CallManager & UnityExpress integration

CallManager & UnityExpress integration

Configuring CUCM for CUP cheats

configuring Unified CM for Presence

1. Go to System -> Application Server and add CUP
2. Go to System -> Service Parameter
-> Cluserwide Parameters (System – Presence )
and change Default Inter-Presence Group Subscription to Allow Subscription
3. Configure End Users and Application Users
*) LDAP
4. Go to User management -> End Users
and check Telephone Number and Device Assotiation
verify Primary Extention
verify Permition (Standard CCM End Users, Standard CTI Enabled)
verify Allow Control Device from CTI

5. Go to Licencing-> Capapabilites Assignement and enable CUP & CUPC ( communnicator) for this particular users

6. For Cisco IP Phone messanger app go to
User Managerment -> Application Users and add new user PhoneMessenger
*) Cisco IP Phone Messenger enables your Cisco Unified IP phone to receive, send, and reply to instant messages

7 Test. Go to User Managermet -> User Group Configuration

Prerequisites for Presence Service on CME

Prerequisites for Presence Service

•Cisco Unified CME 4.1 or a later version.

•Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE require firmware load 8.2(1) or a later version.’

This basically means we cannot use presence on 79{46}0

via Cisco Unified Communications Manager Express System Administrator Guide – Configuring Presence Service  [Cisco Unified Communications Manager Express] – Cisco Systems.

Restrictions for Music on Hold

•IP phones do not support multicast at 224.x.x.x addresses.

•Cisco Unified CME 3.3 and earlier versions do not support MOH for local Cisco Unified CME phones that are on hold with other Cisco Unified CME phones; these parties hear a periodic repeating tone instead.

•Cisco Unified CME 4.0 and later versions support MOH for internal calls only if the multicast moh command is used to enable the flow of packets to the subnet on which the phones are located.

•Internal extensions that are connected through a Cisco VG224 Analog Voice Gateway or through a WAN (remote extensions) do not hear MOH on internal calls.

•Multicast MOH is not supported on a phone if the phone is configured with the mtp command or the paging-dn command with the unicast keyword.

via Cisco Unified Communications Manager Express System Administrator Guide – Configuring Music on Hold  [Cisco Unified Communications Manager Express] – Cisco Systems.

Difference Between a Line and a Button

Note that a line is similar to, but not exactly the same as, a button on the phone. A line represents a phone’s capability to make a call connection, so each button that can make a call connection becomes a line. (For example, unoccupied buttons or speed-dial buttons are not lines.) Note also that a line is not the same as an ephone-dn. A button with overlaid ephone-dns is only one line, regardless of whether it has several ephone-dns (extension numbers) associated with it. In most cases an ephone’s line numbers do match its button numbers, but in a few cases they do not.

via Cisco Unified Communications Manager Express System Administrator Guide – Configuring Headset Auto-Answer  [Cisco Unified Communications Manager Express] – Cisco Systems.

CME conferencing and transcoding working example

sccp local GigabitEthernet0/0.230
sccp ccm 10.16.255.2 identifier 1 priority 1 version 3.1 4.0
sccp ccm 10.16.255.3 identifier 2 priority 2 version 3.1 4.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0.230
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register cnfvts01jp
associate profile 1 register mtpvts01jp
keepalive retries 10
switchover method immediate
switchback method immediate
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g722-64
codec g723r53
codec g723r63
codec g729br8
codec g729r8
codec gsmamr-nb
codec ilbc
codec pass-through
maximum sessions 2
associate application SCCP
no shut
!
dspfarm profile 2 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
codec g722-64
maximum sessions 1
associate application SCCP
no shut

dspfarm profile 2 conference
shut
no dspfarm profile 2 conference
dspfarm profile 1 transcode
shut
no dspfarm profile 1 transcode
no sccp ccm group 1
no sccp ccm 10.16.255.2 identifier 1 priority 1 version 3.1
no sccp ccm 10.16.255.3 identifier 2 priority 2 version 3.1

B2BUA

What is B2BUA ?

Cisco Unified CME 3.4 an d later versions acts as both UA server and UA client; that is, as a B2BUA

ephone-dn s and dial-peers

Cisco Unified CME automatically creates one POTS dial peer for each ephone-dn when it is assigned a primary number. If the ephone-dn is assigned a secondary number, it creates a second POTS dial peer. If the dialplan-pattern command is used to expand the primary and secondary numbers for ephone-dns, it creates two more dial peers, resulting in the creation of the following four dial peers for the ephone-dn:

•A POTS dial peer for the primary number

•A POTS dial peer for the secondary number

•A POTS dial peer for the primary number as expanded by the dialplan-pattern command

•A POTS dial peer for the secondary number as expanded by the dialplan-pattern command

Call forwarding is normally applied to all dial peers created for an ephone-dn. Selective call forwarding allows you to apply call forwarding for busy or no-answer calls only for the dial peers you have specified, based on the called number that was used to route the call to the ephone-dn.

For example, the following commands set up a single ephone-dn (ephone-dn 5) with four dial peers:

telephony-service

dialplan-pattern 1 40855501.. extension-length 4 extension-pattern 50..

ephone-dn 5

number 5066 secondary 5067

via Cisco Unified Communications Manager Express System Administrator Guide – Configuring Call Transfer and Forwarding  [Cisco Unified Communications Manager Express] – Cisco Systems.

Configuration for IP-to-IP Gateway

Example 8-12 Configuration for IP-to-IP Gateway

voice service voip

allow-connections h323 to h323

supplementary-service h450.2

supplementary-service h450.3

supplementary-service h450.12

h323

emptycapability

h225 id-passthru

h225 connect-passthru

h245 passthru tcsnonstd-passthru

interface Loopback0

ip address 2.1.1.1 255.255.255.0

h323-gateway voip interface

h323-gateway voip id IPIPGW-zone ipaddr 1.1.1.1 1719

h323-gateway voip h323-id IPIPGW

h323-gateway voip bind srcaddr 2.1.1.1

dial-peer voice 1 voip /* To Unified CM endpoints */

destination-pattern 4…

session target ras

dtmf-relay h245-alphanumeric

codec g729r8

no vad

dial-peer voice 1 voip /* To Unified CME endpoints */

destination-pattern 3….

session target ras

dtmf-relay h245-alphanumeric

codec g729r8

no vad

via Cisco Unified Communications SRND Based on Cisco Unified Communications Manager 7.x – Call Processing  [Cisco Unified Communications Manager (CallManager)] – Cisco Systems.

WTF is OOD Refer ?

According go SIP System Administration Guide:

Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application. The application using OOD-R triggers a call setup request that specifies the Referee address in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to communicate with Cisco Unified SRST is independent of the end-user device protocol, which can be H.323, POTS, SCCP, or SIP. Click-to-dial is an example of an application that can be created using OOD-R.

A click-to-dial application enables users to combine multiple steps into one click for a call setup. For example, a user can click a web-based directory application from his or her PC to look up a telephone number, off-hook the desktop phone, and dial the called number. The application initiates the call setup without the user having to out-dial from his or her own phone. The directory application sends a REFER message to Cisco Unified SRST, which sets up the call between both parties based on this REFER.

Figure 3 shows an example of OOD-R being used by a click-to-dial application. In this scenario, the following events occur (refer to the event numbers in the illustration):

1. Remote user clicks to dial.

2. Application sends out-of-dialog REFER to Cisco Unified SRST.

3. Cisco Unified SRST 1 connects to SIP phone 1 (Referee).

4. Cisco Unified SRST 1 sends INVITE to SRST 2.

5. Cisco Unified SRST 2 sends INVITE to SIP phone 2 (Refer-Target) and the call is accepted.

6. Voice path is created between the two SIP phones.

Note The connectivity to Cisco Unified Communications Manager has been lost and, therefore, IP phone 1 and IP phone 2 have registered to SRST routers.

Figure 3 Click-to-Dial Application using Out-of-Dialog REFER

via Cisco Unified SIP SRST System Administrator Guide – 4.1 Features  [Cisco Unified Survivable Remote Site Telephony] – Cisco Systems.