Cisco UC – CUCM – Communication Manager

By | 03/11/2014

See Cisco Unified Communications for architecture information

 

Infrastructure Setup

System > Servers – List of servers that are part of the cluster: publisher, subscribers, presence,…. Subscribers should be defined here before the installation. It’s recommended to use IPs instead of Names.

System > Cisco Unified DM – Autoregistration parameters.

System > SRST – Definition of the SRST systems

Device > Gateway – Definition and configuration of the MGCP and H.232 Gateways and their PSTN cards.

Device > Trunk – Configuration of trunk ports to other elements of the network.

SIP Trunks – To connect to gateways
Intercluster Trunk Non-Gatekeeper – To connect to other CUCM clusters
H225 Trunk (Gatekeeper) – When there are more devices than CUCM connected to the GK
Intercluster (Gatekeeper) – When only CUCMs connect to the gatekeeper

Device > Gatekeeper – Configure gatekeepers

 

Devices Setup

System > Device Pool – Usually one device pool per site is configured. It defines some of the characteristic that the devices will obtain: Date/Time Group, Locale, Region, Location and CM Group.

System > Date/Time Group – Defines the time zone of the devices.

System > Region information > Regions – Setup of sites relationship to define which codec the sites will be used when devices call between them.

System > Cisco CM Group – Defines to which server the phones will connect and secondary servers in case of failure.

Systems > Location Info > Location – CAC – Call Admission Control – Configuration of the available bandwidth between sites in order to not allow calls when there is no enough bandwidth.

 

Dial Plan

Call routing > Route Hunt > Route Group – Define a group of gateways with the same characteristics. Usually a Route Group only contains one gateway.

Call routing > Route Hunt > Route List – List of Route Groups with their priority.

Call routing > Route Hunt > Route Patterns – Used to define where should be routed a call with a specific pattern in the destination number.

If a gateway is used directly in any Route Pattern, it won’t be shown as available for the Route Lists.

Route Pattern: 0.[6789]XXXXXXXX
Gateway/Route Option: E1@Gateway or Route List
Discard Digits: predot

This configuration will forward all the calls starting with 0 and with 6,7,8 or 9 as a second digit followed by any 8 digits call to the E1 Gateway, and it will remove the characters before the . before sending it to the gateway.

Typical route patterns created for a Spanish dial plan

0.[89]XXXXXXXX // Land Line phones
0.[67]XXXXXXXX // Mobiles
0.112 // Emergency numbers
112 // Emergency numbers
0.118XX // Information numbers
0.00! // International numbers
0.00!# // International numbers
21XX // Internal extensions in a remote site

When using !, CUCM doesn’t know how long the number will be. CUCM will wait some time before processing the call. This time is defined by the variable Interdigit Timeout T302. The default is 15 seconds and it can be modified in Systems > Service Parameters > Cisco Call Manager > T302 Timer.

As an alternative, it’s possible to use the #. With this symbol, the user will tell the system that he has finished adding digits to the number.

Call routing > Translation pattern – It allows to transform a called number for inbound and outbound calls before looking at the route pattern or the DN list.

Incoming calls:

Translation Pattern: 914002XXX
Called Party Transform Mask: 2XXX

This configuration will convert incoming calls to 914002XXX to the short numbers 2XXX. Then CUCM will look for the short number in the DN list.

To allow the redial function, it’s needed to add a 0 prefix for the calling party in the same Translation Pattern.

Outgoing calls:

Translation Pattern: 3XXX
Prefix digits: 049

This configuration will add 049 to the 3XXX extensions resulting in a 0493XXX, then CUCM will look at the route pattern to decide where to forward the outgoing call.

Call routing > AAR Group – Automate Alternative Route. Forward calls discarded by CAC (Location bandiwith limitation) through the PSTN line using AAR .

Dial Plan Techniques

Caller ID Masks

1) In the route pattern – define the full Calling Party Transformation Mask. It will apply to all the calls using that pattern.

2) In the route pattern – with the prefix digits field. It will add the digits to the extension number if they are a valid mask. Only valid if there is a correlation between internal DNs and public numbers.

3) In the Phone > DN > Mask field. It needs to be added for each DN.

When using a PSTN as a failover for internal extension calls, CUCM will need to transform both caller ID and calling ID to the public numbers. This needs to be done in the Route Group configuration for the Group List (not directly in the route group!!)

TEHO – Tail End Hope Off – Is used to choose the GW of a call depending the called party and considering what GW is closer and is cheaper to the destination. This practice is not considered legal in some countries.

EIGRP-SAF – It’s possible to send CUCM Cluster information to other clusters through EIGRP and SAF protocol. This is only useful when there are lots of CUCM clusters in the same EIGRP domain network.

Hunt Pilot

Call Routing > Route/Hunt > Hunt Pilot – Defines a DN number and a Hunt List. Since version 9, it supports queuing.

Call Routing > Route/Hunt > Hunt List – It’s a priorized list of Line Groups.

Call Routing > Route/Hunt > Line Group – It contains the actual extensions that are part of the Hunt Pilot. Here is’t defined the behavior of the incoming calls (broadcast or top-down) and also how the Hunt Pilot should behave when the lines are busy or don’t answer (jump to the other group? Try next line?)

 

Class of Control – Partition / CSS

Call Routing > Class of Control > Partition – Used to classify the different elements in the CUCM so a access layer can be configured using CSS. It also allows the craetion of duplicate route patterns and DNs, if they are in different partitions.

Typical configuration for partitions is:

Partitions for route patterns: PT-Emergencies, PT-National, PT-Mobiles, PT-Internacional, PT-Premium
Partitions for user devices: PT-Internal, PT-Managers

Call Routing > Class of Control > Calling Search Space – A CSS contains partitions and it’s used to define permissions (who can call what numbers)

CSS can be defined in the DN and also in the Phone device. When both are configured, the result is the union of both CSS. The first to be queried is the DN and this is only important when there are blocking partitions configured in the system. A best practice is to always assign the CSS to the phone device.

If a device is trying to contact a number which his partition is in his CSS, the call will be allowed. If the destination partition is not in the CSS, the call will be blocked.

When a destination (or pattern) is duplicated with different partitions, the device will call the destination which one the partition is in his CSS.

Typical confiugration for CSS:

CSS-Internal (PT-Internal, PT-Emergencies)
CSS-National (PT-Internal, PT-Emergencies, PT-National)
CSS-Mobiles (PT-Internal, PT-Emergencies, PT-National, PT-Mobiles)
CSS-International (PT-Internal, PT-Emergencies, PT-National, PT-Mobiles, PT-International)
CSS-Managers (PT-Internal, PT-Emergencies, PT-National, PT-Mobiles, PT-International,PT-Managers)

Employees will be assigned the PT-Internal partition and CSS-International CSS while the managers will have the PT-Managers partition and CSS-Manager CSS. Manager secretaries will have the PT-Internal partition with the CSS-Manager CSS. That way, managers can’t be called directly by employees, they will need to call first to the secretary, but managers will be able to call anyone and call between them.

Local Routing using CSS and Partitions

CSS and Partitions can be used to define which gateway should each of the devies use for calling some numbers, for example, emergency 112. Then a PT-Emergencies-Site1 and PT-Emergencies-Site2 route patterns will need to be created for each of the sites.
CSS will need also to be duplicated so site 1 uses CSS-International-Site1 (that includes PT-Emergencies Site1) and CSS-International-Site2 (including PT-Emergencies-Site2).
This method is not escalable at all and requires to create lots of Partitions and CSS

An alternative to this system is to use Route Local Patterns, available since version 7.

It’s needed to crate a Route list “Local Route List” with the type Standard Local Route Group and then create a Pattern 112 in the PT-Emergencies that points the gateway to the Route List.
It’s supposed that there is a Device Pool for each site and it’s in this Device Pool that should be configured the “Local Route Group” that the devices will use when trying to call the number 112.

User Features

Call Routing > Call Park – It allows to park a call into an extension, so it can be recovered from another phone device calling to that extension.

Call Routing > Call PickUp Group – Pickup groups allow members of the group to capture calls ringing on other phones of the same group. It’s possible to configure audio and visual signals to the people in the pickup group when there is a call and also showing who is calling to who.

When combining Pick Up groups with Hunt Groups, the hunt group extension also needs to be included in the Pickup group.
To avoid having to manually answer the call after the pickup, a “Auto answer” setting can be checked on System Parameters > Auto Answers

DND – Do Not Disturb – User option to put the phone in silent mode.

Call Back – If the called user is busy, the callback funcion sends a message to the caller when the called becaomes available.

End User Website – https://CUCM-IP/ucmuser – End users needs to be created and assgined permissions as Standard End Users. Website provides functionalities that are also accessible through the phone such as dial peers, forwardings, directory,…

Call Routing > Intercom – allows the functionallity to speak with the destination without it needed to answer the call. There will be an additional button in the phone to communicate with the destination and speakier-mute features will be activated in the destination. Useful for secretary-manager scenarios.

Media Resources

The service IP Voice Media Streaming should be activated under Serviciability to activate Media Resources such as Music on Hold and Conferences.

Media Resources > Annunciator – These are the messages that are given to the callers when there is an error.

Media Resources > Media Termination Point – Software MTP that will support singaling transformations (inband – outband) such as DTMF (Dial Tone Multi Frequency). SCCP sends tones out-of-band while SIP needs the tones to be in-band (RFC 2833).

Media Resources > Media resource group –  are used to allocate resources to a device pools (or devices).  It includes Transcoders, Conference Briges, MTPs, MOH servers and annunciator servers.

Media Resources > Media Resource Group List – is a priorized list of Media Resources Groups and it’s the element that will be assigned to a device or a device pool. If no MRGL is assigned to a device pool or there are no resources available, CUCM will try to assign any resource that are not allocated to any MRG. For this reason it’s convenient to leave some resources without allocation.

Music on Hold

Media Resources > Music on Hold Servers – List of servers that can be used to play Music on Hold. Usually publisher and subscribers.

Media Resources > Music on Hold Audio Sources – List of files available for playing MOH. The file details will provide information about the supported codecs.

Music on Hold between sites will play depending on the codec used between sites. By default, MoH is not played when G.729 is in use because this codec is not appropriate to play music. Instead, it will sound the TOH (Tone on Hold). To enable MOH with G729, go to System > Service Parameters, choose the Publisher / IP Voice Media Streaming. Select G729 as allowed coded under Supported MOH Codecs option

Multicast can also be used to save bandwidht on the intersite channels, but the activation of Multicast is not common on the network. As an alternative, it’s possible to configure the audio file to be streamed to the Multicast address from the local gateway in the remote site if it has enabled SRST or Call Manager Express options. Then the MOH file in  CUCM should be enabled for multicasting (Media Resources > Music on Hold Audio Sources), so the remote phones try to locate it in the Multicast IP and Multicast audio sources should be enabled too in Media Resources > Music on Hold Server Configuration.

Conferences

Devices can’t perform conferences by themselves because it would mean lots of RTP streams between them. They need to use a Conference Bridge (CFB) where they will send their stream and receive a unique stream from the CFB with the combined audio of the other participants.

Software conference bridge

  • Uses CPU
  • Cheaper
  • All the participants should access with G711 codec
  • Number of participants depends on the server. 64 participants by conference (it could require a dedicated server for conferences)

Hardware conference bridge

  • Each participant use one DSP channel, that is provided by PVDM hardware.
  • Supports both G729 or G711
  • Expensive
  • PVDM2 only allows 8 participants per conference. PVDM3 can support up to 64 participants per conference.
  • Each participant uses one DSP channel

When Hardware and Software CFBs are available, CUCM will prioritize the software CFBs.

By default CUCM have a limitation of 4 participants per conference. This parameter can be increased in System > Service Parameters > Maximum adhoc Conference.

Transcoders

Transcoders are used to transform an audio stream from one codec to another codec. CUCM doesn’t support transcoding by software.

Software CFB only accept G.7111 calls, so CUCM will send the G729 audio streams to a Transcoder to get them convereted to G711.

PVDM

PVDM are the hardware cards that can be installed in a UC routers (dedicated or GW, CUBE,…) and provide Media Resources services to the CUCM such as Hardware conference bridge, transcoding, voice codification or MTP iOS.

PVDM2 and PVDM3 (newer routers 29XX and 39XX) have two different type of connectors but there is an adaptator to install PVDM2 in PVDM3 slots. It’s not possible to mix PVDM2 and PVDM3 cards in the same router.

Each PVDM has a number of DSPs and each of them allows 16 DSP channels . One DSP can only be allocated to a role although only a few channels are used. The other channels of the DSP won’t be available for other roles.

PVDM2-8 – 1/2 DSP with 8 channels
PVDM2-16 – 1 DSP with 16 channels
PVDM2-32 – 2 DSP with 16 channels each
PVDM2-64 – 4 DSP with 16 channels each

CUCM uses SCCP to register the DSP resources.

PVDM Configuration

@GW

vocie-card 0
 dspfarm
 dsp services dspfarm
sccp local fastEthernet [INTERFACE] //Interface of the IP address that will provide the service
sccp ccm [CUCM-IP] identifier [NUMID] version 7.0+
sccp
dspfarm profile [PROFILEID=1] conference //enable role conference
 codec g711alaw
 codec g722-64
 …
 maximum sessions 1 // ? Will tell how many DSPs are available
 associate application sccp
 no shutdown
dspfarm profile [PROFILEID=2] transcode //enable role transcode
 codec g711alaw
 codec g722-64
 …
 maximum sessions 2 // ? Will tell how many DSPs are available
 associate application sccp
 no shutdown
sscp ccm group 1
 associate ccm [NUMID] priority 1 // Where the first 1 is the sccp ccm identifier)
 associate profile [PROFILEID=1] register [CONFERENCE-NAME] //name that will be used in the CUCM
 associate profile [PROFILEID=2] register [TRANSCODER-NAME] //name that will be used in the CUCM

Troubleshooting:

show dspfarm dsp idle
show sccp
show dspfarm dsp all
show ip rtp connection
show sccp connections

@CUCM

Media Resources > Conference Bridge – Add new CFB type Cisco IOS Enhanced Conference Bridge with exactly the same name configured in the GW [CONFERENCE-NAME].

Media resources > Transcoder – Add new Transcoder type Cisco IOS Enhanced Media Termination point with the same name configured in the GW  [TRANSCODER-NAME]

 

Bulk administration

Allows the creation of devices, users, DN,… massively. Template is available in Bulk Administration > Upload/Download files > bat.xlt.

 

Serviciability

This section of the CUCM allows the management of the services.

Tools > Service Activation – To activate any of the required services. Main services are Cisco Call Manager, Cisco IP Voice Media Streaming, Cisco TFTP, Cisco Dir Sync.

Tools > Control Center > Feature Services – List of operative services. It allows the stop and restart of services.

 

Disaster Recovery System

Backup > Backup Device – Setup a SFTP server where the backups will be stored

Backup > Schedule – Setup the schedule of the backup

 

CLI

Show commands

show status

show dbreplication status

 

Reporting

CDR is the report of the calls done per extension. It’s accessible thorugh the URL https://CUCM-IP/car/. They need to be enabled in Service Parameters.

 

VM requirements

VM Requirement Settings to install CUCM:
Vmware type Red hat 5 64 bits
1 CPU with 2 cores
RAM 4096MB
SCSI LSI Logic
HD: 80GB

 

Password mangement

  • Web user
  • CLI platform
  • Security Password (for BBDD, careful with special characters, usually only accept alfanumberic)

Reset/Restart/Apply config

Phones will often need to reregister or restart when doing changes on its configuration. There are three options available to do so:

1) Apply config button: It will choose the appropiate reset or restart option
2) Reset: Software restart. Phone will just reregister to the CUCM
3) Restart: The phone will physically reboot and start the boot process.

When requesting any of the previous orders, the phone will wait until any call in progress finishes.