CIPT1 Notes

Table of Contents

CCM Cluster

Up to 20 servers per cluster
Up to 8 servers can process calls
Up to 3 in Call Manager Group
1 read/write

use IP addresses instead of hostnames to avoid using dns lookup for phones

Phones need 150ms one way latency (or less) to register 2nd phone

publisher does backup/recovery, api interfaces,

RTP - 16384-65335
- Even ports media
- odd ports RTCP (monitoring/control)
cRTP (Note lower case c)

Endpoint - SCCP / SIP / H.323
3rd party phones supported but cost more money

Media Convergence Server (MCS)
- Can grab IBM /HP servers (not HP anymore)
- or Cisco OEM
- No GUI and CLI access to appliance
- 3rd party access to documented APIs only
Unified Convergence Server (UCS)

Keep Pub separate if possible

SRTP cuts how many clients (7500) that could be supported on MCS

Inter Cluster Communication Signaling (ICCS)

Used to be Redhat based...moved to CentOS


- 7816
- 7825
- 7828 (Only one that supports Business SErver Edition)
- 7835
- 7845

Min HW Reqs
- 2GHz
- 72GB hard disk

- other CM functions to split out
- CTI Mgr
- MOH Svr
- TFTP Server
- SW Conferencing

3rd party servers
- HP (not any more)
- VMware vSphere (ESXi 4.0)
    - Supported in 7.1(3)+

Cisco Unified Communication OS


Dynamic information can be updated at subscriber if publisher down (user - facing features)
Everything else relies on publisher

User-Facing features

Privacy enable diable
Cisco ExtensionMobility
Hut group
Device Mobility
CTI CAPF - Certificate Auth Proxy Functions)

Database Access Control

Subscriber allowed control through iptables


PAK + MAC address to CIsco for license back

4.0 upgrade - make sure to register all phones before upgrade...

Device Packs needed to update devices listed

License Server

Licenses Server - Keeps track of licenses purchased and used
adminstration subsystem - keeps info about licenseunits required for each phone type
- provides license unit calculator

utils service list page

CUCM Deployment

CM Group can contain SRST of the local GW
Make CM Group name intuitive (e.g. CMG_SUB1_PUB_SRST)
Keepalives configured for phone (both SIP and SCCP) - Default 15 seconds
    Configurable under Service Parameters (menu System/Service Parameters)

CME includes hunt groups, presence, extension mobility
SRST can handle up to 1500 phones
CME can handle up to 400 phones

Multisite WAN with distributed call proc

minimum T1 worth of bw at 80ms bandwidth (10000)
BHCA - Busy Hour Call Attempts
Hav to touch CME for all hunt gruop, presence, extension mobility config
Phone Call Plan / number get copied over from phones config from original call manager
    - unplug, and lose dial-plan

G711 mulaw - ~80 kbs with l3 overhead
G729 - ~24 kbs with l3 overhead
G722 - double sample rate
DSP - Digital Signal Processor - e.g. Cisco PVDM2-32 (32 voice terminations of G711 - no dspfarm commands needed)

Alternate Automatic Routing (AAR) - Only used with Location Based CAC in central based call manager model - Kicks in when exceeds CAC
- AAR only works with centralized CAC
- Call Forward Unregistered - CM dials this # when phone unregistered (e.g. WAN down)...
    - Can format with wildcards for bulk administration 9.16031233xxx
    - wildcard is grabbed from dialed digits from user dialing the #

Max 2000 locations
Max 2100 H.323 devices or 1100 MGCP GWs
1200 phones per SRST 3945 Integrated Services Router (ISR)

Tail End Hop Off (TEHO) - go across WAN and call locally even if locally is remote office

Multisite WAN with Distributed Call Proc

Clustering over WAN

80 ms msax round-trip delay delay
min bandwidth 1.544 Mb/sforevery 10000 BHCAs or Subscriber
Up to 8small sites with remote failover deployjment
Failover across wan supported

Deployment on Virt Svrs

3 CMs per group

Redundancy Design

1 to 1 servers (each server is backed up by 1
2-1 servers (2 servers are backed up by 1) - more commonly done in practice...

20 servers / cluster
8 call processing nodes
3 CMs / cluster

Initial CM Setup Checklist

  1. Network Settings - NTP Servers, DHCP services, remove DNS reliance
  2. Activate necessary feature services and check network services
  3. Enterprise Params - mod enterprise params as requires
  4. Service Params - mod svc params as required

NTP Servers

Need them for deployment...for pub, sub, phone time...

DHCP services

  1. Activate DHCP Monitor Service
  2. Add andconfig DHCP server
  3. Config DHCP subnets

Remove DNS reliance

e.g. decreases time to get dial-tone
  1. put in numeric for server name
  2. put in numeric for phone URL applications in service parameters

Enable Network and Feature Services

System / Enterprise Parameters

If it's not here, it's under service parameters...

System / Enterprise Phone Configuration

System / Service Parameters

User Account options

End users

Data associated with End users

Application users

LDAP Note - Think about using UserPrincipalname ( instead of SamAcctName (username))
    May need to point to Global Catalog (TCP 3268) vs LDAP (TCP 389)
Redirect athentication to LDAP/AD svr

User Mgmt Options

1 by 1 manual config
bulk cfg
LDAP integration (synch and/or auth)

LDAP cfg

up to 5 connections by default
ADAM/ Lightweight Directory Services
CUCM End-User Data Location
No LDAP Integration LDAP Sync LDAP Sync and Auth
Personal org settings Local LDAP LDAP
Password Local Local LDAP
CUCM Settings
(including PIN/Digest Credentials)
Groups and Roles
Associated PCs
Controlled Devices
Extension Mobilty Profile and CAPF
Presence Group and Mobility
Local Local Local

Credentail Policy Default (default pw et al)
Credential Policy - failed login timeouts, minimum credential lengthetc...

User/Roles/Privileges (sp)

CAR - CDR Analysis and Reporting

if you turn off phone web pages, some other services stop working


Bulk Administration Tool
- Backup for Backup (e.g. list of phones is list of phones and can restore)
- bulk transaction to DB
- Upload/Download files

Bulk Provisioning Service

Needed for BAT


  1. Cfg BAT user template
  2. create csv data input file - Cisco provides default...
  3. upload csv data input file
  4. start BAT job
  5. verify status of bat job

LDAP Characteristics

User lookups
user auth
user provisioning (db sync)
look at for supported directoris - includes
 - AD 2000 or higher
 - ADAM - Microsoft Active Directory Aplication Mode (proxy AD)
 - iPLanet orSUN ONE LDAPsvrs
 - ....

Single Site On-Net Calling

CIsco SCCP IP Phone Startup Proc

  1. power
  2. load locally stored image
  3. if no local voice vlan, send Cisco Discovery Protocol for VoIP VLAN query
  4. if cisco switch has voice vlan, it will send CDP frame with voice vlan ID
  5. If DHCP enabled, reqs IP addr and TFTP server, otherwise static IP cfg
  6. connects to TFTPserver, loks for files in the order
    1. CTLSEP<MAC>.tlv (security environments - cisco cert trust list)
    2. SEP<MAC>.cnf.xml
    3. SIP<MAC>.cnf
  7. If none ofreq cfg files found, phone reqs def cfgfile calledXMLDefault.cnf.xml
  8. phone compares installed phoneloadversion with loadversion defined within received cfg file.  If laoad diff, request configured load from tftp server.
  9. phone tries to register with CUCM call proc node
  10. if phone alrady cfged, it will register and SCCP picup cfg
  11. if localization or customer ringers configured for the phone, add files downloaded
  12. If phone not configued
    1. try autoregistration if enabled.
    2. if not, phone displays "Registration Rejected"

Cisco SIP IP Phone Startup Proc

  1. CTL file 
  2. SIP<MAC>.cnf.xml
  3. phone load file
  4. dial rules
  5. phone registers
  6. lcolaizatino files
  7. softkeys (Type-B only)
  8. Custom ringers

H.323 endpoints supported

H.323 phones can have multi lines
can be voice/video
normallhy term devices, espvideo
not supported
Video Advantage, call pickup
unified presence
requires fewer cfg steps in CUCM
  1. setup IP addr and dir #s
  2. at phoen, enable call routing toward CUCM by specifying IP address

3rd party phone support SIP

6 DLUs
Cisco phone models can be ordered without license (= on end of phone part)
Features not supported on SIP
    Phone Services
    CUCM Assistant
    CU Video Advantage
    Call Pickup
    CU Presence
old phone type A phone, never intended to use SIP.  Use SCCP if option available.  Use Dial softkey.
Type B phone don't have to

MD5 used for SIP Digest Authentication

Config Methods


Consider AutoReg with extension mobility for IPCC agents
Default settings
random DN
Mods needed


MAC addrs required in BAT files

CUCM Auto-register Phone Tool (TAPS)

CiscoCRS required
complex cfg

Manual Config

mac addrs required

Endpoint Basic Cfg Elements

Phone NTP Reference

Doesn't get used much
SIP 200 OK Registration message will have time stamp
Directed Broadcast, Unicast work, others do not

Date/Time Group

Default group - CMLocal
Put into Device Pool

Device Pool

SRST group

CUCM Group

What call managers registered to...


Codec set here
in codec set lossy (link is questionable) vs Low Loss (reliable)
Be careful to convert mulaw to alaw if necesary
G729 Annex B supports VAD (and CNG) (commonly turn off - messes with fax and modem)


Location are tag for bandwidth to that location
Think about using location for links thus bw is limited per link if necessary

Enterprise Phone cfg

Set phone cfg at enterprise level

double question mark on phone shows codec/packets etc...

Phone Security Profile

Set up Cert (CAPF) settings

Softkey Template

Phone Button Template

All keys on phone (Not soft keys

SIP Profile (SIP phones only)

Can disable early media per SIP Profile

Common Phone profile


Device Defaults

Device list for loads
Device Pack can update list...
If things get hosed and you need to roll back...think about renaming
(e.g. pull down through TFTP, rename, put back up)

XMLDefault.cnf.xml file lists default cfg
79xx with side cars - order power brick and plastic connector ahead of time
793x has ~30 buttons
Additional lines - abbrieviated dialing cheat...

Planning for phone config

Device Pool

NTP Referenes ->Date/Time Group-> Device Pool
Regions -> Device Pool
Locations -> Device Pool & Phone(?)


Device Pool -> Phone
Phone softkey Template -> Phone
Common PHone Profile -> PHone
SIP Profile ((SIP phones only) -> Phone
Enterprise Phone Cfg -> Phone
Phone BUtton Template -> Phone
Device Security Profile -> Phone

Autoregistration Process

  1. Req SEP<MAC>.cnf.xml

Autoregistration check box in CUCM Group
Autoregistration range needs to be set on CUCM
  1. Verfy autoreg phone protocol
  2. verify CM gruop enabled
  3. CUCM enabel / disable cfg / range fr autoregistration
  4. manual cfg or CUCM BAT to personalize autoreg
Can request BAT template from Cisco when order phones...(MAC Addresses pre-populate)
CUCM Auto-Register Phone Tool
    DUmmy MAC addrs in BAT file with phone #...

Get BAT template file bat.xlt file to create starter csv files

PHones can be registered, unregistered (not currently plugged in) or unknown
utility - vomit (wireshark with adaptions for voip) puts together RTP streams...
8.x+ Jabber based client connects through to Clident Services Frmawork (CSF)- CSF is client type on CUCM, need presence - CSF is dummy SIP softphone on CUCM
    CIPC -

Call Routing -> Directory Number

Route Pattern in CUCM
9.! (9 followed by 1 or more digits)  Better than 9.@ probably...
also 9.!#


DTMF relay method
Supplementary services
CUCM redundancy
Call survivability

MGCP controlled FXS/FXO ports can show up in CM as SCCP controlled ports
    T1 CAS, and FXS/FXO ports stay up in MGCP, but now ISDN PRI
    MGCP T1 CAS assumes all 24 channels are voice
H.3232 TCP only
ccm fallback needed on gw for MGCP to support srst
MGCP port 2427
2600, 2700, 3600, 3700 had dsp on network modules later routers can get PVDMs on motherboard

ccm-manager config server <ip addr>
ccm-manager config


  1. Add MGCP GW in CUCM
  2. Cfg MGCP GW
  3. Add Voice mods
  4. Add VICs to module
  5. Cfg MGCP endpoints


qsig NOT supported by H.323 or SIP


ccm-manager config server <ip addr>
ccm-manager config

Everything else get's filled in.

Fractional T1/PRI setup

no ccm-manager config
voice-port 0/1/0:15 (or 23)
interface serial 0/1/0:15
    no isdn bind-l3 ccm-manager
controller e1 0/1/0
    pri-group timeslots 1-31,16 servicemgcp
controller e1 0/1/0
    no pri-group timeslots 1-31
    pri-group timeslots 1-4,16 service mgcp
    no shutdown
interface serial 0/1/0:15 (or 23)
    isdn bind-l3 ccm-manager

H.323 GW Config

Default dial-peer is H.323
to make SIP dial peer
    session protocol SIPv2

Default port TCP 1720
interface FastEthernet 0/0
ip address ipaddr mask
h323-gateway voip interface
h-323-gateway voip bind srcaddr
dial-peer voice 1 voip
destination-pattern 2...
session target ipv4:ipaddr
voic class h3232 1
h225 timeout tcp establish 2
h225 timeout setup 2
dial-peer voice 1 voip
destination-pattern 2...
voice-class h323 1
session target ipv4:<ipaddr>
dial-peer voice 1 voip
preference 2
destination-pattern 2...
voice-class h323 1
session target ipv4:<ipaddr>
voice service voip
no h225 timeout keepalive (no dropped calls when H225/wan connection is lost)
allow-connections h323 to h323
voice class h32321
h225 timeout tcp establish 2
h225 timeout setup 2
call preserve (disable media inactivity detection)
if you forget dial-peer preference you get default 0 and random selection

SIP Trunks

<insert graphic for config from CUCM>
voice service voip
        bind control source-interface FastEthernet0/0
        bind mediasource-interface Fastethernet0/0
        session transport tcp
interface Fastethernet 0/0
    ip address <ip addr> <mask>dial-peer voice - voip
dial-peer voice 1 voip
    session-protocol sipv2
    retry invite 5
    retry response 10
    sip-server ipv4:<ipaddr>
on Call Manager - SIP trunk always allocates an MTP even if a common DTMF method betweeen calling devices is available


SIP Notify for out-of-band
rtp-nte (RFC 2833) for in band

Good debug commands for call control

debug isdn q931
debug voice ccapi inout
    show log | called

Serviceability Tools / Dial Number Analyzer
from gw
show dialplan <number>

Standard Local Route Group

CUCM 7.x+
Set the Device Pool 'Standard Local Route Group' to be a Local GW Route Group
In your route pattern, use 'Standard Local Route Group'.

T302 timer reset to shorten (International or regular) end of dialstring dial Timeout

CUCM System/Service Parameters
search for both regular and h.225 parameters

Route Patterns/Numbering Patterns

longest explicit match wins...


x Single digit
! One or more digits(0-9) (t302 timer will kick in unless # is included in dial pattern
[x-y] generic range notation
[^x-y] exclusion range notation
. Terminates access code
# Terminates interdigit timeout
<wildcard>? matches 0+ of wildcard
<wildcard>+ matches 1+ of wildcard
\+ matches + sign as part of number for e.164 dialing
[^1] (any digit that isn't '1')
\+ in route pattern = '+' (e.g. international dial pattern - 011, 00, or whatever)'
\+!# (! is variable length dial pattern)

CUCM Addressing methods

IP Phone SCCP - Digit-by-digit or En bloc (Type-B phones only)
IP phone SIP - en block or KPML (Type B phoens) or SIP dial rules
Gateway - MGCP/SIP/H323 - overlap and sendingand receiving
trunk - SIP/H323 - En bloc or Overlap sending and receiving

Look up Digit-Signaling Methods 4-95 in lab books

Urgent Priority

Ship it!  (e.g. 911)
Blocked Patterns

Route Filters

Only used with 9.@

Blocked Pattern

Route Pattern with Block selected (as opposed to route)


Translation patterns

have preference over route patterns

Hunt Groups

Hunt Pilot
Hunt List
Line Group


Call-Routing Table Targets

Basic Partitions to use

Line/Device Approach for CoS/Path Selection

Device is path selection (allow everything out local gateway)
LIne is CoS (Local LD) (block specific patterns)

Use Standard Local Route Group to assign to local GW...

CSS goes across all partitions
  1. best match wins first
  2. if equal matches, 1st partition wins (line css first if necessary)

Digit Manipulation

Can do digit manipulation on calling ID
Can do in Translation Pattern, Route Pattern
Can apply on device pools,gateways trunks and so on...
Transformation settins no applicaable when 'Block This Pattern' is selected.

CMC and FAC Implementation

CMC -0 forces user to enter any configured client matter code forbilling andtracking calls made per client
FAC forces user to enter cfged auth code with a sufficient auth level
- both generate CDRs
  1. enable CDRs in System/Services
  2. Call Routing/Client Matter Codes
  3. check bosx on pattern to require client matterCode

  1. Cal
  2. CHeck Require Forced Auth

Call Coverage

Shared Lines

same DN/line on multipephones
System shows 'shared' automatically


Call Routing / Route/Hunt/Menu
  1. Hunt pilot - matches# dialed, dgt mani, last-resort call forwarding,, max hunt timer, 
  2. hunt list
  3. line group - circular, longest idle, broadcast, or member
  4. endpoints - ip phones , voice-mail ports

build from bottom up...

Multiple hunt pilots can point to same list

circuilar, longest

Call Forward No Coverage (last-resort call forwarding) - Need to have another CF filled out...

Media Resources

Transcoders (always HW),
voice termination (always HW)
Audio conf bridge,

All Media resources register with CUCM using SCCP
Audio streams always terminated by media resources (keeps CM or GW in the loop)
Try MTP setting on or off for trunks to see if needed

To create software MTP / Conferencing- enable IP Media Streaming Service
Put in Call Count in Service Paraemeters

BUilt-in Conf Bridge Resource Char invoked by the Barge feature only - G.711 support only
Make sure to turn on Builtin Bridge Enabel in Service Parameters

Meetme - predefined #s
Basic Ad Hoc
Advanced Ad Hoc - Conference Ad Hoc Conferences together

Media Resource Group List (MRGL) configured on Device Pool


unicast or multicast
configurable IP address....
Co-resident or standalone (are other things running on the CM)
Max of 51 unique sources -
#51 is always fixed audio source,buy from Cisco USB
Need public rebroadcasting license to rebroadcasting legally

Media Resources / MOH Audio File Management

remember it's transcoding uploaded
Software Conference Bridge can only support G.711

Media Resource Group List

Assign MRGL on Device Group
  1. MRGL
  2. Media Resource Group
  3. Media resources
    1. Hardware resources
    2. MOH_
    3. CFB_
    4. MTP_
    5. ...
Round robin approach until exhause resourcesin Media Resource Group


  1. Cfg MRGs
  2. Cfg MRGLs
  3. Assign MRGLs to Phones

Cisco IP Phone Services

Newer phones can support better applications
Service has to be subscribed to a phone.  Shows up in Services list
Cisco IP Phone Service Subscriptions

Device/Device Services/Phone Services
redundancy can be load balancer


SIP for Instant Messaging and Presenceleverting Extensions (SIMPLE)
Watcher - watches presence
Subscriber -

Native Presence

Monitor line states, what users are associated with what line
v7 SIP
v8 XPPC port 5222
Speed Dial/BLF willmonitor whether online or not
- speed-dial presence (needs to have BLF)
- call history presence
- presence policy

Cisco Unified Presence

U- User status info
Cisco IP Phone Messenger application
Cisco Unified Personal Communicator (CUPC)
3rd party presence server integration
Need CUPC Server

Subscribe CSSes

What Partitions can be monitored for presence

Presence Group

Enable BLF and Presence
  1. Customize phone button templates to include presence enablesd speed-dieal buttons
  2. Apply phone buttons templates to phones
  3. System / Enterprise Parameters - Enable BLF For Call Lists
  4. Apply subscribe CSSes to the phones
  5. System / Security Profile / SIP Trunk Security Profile - Enable

Presence Policies Configuration

  1. Cfg partitions and CSSes
  2. assign partiionts to lines and route patterns
  3. assign subscribe CDSSes to phoens and trunks

  1. Cfg presence groups
  2. set defualt interpresnce group policy - System / Service Parameter (Cisco Call Manager) - presence subscriptions toward prescence groups for which no explicit permission has been configured.
  3. assign presence groups to lines phones and SIP trunks


Single Number Reach/Mobile Connect

Mobile Connect in CUCM
Mobility Softkey on phone will allow transfer back and forth...
1 more DLU for each remote location

CSS Handling in Mobile Connect

Rerouting CSS of remote destination profile needs access to remote destination(s)
ToD can be applied

Mobility phone number matching

partial Match in Service parameter 'Matching Caller ID with Remote Destination'
  1. Add mobility softkey to IP pohone softkey templates
  2. cfg uend user

Mobile Voice Access

Allows user to call enterprise from any phone and place outgoing calls and appear to be calling from office phone
  1. Call access #
  2. Put in ID and PIN
  3. Put in # to dial out with
H.323 or SIP Gateway
1 DLU per user that uses it...