Bitswap

CallManager & UnityExpress integration

Andy — Thu, 13 Aug 2009 01:44:36 +0000

Configuring CUCM for CUP cheats

Andy — Sun, 26 Jul 2009 01:35:54 +0000

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

1. Create NON-Secure Trunk profile
2. Create SIP trunk to CUP
3. Go to Service Parameter Configuration – CUP PUBLISH TRUNK (change)
4. Go to Serviceability and check
o Cisco AXL Web Service (Copy users from CM to CUP)
o Cisco DirSync (for LDAP sync)
o Cisco CTIManager (Phone control / CUPC )
5. CTI Gateway (user/group/device)
6. Go to User Management -> Application Users
7. Add CTI user account, Select ip phones (all ?). Save
8. Go to User Management -> User Group , select “Standard CTI Enabled”
and add (Add App Users to Group) user we just created in 7.
9. Repeat 8 for group “Standard CTI Allow Control of All Devices”

Presence configuration

1. Finish post-install. Hostname = cm1, ip address = 10.128.2.100
2. Enter AXl user/password (you may use CCMAdministrator/application username)
3. Security password (in my case ccievoice , not cisco , set during installation)
4. success ! -> go to topology !!! cool
5. Go to CUP Serviceability , Tools -> Service Activation
o Cisco AXL Web Service
o cisco UP Sip Proxy
o Cisco UP Presence Engine
o Cisco UP Sync Agent
6. Go to Administration -> Service Parameters -> Cisco UP SiP Proxy
7. Modify ProxyDomain from PROXY_DOMAIN_NOT_SET to your domain (in my case cup.lab.voip)
8. Transport preffered order to TCP-UDP-TLS
9. Go to Precense->Routing->Proxy Configuration and Restart
10. Go to Application Deskphone control Settings (CM and client app (click to dial))
11. Change to ON , put same app name as you created in CM and password
12. Add CUCM address (Deskphone Control Settings)
13. Go to Application->Deskphone Control->User Assignement (grant rights to use this feature)
14. Find->Select All users-> Assign Selected Users (check Enable Deskphone Control!) -> Save
15. Check for mark on Assigned Deskphone control.
16. Go to Presence->Settings and check Enable IM(cluster-wide)
17. Go to Presence->Settings and check CUCM SIP Publish Trunk

Presence Applications

Note:
Before you can synchronize the LDAP directory, you must activate the Cisco DirSync service
Note:
Need to enable LDAP authentication
( LDAP User Search Base cn=Users,dc=voip,dc=lab ; LDAP Manager DN CCMDirSync )

Presence Applications configuration || Configuring CUPC in Presence.

1. On presence server go to Application->IP Phone messenger->Settings (enables messaging from phone)
check PhoneMessenger uername/password
2. Go to Presence->Gateways ADDNEW (cm1) (should we add subscriber as well ??? check docs)
3. Go to Application->CUPC->LDAP Server (used of ldap auth of cupc users during login) and add LDAP server , than
4. Go to LDAP Profile , Application->CUPC->LDAP Profile
Bind DN : cn=CCMDirSync,cn=Users,dc=voip,dc=lab
Search Context : cn=Users,dc=voip,dc=lab
5. Now lets configure CUPC Settings (under Application)
Enter TFTP addresses of publisher (and subscriber?) servers , check Microsoft AD is selected and save
6. Heh… Go to Application -> CUPC -> CTI Gateway Server,
select servers , and , in case u used hostnames in #5 change it to ip address.
7. Lets verify CTI Gateway profile
8. Lets verify Applications->UPC Settings and check what Proxy Listener set to Default Cisco SIP Proxy TCP Listener.
9. Lets ADD VM server. Applications->CUPC->VM Server:
Server Type Unity Connection
ip
web service port 80
protocol http
10. Lets configrure VM Profile
VM Pilot – Default
select Primary VM Server

Configureing CUPC in CUCM

1. Add an new device
2. Add a DN
3. Associate end user with CUPC
4. Set settings of CUPC

Explanation:
1. Add a new device CUPC, Name according to UPC + username , all capital letters (Tools/Control Center)->UP Intercluster Sync Agent should be running (by default it does)
6. Application -> CUPC -> User Settings and assign VM Profile, Conf Profile , CTI GW profile and LDAP Profile to users.
(in my case to bknight and tbriscoe)

Configure Proxy Server Settings
1. ACL
2. Transport Listener
3. Cisco UP services
4. Proxy server settings and restart
5. System troubleshooter

Explanation:
1. System->Security-> Incoming ACL : Description Allow All , Address Pattern ALL (exact,strict)
2. Configure Application listener : System -> App Listener
Select Default Cisco Presence Enginge TCP Listener by default port-5070/tcp (ok , default)
3. next dumb check : Go to Presence -> Routing -> Settings and check On/ and SET
preffered Proxy Listener to Default Cisco SiP proxy TCP Listener.
4. Look on message from #3 and RESTART!

Prerequisites for Presence Service on CME

Andy — Sun, 17 May 2009 23:19:29 +0000

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.

911 terms

Andy — Sun, 17 May 2009 22:01:04 +0000

PSAP public safety answering point
ERL emergency response location
ELIN emergency location identification number
ALI automatic location identification

Restrictions for Music on Hold

Andy — Sun, 17 May 2009 21:50:13 +0000

•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

Andy — Sun, 17 May 2009 20:41:08 +0000

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

Andy — Sun, 17 May 2009 17:37:56 +0000

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

Andy — Sat, 16 May 2009 17:53:14 +0000

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

Andy — Sat, 16 May 2009 17:50:38 +0000

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

Andy — Thu, 07 May 2009 00:46:00 +0000

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.