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)
SRTP
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
MCS
7800
- 7816
- 7825
- 7828 (Only one that supports Business SErver Edition)
- 7835
- 7845
Min HW Reqs
- 2GHz
- 2GB RAM
- 72GB hard disk
- other CM functions to split out
- CTI Mgr
- MOH Svr
- TFTP Server
- SW Conferencing
3rd party servers
- HP (not any more)
- IBM
- VMware vSphere (ESXi 4.0)
- Supported in 7.1(3)+
Cisco Unified Communication OS
- Redhat/Centos based)
- 1000 or less phones, can run DHCP server on CUCM
- CSA
- no root access
- basic changes can be made through CLI/GUI
- Can put time and date in for expiring temporary access
- Informix
Dynamic System (IDS)
Database
Dynamic information can be updated at subscriber if publisher down
(user - facing features)
Everything else relies on publisher
- CCMAdmin
- CCMUser
- BAT
- TAPS
- AXL
- AXIS-SOAP (sometimes) - enables and disables services
- CCM
- LDAP Sync
- License
User-Facing features
CFA
MWI
Privacy enable diable
DND
Cisco ExtensionMobility
Hut group
Device Mobility
CTI CAPF - Certificate Auth Proxy Functions)
Database Access Control
Subscriber allowed control through iptables
Licensing
- CUWL
- DLUs till used, but you get multi-platform/funcationality licences -
if you are going to use multi-application/platform - consider using -
low number of demo licenses included
- Standard User - ~$250
- Professional User - ~$500 - sw conf - buy at
least a few to get sw / demo licenses
- Remember to increase number of servers (from future
servers added)
- UCL
- DLUs - Device License Units - basic unit
- Software Licenses
- Node Licenses
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
Commands
utils service list page
CUCM Deployment
- Single-site deployment
- Multisite WANw/ centralizedcall proc
- Multisite WANwith distributedcall proc
- clustering overIP WAN
- Deployment on Virt Svrs
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
- Local CM Clusters or CME setups
- GKs for scalability
- Transparent use of PSTTN if WAN is unavailable
- Single WAN codec recommended
- Can use TEHO
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
8CMspercluster
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
- Network Settings - NTP Servers, DHCP services, remove DNS
reliance
- Activate necessary feature services and check network
services
- Enterprise Params - mod enterprise params as requires
- Service Params - mod svc params as required
NTP Servers
Need them for deployment...for pub, sub, phone time...
DHCP services
- Cisco recommends up to 1000 phones can run DHCP
off of CUCM (some people have run more than that)
- If you don't, make sure to set option 150 to be your TFTP
server
- Activate DHCP Monitor Service
- Add andconfig DHCP server
- Config DHCP subnets
Remove DNS reliance
e.g. decreases time to get dial-tone
- put in numeric for server name
- put in numeric for phone URL applications in service
parameters
Enable Network and Feature Services
- Tools - Service activation, THEN
- Make sure they're running Control Center Network Services
or Feature Services
System / Enterprise Parameters
If it's not here, it's under service parameters...
- Cluster ID
- AUtoregistration Phone Protool
- Enable Dependcy Records
- CCMUser Parameters (used to display or hide certain
user-configurable settings from CCMUser web page)
- Phone URL Parameters (applications from phoen - update to
IP addresses instead of hostnames)
- User Search Limit - max # of users retrieved from a search
in corp dir feature on phone
System / Enterprise Phone Configuration
- set Cisco Camera Enabled (?) - Overide Common Settings
System / Service Parameters
- T302 timer (interdigit timeout) 15sec default
- enable CDRs (CDR Enabled FLag)
- Define Cisco Extension Mobiilty max login time (under
Extension Mobility area of Service Parameters)
- Define voice media streraming app codecs
- Station Keepalive Interval (how often are keepalive
messages are sent) 30 sec default
- Automate Alternate Routing Enable (use alternate routing if
not enough bw)
- Change B-Channel Maintenance Status - change individual
B-chan maintstatus for PRI and channelassociated
User Account options
End users
- individual person
- interactive logins
- features and adminlogins
- included in user directory
- can be provisioned and authenicated using an external
directory service
Data associated with End users
- User names et. al.
- PIN / SIP digiest crednetials (always
- User privileges - assign roles to groups to users
- Associated PCs, controlled devices, dir #s
- app/feature params (Cisco Extension, Mobility
profiile, Presence Group, Mobility, Certificate Auth Proxy Function
(CAPF) - may be needed for SSL authentication features on phone etc,
...)
Application users
- associated with app
- noninteractive logins
- used for app authorization
- not included in user directory
- cannot use LDAP
LDAP Note - Think about using UserPrincipalname
(username@domain.com) 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
BAT/BPS
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
Use BAT
- Cfg BAT user template
- create csv data input file - Cisco provides default...
- upload csv data input file
- start BAT job
- verify status of bat job
LDAP Characteristics
User lookups
user auth
user provisioning (db sync)
look at cisco.com 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
- power
- load locally stored image
- if no local voice vlan, send Cisco Discovery Protocol for
VoIP VLAN query
- if cisco switch has voice vlan, it will send CDP frame with
voice vlan ID
- If DHCP enabled, reqs IP addr and TFTP server, otherwise
static IP cfg
- connects to TFTPserver, loks for files in the order
- CTLSEP<MAC>.tlv (security environments -
cisco cert trust list)
- SEP<MAC>.cnf.xml
- SIP<MAC>.cnf
- If none ofreq cfg files found, phone reqs def cfgfile
calledXMLDefault.cnf.xml
- phone
compares installed phoneloadversion with loadversion defined within
received cfg file. If laoad diff, request configured load
from
tftp server.
- phone tries to register with CUCM call proc node
- if phone alrady cfged, it will register and SCCP picup cfg
- if localization or customer ringers configured for the
phone, add files downloaded
- If phone not configued
- try autoregistration if enabled.
- if not, phone displays "Registration Rejected"
Cisco SIP IP Phone Startup Proc
- get all of cfg from file, therfore file much larger for SIP
than SCCP
- if local dial rules cfged, rules downloaded
- soem sip phones also download separate softkey cfg file
- CTL file
- SIP<MAC>.cnf.xml
- phone load file
- dial rules
- phone registers
- lcolaizatino files
- softkeys (Type-B only)
- Custom ringers
H.323 endpoints supported
H.323 phones can have multi lines
can be voice/video
normallhy term devices, espvideo
...
not supported
BARGE
Video Advantage, call pickup
unified presence
...
requires fewer cfg steps in CUCM
- setup IP addr and dir #s
- 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
Barge
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
Autoregistration
Consider AutoReg with extension mobility for IPCC agents
Default settings
random DN
Mods needed
BAT
MAC addrs required in BAT files
CUCM Auto-register Phone Tool (TAPS)
CiscoCRS required
complex cfg
Manual Config
mac addrs required
time-consuming
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...
Regions
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
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
SIP or SCCP
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)
Notes
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(?)
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
FILL THIS IN (3-54)
Autoregistration Process
- Req SEP<MAC>.cnf.xml
- FILL THIS IN
Autoregistration check box in CUCM Group
Autoregistration range needs to be set on CUCM
- Verfy autoreg phone protocol
- verify CM gruop enabled
- CUCM enabel / disable cfg / range fr autoregistration
- 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.!#
Gateways
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
CUCM MGCP GW Cfg
- Add MGCP GW in CUCM
- Cfg MGCP GW
- Add Voice mods
- Add VICs to module
- Cfg MGCP endpoints
Methods
- Config Svr
- Manual config (totally config)
- Mixed config removes 'ccm-manager
config' command and re-configures after MGCP (no
ccm-manager config)
- leave 'ccm-manager
config server <ip addr>' so control can
continue to happen
- May want to do this if you have a partial T1/PRI (change
the timeslots in the pri-group
timeslots 1-15 service mgcp)
qsig NOT supported by H.323 or SIP
For MGCP ON GATEWAY ONLY NEED
ccm-manager config server <ip
addr>
ccm-manager config
Everything else get's filled in.
Fractional T1/PRI setup
- In Call Manager, check 'Enable Status Poll' with channels
that you want in maintenance mode under the gateway config
- in System/Service
Parameters you can disable particular channels in Change
B-Channel Maintenance Status
no ccm-manager config
voice-port 0/1/0:15 (or 23)
shutdown
interface serial 0/1/0:15
no isdn
bind-l3 ccm-manager
controller e1 0/1/0
pri-group timeslots 1-31,16 servicemgcp
shutdown
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 10.1.1.101
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
h323
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
sip
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
...
sip-ua
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
DTMF
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...
Wildcards
x |
Single digit |
@ |
NANP |
! |
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.@
BETTER TO USE 9.!
Blocked Pattern
Route Pattern with Block selected (as opposed to route)
Detail
- Allow Device Override - OnNet vs. Offnet
- external dial tone
Translation patterns
have preference over route patternsHunt Groups
Hunt Pilot
- Pilot # to get to Hunt List
Hunt List
- List to get you to one or more line groups
Line Group
Counter
Broadcast/Parallel
Call-Routing Table Targets
- Directory numbers
- Translation pattern
- Route Pattern
- Hunt pilot
- Call Park numbers
- Meet-Me numbers
Basic Partitions to use
- Internal_PT (extensions)
- Emer_PT (911 et.al.)
- TollFree_PT
- Blocked_PT
- <area>Local_PT (e.g. BostonLocal_PT)
- <area>LD_PT (e.g. BostonLD_PT)
- International_PT
- TOD_International_PT
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
- best match wins first
- 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
CMC
- enable CDRs in System/Services
- Call Routing/Client Matter Codes
- check bosx on pattern to require client matterCode
FAC
- Cal
- CHeck Require Forced Auth
Call Coverage
Shared Lines
same DN/line on multipephones
System shows 'shared' automatically
Hunting
Call Routing / Route/Hunt/Menu- Hunt pilot - matches# dialed, dgt mani, last-resort call forwarding,, max hunt timer,
- hunt list
- line group - circular, longest idle, broadcast, or member
- 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),
MOH,
annunciator,
voice termination (always HW)
Audio conf bridge,
MTP
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
MOH
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- MRGL
- Media Resource Group
- Media resources
- Hardware resources
- MOH_
- CFB_
- MTP_
- ...
Round robin approach until exhause resourcesin Media Resource Group
Cfg
- Cfg MRGs
- Cfg MRGLs
- 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
Presence
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
adds:
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
- Customize phone button templates to include presence enablesd speed-dieal buttons
- Apply phone buttons templates to phones
- System / Enterprise Parameters - Enable BLF For Call Lists
- Apply subscribe CSSes to the phones
- System / Security Profile / SIP Trunk Security Profile - Enable
Presence Policies Configuration
- Cfg partitions and CSSes
- assign partiionts to lines and route patterns
- assign subscribe CDSSes to phoens and trunks
implement
- Cfg presence groups
- set
defualt interpresnce group policy - System / Service Parameter (Cisco
Call Manager) - presence subscriptions toward prescence groups for
which no explicit permission has been configured.
- assign presence groups to lines phones and SIP trunks
Mobility
Single Number Reach/Mobile Connect
SNR in CME
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)
- For remote to internal or external....
ToD can be applied
Mobility phone number matching
partial Match in Service parameter 'Matching Caller ID with Remote Destination'
- Add mobility softkey to IP pohone softkey templates
- 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
- Call access #
- Put in ID and PIN
- Put in # to dial out with
Reuquires:
H.323 or SIP Gateway
1 DLU per user that uses it...