Sccp and sip phone registration

SCCP phones registration:

telephony-service

ephone

ephone-dn

ephone-template


SIP phones registration:

voice register global

voice register pool

voice register dn

voice register template

Cue voice view and integrated messaging - how to enable

Integrated messaging:

service imap
enable
groupname sales privilege vm-imap

Voice view feature:

service voiceview
enable

Voice mail notification:

voicemail notification enable
voicemail notification email attach

Cue evaluation license

Command show license all will show you the license. Pay attention to the fact that if number of ivr sessions exceeds or is equal with the number of ivr ports, the voice mailbox feature will be disabled.
 https://communities.cisco.com/docs/DOC-17720

Port Calculation for CUE 7.1 and later

Common error message:

CUE8-0-1# show license status application
voicemail disabled, ivr session quantity (8) is equal to or exceeds
available ports (8)
ivr enabled: 8 sessions

IVR sessions reduce the number of sessions available for VM.
VM Sessions = Ports minus IVR Sessions.
If Ports = 8 and IVR Sessions = 8 then VM Sessions = 0

Majority of the users do not use IVR features so you may not install and/or activate an IVR license.

CUE# license activate ?
  ivr               Activate IVR feature license
  ports             Activate CUE/IVR ports license
  voicemail         Activate CUE voicemail feature license

Please issue 'license activate ivr sessions N' where N should be 0, 2,
or 4 or 6 so that voicemail will get 8-N sessions, then issue 'reload'.

For installing/upgrading the software on ISM-SRE/SM-SRE modules:

These new modules offer to install the files directly form the router's interface.
You need not session in the CUE's inetrface.
Syntax:
service-module sm slot/0 install url url [script script-name] [argument argument] [force]
Examples:
1.Router(config)# service-module ism 0/0 install url ftp://username:password@128.107.146.189/dir/cue-vm-k9.sme.7.1.2.pkg script cue-vm-k9.sme.7.1.2.sre

In case of anonymous configuration on the FTP
2.Router(config)# service-module ism 0/0 install url ftp://128.107.146.189/cue-vm-k9.sme.7.1.2.pkg script cue-vm-k9.sme.7.1.2.sre

For fresh installation the file to be installed first is
The package file to install is cue-vm-k9.sme.7.1.2.pkg

Make sure the rest of the files are stored in the directory mentioned in the URL above:
cue-installer.sme.8.0.x
cue-vm-k9.sme.8.0.x.pkg
cue-vm-full-k9.sme.8.0.x.prt1
cue-vm-installer-k9.sme.8.0.x.prt1
cue-vm-langpack.sme.8.0.x.prt1
The lang .pkg file you wish to install.

Link for detali inofmration:
Installing Cisco Unity Express 8.0 Software on Services Ready Engine (SRE) Modules:
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel8_0/install/installsre.html

For activating the permanent license:

In some cases even though the permanent license has been installed, the show license in use would show up the evaluation license.

CUE8-0-1# show license in-use
StoreIndex:  4  Feature: VMIVR-PORT                        Version: 1.0
        License Type: Evaluation
        License State: Active, In Use
        License Count: 8 /8
        License Priority: Medium
StoreIndex:  5  Feature: VMIVR-IVR-SESS                    Version: 1.0
        License Type: Evaluation
        License State: Active, In Use
        License Count: 2 /2
        License Priority: Medium
StoreIndex:  7  Feature: VMIVR-VM-MBX                      Version: 1.0
        License Type: Evaluation
        License State: Active, In Use
        License Count: 105 /5
        License Priority: Medium

 

Please run the following command and reboot the CUE:

CUE8-0-1# license modify priority VMIVR-IVR-SESS low
Similarly for all the other features
After the CUE is rebooted the CUE would boot up with the permanent licenses.

DSP Hardware Based Conferencing configuring steps

1    Enable DSP farm services.

2    Enable SCCP for conference resource integration.

3    Associate Cisco Unified Communications Manager Express with a DSP farm profile.

4    Enable telephony service for hardware conferencing.

5    Configure ad hoc and Meet-Me numbers.

6    Configure IP phone softkey and administrator options.

7    Verify ad hoc and Meet-Me conferencing.

Unified mobility (mobile connect) - SNR configuration

1. Assign mobility softkey to phones by using a customized sofkey template
2. Enable mobility on end users




3. Assign user to ip phone as owner on phone configuration

4.  Create a remote destination profile

5. Create remote destination information

Extension mobility steps

1. Verify Extension Mobility Service by going to Cisco Unified CallManager Serviceability - Feature Services
2. Configure Extension Mobility Service by using url: http://:8080/emapp/EMAppServlet?device=#DEVICENAME#
3. Create Device Profile Default for Each Phone Model with Cisco Extension Mobility
4. Create Device User Profile for a User
5. Associate User Device Profile to a User
6. Configure and Subscribe Cisco Unified Ip Phones to Service

Device mobility steps

1. configure phisycal locations (ex. Arizona-Paris)
2. configure device mobility groups (ex. USA-Europe)
3. configure device pools
4. configure device mobility infos (subnets)
5. enable device mobility (per cluster or per phone)

How to add a hardware conference bridge to callmanager

Hello,

For this configuration i used a 2911 router with dsp's configured as a dspfarm for CUCM server. You have to configure that if you want to do a meet me conference between 2 ip phones that uses different codecs: one g711 and the other g729. So, the router configuration is the following:

voice-card 0
dsp services dspfarm
sccp local gigabitEthernet 0/0
sccp ccm 10.1.1.40 identifier 1 version 7.0+  // 10.1.1.40 is the address of CUCM server

sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register 2911voice            //2911voice must be THE SAME on CUCM server config
sccp

dspfarm profile 1 conference           //specify the use of dspfarm
codec g711ulaw               //supported codecs
codec g729br8
maximum sessions 1         //support only 1 conference call

associate application sccp     //associates dsp farm with sccp application
no shut

On CUCM, this is the configuration made under the Conference bridge:

In order to user this hardware bridge, you have to do a media resource group and a media resource list:


















The we apply this media resource group list to a device pool or to a specific phone.

Extension mobility on CME

Hello,

In order to do extension mobility, you must first create the extension:
ephone-dn  18
 number 5002
 description Extension for mobility
 name Extension mobility

The we must create user profile and user logout profile:
voice logout-profile 1
 user test password 7911
 number 6000 type feature-ring  //extension to assign to the phone after logout
!
voice user-profile 2
 pin 1111
 user user password 1111
 number 5002 type normal //extension to assign to the phone

The final step is to assign the logout profile to a phone:
ephone  6
 device-security-mode none
 description Phone Extension mobility
 mac-address xxxx.xxxx.xxxx
 type 7911
 logout-profile 1

CME b-acd script how to

I'll show you how to configure b-acd script on cme version 7.0. I used a 2811 router with ios version 12.4(13r)T.
This script is used to do a mini callcenter with callmanager express.
I downloaded b-acd-2.1.2.3 script from cisco and copied to router flash by tftp.
Then i defined 1235 and 1236 as 2 hunt groups:
ephone-hunt 1 longest-idle
 pilot 1235
 list 1002, 1115 #those are the extensions
!
ephone-hunt 2 longest-idle
 pilot 1236
 list 1003, 1116

This is the working script:

application
 service callcenter_queue flash:cme-b-acd-2.1.2.3/app-b-acd-2.1.2.3.tcl
  param queue-len 10
  param aa-hunt0 1235
  param aa-hunt1 1236
  param queue-manager-debugs 1
  param number-of-hunt-grps 2
 !
 service callcenter_aa flash:cme-b-acd-2.1.2.3/app-b-acd-aa-2.1.2.3.tcl
  paramspace english index 1
  param call-retry-timer 15
  param service-name callcenter_queue
  paramspace english language en
  param second-greeting-time 60
  paramspace english location flash:cme-b-acd-2.1.2.3/
  param max-time-vm-retry 2
  param voice-mail 5003
  param max-time-call-retry 700
  param aa-pilot 1234
  param number-of-hunt-grps 2
  param dial-by-extension-option 3
  param handoff-string callcenter_aa
  param welcome-prompt flash:cme-b-acd-2.1.2.3/en_bacd_welcome.au

Dial-peer made for reaching the script pilot (1234) is:

dial-peer voice 22222 voip
 service callcenter_aa
 destination-pattern 1234
 session target ipv4:192.168.1.1 #this is the loopback address of the router
 incoming called-number 1234
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

To make things happen, you have to also assign the moh file:
telephony-service
 moh flash:cme-b-acd-2.1.2.3/en_bacd_music_on_hold.au

 The debug commands that i used were:
 debug voip application script
 debug voip application tcl
  If you have any questions feel free to ask and i'll answer you as soon as possible.

Night service feature configuration

Night service bell feature activated on a phone means that if that phone is called in a preconfigured interval of time, another phone will be notified of this. This are the lines of configuration in cme:

telephony-service
night-service code *12345           //a code user to activate/deactivate night service
night-service day Mon 18:00 09:00      //interval of time when night service is active

ephone-dn 1
night-service bell              //the line to monitor

ephone 2
night-service bell             //the phone to notify

If we want to also callforward the call, not just notify:
ephone-dn 1
call-forward night-service other-phone-dn

CME sip phone registration how to

Hello,

The configuration below is running on CME and it was tested with X-lite sip phone.

voice service voip
   allow-connections sip to sip //Allows SIP phones to call other SIP phones
   sip
   registrar server

voice register global
    mode cme
    source-address 10.1.1.80 port 5060 //cme ip and default sip port
    authenticate register
    tftp-path flash:
    create profile

voice register dn  1
    number 4001
    name SIPtest

voice register pool  1
    id mac 000A.E4C4.E777 //phone mac
    number 1 dn 1
    username 4001 password test //sip username and password
    codec g711ulaw

X-lite phone configuration is below. Pay attention that authentication username and registration number must be the same, eg 4001.

Debug commands i used on router are:
debug voice register events and debug voice register errors

How to implement restrictions in Call Manager Express

COR lists are assigned to dial-peers or ephone dns.
There are incoming and outgoing cor lists.
The incoming call list is matched against the outgoing. The call will be routed only if the outgoing call list is a subset of incoming cor list.
This is a great article from Cisco:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml
For those who have a background with Call Manager, the similarities of concepts between CME and CM are in the "COR vs Cisco CallManager" section of the article.

How to calculate bandwidth for a g711 call without tunneling

I’ll show you how to calculate the bandwidth allocated for a voip call. The results and the method of calculation are shown in the table from this link.I will show more detailed how to calculate the bandwidth in relation with the codec chosed for transmission. There are 2 formulas that we will use:

1) Codec bytes per sample= (codec sample interval * codec bandwidth) / 8
Codec bandwidth value can be taken from the router:

Cisco default value for codec bytes per sample for g711 codec is 160. This and the other possible values can also be taken from the router:

So, for the g711ulaw codec we calculate the codec sample interval:
160 = (codec sample interval * 64000) / 8
Codec sample interval = 0.02 seconds = 20 ms

2) Total bandwidth = Packet size * Packet per sample
We calculate packets per sample: 1000/20 = 50 (because 1 second = 1000ms)
As it said in the link above: "These protocol header assumptions are used for the calculations:
• 40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes) headers.
• Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over Ethernet).
• 6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header.
• 18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check (CRC)."

So, for 160 bytes codec sample size we add 40 bytes for IP and because we transmit voice packets over Ethernet we add another 18 bytes:
160 + 40 + 18 = 218 bytes per sample
If we would use tunneling , we would also add these values to the sum above:
IPSEC – 50-57 bytes
GRE – 24 bytes
Using the second formula (Total bandwidth = Packet size * Packet per sample) we have:
Total bandwidth = 218 * 50 = 10900 bytes per second.
Because 1 byte= 8 bits we have:
10900 * 8 = 87200 bps = 87,2 kbps bandwidth that a call with g711ulaw codec over Ethernet without tunneling will take.