10 2:11

Paging

It is the process of broadcasting audio to multiple phones. How it works?

A dummy DN will be created for the purpose of paging. This DN won't be assigned to any phone button. However, phones who need to hear broadcast over this DN should participate in it.

The initiator will dial this DN. Automatically all idle phones which participated in this DN will go to off-hook state using speakerphone mode to play the paging audio. Once the initiator disconnects the call, all paging phones will go back to on-hook state.

You can create paging groups which can combine multiple paging DNs. In this case the initiator will dial the group DN number.

Note: IP phones can still be used to make or receive calls during paging. When the phone is used, it simply drops out of the page.

This feature does not have a press-to-answer option like the intercom feature. Its pure one-way audio. The audio stream will be played from initiator to all destinations in either multicast, replicated unicast, or mixture of both (depends on phones support).

SIP phones support whisper paging where a busy phone can hear the paging audio as well (two audio streams combined).

Configuration Template

ephone-dn paging-dn-tag

 number number

 name name

 paging [ip multicast-address port udp-port-number]

!

ephone-dn paging-dn-tag

 number number

 name name

 paging [ip multicast-address port udp-port-number]

 paging group paging-dn-tag,paging-dn-tag[[,paging-dn-tag]...]

!

ephone phone-tag

 paging-dn paging-dn-tag {multicast | unicast}

!

voice register dn dn-tag

 number number

!

voice register pool pool-tag

 id mac address

 type phone-type

 number tag dn dn-tag

 paging-dn paging-dn-tag