Cisco Spiad - Lab 551c5h
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Cisco SPIAD
Hands On Technical Training
Lab Guide
Table of Contents
TASK 1:
LAB TOPOLOGY
TASK 2:
LAB OVERVIEW
TASK 3:
NETWORK SETUP
TASK 4:
CONFIGURING CCME AS KEY SYSTEM
TASK 5:
VOICEMAIL & AUTO ATTENDANT
TASK 6:
ADVANCED FEATURES
Disclaimer This lab is primarily intended to be a learning tool. In order to convey specific information, the lab may not necessarily follow best practice recommendation at all times. This exercise is intended to demonstrate one way to configure the network, servers and applications to meet specified requirements for the lab environment. There are various ways that this can be accomplished, depending on the situation and the customer’s goals/requirements. Please ensure that you consult all current official Cisco documentation before proceeding with a production/lab design or installation. By enrolling in this class or having access to this document you acknowledge you are aware of this disclaimer and its implications.
TASK 1: LAB TOPOLOGY
In the lab document, ‘xx’ refers to your Pod number. For example, if you are seated at POD01, you would replace all instances of ‘xx’ in the lab documents with ‘01’. In order to get PSTN access a SIP trunk was set up for each POD, this is to emulate a PSTN connection for the lab environment. Normally, you would plug trunks from the PSTN into the FXO ports or T1/E1 port. Hardware and software requirements: Cisco SPIAD running CME 10.5 LAN Switch (2900, 300 or 500 series, Meraki MS) Serial Console Cable Cisco Phone 78xx, 79xx or any other ed model Windows PC Cisco IP Communicator Client SSH client and terminal emulator (putty client) TFTP server software FTP Server software
WAN IP addressing, Interface GigabitEthernet 0/0: WAN IP Address WAN Subnet Mask WAN Default Router
DH DH DH
LAN IP addressing Interface GigabitEthernet 0/1: LAN_DATA IP Address LAN_DATA Subnet Mask VLAN_DATA
192.168.10.1 255.255.255.0 VLAN 1
LAN_VOICE IP Address LAN_VOICE Subnet Mask VLAN_VOICE
10.1.1.1 255.255.255.0 VLAN 100
ISM Module0/0 IP addressing: ISM IP Address ISM Subnet Mask ISM Default Router
10.1.10.1 255.255.255.252 10.1.10.2
Inbound SIP Trunk DID (PSTN Call in) Numbers (Use PSTN phone to test) Auto attendant (200) Extension 1 (201) Jim Smith Extension 2 (202) Sara Noa Extension 2 (203) Emma Smith VoiceMail (399)
4085xx1200 4085xx1201 4085xx1202 4085xx1203 4085xx1209
Outbound Numbers to Call (From IP Phones on SPIAD to PSTN Phone) Emergency Number Local Call Long Distance International
9060 952671800 9018182212462 90014085256800
SPIAD name and name
spiad spiad
Cisco Unity Express name and name
spiad spiad
Cisco Communications Manager Express GUI name and name
spiad
TASK 2: LAB OVERVIEW Lab Objective: To gain experience configuring the Cisco SPIAD solution, configuring call processing, Key System, PBX, Voicemail, Auto Attendant, hunt groups, and additional features of the Cisco Communications Manager Express
Audience and Prerequisites This document is intended to assist solution architects, sales engineers, field engineers, and consultants in learning many of the features of Cisco Communications Manager Express 10 running on a bundled system (CISCO SPIAD). This document assumes the reader has an architectural and istrative understanding of the CCME and has reviewed the latest istration guide. Basic knowledge of how to install and ister CCME and CUE is recommended however not necessary. All configurations are based on Command Line Interface (CLI) so is highly recommendable that you were familiar with the IOS commands. Please refer to the following links to get further information about operation of Cisco Communications Manager Express and Cisco Unity Express: Cisco CME main page: http://www.cisco.com/go/ccme Cisco CME Install Guides: http://www.cisco.com/c/en/us//unified-communications/unified-communicationsmanager-express/products-installation-guides-list.html Cisco Unity Express: http://www.cisco.com/go/cue Cisco Unity Express Maintain and Operate Guides: http://www.cisco.com/c/en/us//unified-communications/unity-express/productsmaintenance-guides-list.html Cisco IOS compatibility matrix http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/requirements/guide/33matri x.html
TASK 3: NETWORK SETUP In order to get access to the SPIAD system a previous configuration for voice and data VLAN was loaded, as well as DH pool for phones and s.
config terminal hostname CISCO_SPIAD ip dns server name spiad privilege 15 secret spiad ntp server 24.56.178.140 interface Vlan1 description DATA_VLAN no ip address interface Vlan100 description VOICE_VLAN no ip address interface GigabitEthernet 0/1 no ip address no shutdown interface GigabitEthernet0/1.1 description DATA_INTERFACE encapsulation dot1Q 1 native ip address 192.168.10.1 255.255.255.0 interface GigabitEthernet0/1.100 description VOICE_INTERFACE encapsulation dot1Q 100 ip address 10.1.1.1 255.255.255.0
interface GigabitEthernet0/0 description WAN_INTERFACE ip address dh
ip dh excluded-address 192.168.10.1 192.168.10.10 ip dh excluded-address 10.1.1.1 10.1.1.10 ip dh pool data import all network 192.168.10.0 255.255.255.0 default-router 192.168.10.1 dns-server 192.168.10.1 option 150 ip 10.1.1.1 ip dh pool phone import all network 10.1.1.0 255.255.255.0 default-router 10.1.1.1 option 150 ip 10.1.1.1 Assuming that LAN switch has been configured and both VLANs (voice and data) were assigned on designated ports, plug the interface Gigabit Ethernet 0/1 to the uplink switch trunk port. Check that your PC is connected to one of the switch ports and validate DH assignment C:\> ipconfig
Test connectivity to the default gateway assigned by SPIAD. C:\> ping 192.168.10.1
ISM Module Configuration On Cisco SPIAD Cisco Unity Express is loaded from factor in ISM 0/0, Voice Mail and Auto attendant Ports licenses are loaded also, so in order to get access to CUE IP addressing; configuration is needed, this section describes the configuration procedure. Open SSH putty client and create a new session, use the IP address 192.168.10.1 click open and accept the security alert.
When prompt appears, type spiad for and spiad for .
Load the following configuration that is needed to get Internal Service Module reachable:
config terminal ! Create a loopback interface interface Loopback0 ip address 10.1.10.2 255.255.255.252 ip virtual-reassembly in ! according to the lab topology assign the ip address 10.1.10.1/30 and default gateway 10.1.10.2 interface ISM0/0 ip unnumbered Loopback0 ip nat inside ip virtual-reassembly in service-module ip address 10.1.10.1 255.255.255.252 !Application: CUE Running on ISM service-module ip default-gateway 10.1.10.2 no shutdown ! add an static route in order to reach CUE via ISM0/0 ip route 10.1.10.1 255.255.255.255 ISM0/0
Validate that you can reach ISM IP address and its default gateway from the router pinging both IP addresses 10.1.10.1 and 10.1.10.2
Do the same tests, now form your PC.
Cisco Unity Express will be configured later in this lab at this point you have done with ISM configuration.
Internet Connection PSTN access will be provided via SIP trunk in this case internet connection is needed, this section will provide a basic configuration to gain internet access for voice, data VLAN and ISM deploying NAT.
config terminal ! Define NAT inside interfaces interface GigabitEthernet0/1.1 ip nat inside interface GigabitEthernet0/1.100 ip nat inside interface ISM0/0 ip nat inside ! Define NAT outside interfaces interface GigabitEthernet0/0 ip nat outside ! Create an access list to allow Internet access to the following source subnets access-list 1 permit 192.168.10.0 0.0.0.255 access-list 1 permit 10.1.1.0 0.0.0.255 access-list 1 permit 10.1.10.0 0.0.0.3 ! Packets received on inside interfaces and permitted on access list 1 will be translated to the IP address assigned to interface Gigabit Ethernet 0/0 ip nat inside source list 1 interface GigabitEthernet0/0 overload end Don’t forget to save your configuration copying running configuration to startup configuration.
Validate that you can get internet access, open your browser and go to: http://www.cisco.com.
Internet connection is done.
TASK 4: CONFIGURING CCME AS IPPBX SYSTEM In this section CCME will be configured as key system, we will see a couple of examples when CCME is enabled to Cisco Proprietary Skinny Client Control Protocol (SC) phones and Session Initiation Protocol (SIP) phones. The simplest model is the PBX model, in which most of the IP phones in your system have a single unique extension number. Incoming PSTN calls are routed to a receptionist at an attendant console or to an automated attendant. Phone s may be in separate offices or be geographically separated and therefore often use the telephone to each other. For this model, is recommended that you configure directory numbers as dual-lines so that each button that appears on an IP phone can handle two concurrent calls. Dual-line directory numbers enable your configuration to call waiting, call transfer with consultation, and three-party conferencing. This section contains a list of the types of files that must be ed and installed in the router flash memory to use with Cisco Unified CME. The files listed in this section are included in zipped or tar archives that are ed from the Cisco Unified CME software website at: https://software.cisco.com//type.html?mdfid=277641082 To get software a valid CCO ID is needed with and partner level privileges for such section. Now let’s continue with the
Configure the Cisco Unified Communications Manager Express Parameters The following configuration besides to start the telephony service will allow SC phones registration.
configure terminal ! Enter the command telephony−service in order to enter telephone configuration mode.
telephony−service ! Enter the command max−ephones max−num−phones in order to set the maximum number of IP phones to be ed by this platform.
max-ephones 30 ! Enter the command max−dn max−directory−numbers in order to set the maximum number of extensions that can exist in this platform.
max-dn 200
! For security reasons enter the command no auto−reg−ephone in order to prevent the connection of any phone to the system.
no auto-reg-ephone ! Enter the command load phone−type firmware−file in order to identify the firmware file that the IP phone uses to in the system.
load 7975 SC75.9-2-1S ! Set time zone for Mexico Standard/Daylight, display time format as 24 hours and set date format as dd-mm-yy
time-zone 9 time-format 24 date-format dd-mm-yy ! Assign the voice mail extension 399 according to the Lab Topology
voicemail 399 ! Define the call forward and call park behavior
call-park system application call-forward pattern .T call-forward system redirecting-expanded ! Define the call transfer behavior allowing transfer to any number and call transfer method
transfer-system full-consult dss transfer-pattern 9.T transfer-pattern .T ! Create another tone when you dial 9 to place an outside call.
secondary-dialtone 9 ! Enter the command ip source−address ip−address in order to identify the IP address and port number that the Cisco CallManager Express router uses for IP phone registration. The default port is 2000.
ip source-address 10.1.1.1 ! Set the interdigit timeout
timeouts interdigit 5
! Enter the command create cnf−files in order to build the XML configuration files.
create cnf-files end This is a generic phone template that will be used by SC phones load it.
configure terminal ephone-template 16 url services 1 http://10.1.10.1/voiceview/common/.do VoiceviewExpress softkeys remote-in-use Newcall softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback softkeys connected Hold Endcall Trnsfer TrnsfVM Confrn ConfList RmLstC Acct Park Select Don’t forget to save your configuration copying running configuration to startup configuration.
Configure the Cisco Unified Communications Manager Express to allow SIP phones registration. To enter voice global configuration mode in order to set global parameters for all ed Cisco SIP IP phones in a Cisco Unified CME or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) environment, use the voice global command in global configuration mode. To automatically remove the existing DNs, pools, and global dialplan patterns, use the no form of this command.
config terminal voice global ! Enables mode for provisioning SIP phones in Cisco Unified CME
mode cme ! Enter the command source−address ip−address in order to identify the IP address and port number that the Cisco CallManager Express router uses for IP phone registration.
source-address 10.1.1.1 port 5060 ! SIP Outbound Proxy won’t be needed for SIP phones
no outbound-proxy ! Enter the command max−pool max−num−phones in order to set the maximum number of IP phones to be ed by this platform.
max-pool 30 ! Enter the command max−dn max−directory−numbers in order to set the maximum number of extensions that can exist in this platform.
max-dn 200 ! Enter the command load phone−type firmware−file in order to identify the firmware file that the IP phone uses to in the system.
load 7841 sip78xx.10-2-1-12 ! Registration requests from SIP phones in a Cisco Unified CME system must be authenticated
authenticate
! Assign the voice mail extension 399 to the message button according to the Lab Topology
voicemail 399 ! Set a repeating audible alert notification when a call is on hold on all ed SIP phones directly connected in Cisco Unified CME
hold-alert ! Specify the directory to which the configuring files for SIP phones in Cisco Unified CME are written
tftp-path flash: ! Set time zone for Mexico Standard/Daylight, display time format as 24 hours and set date format as dd-mm-yy
timezone 9 time-format 24 date-format D/M/Y ! Enter the command create profile in order to build the XML configuration files.
create profile exit
Configure voice service voip
! Create a SIP profile in order to normalize SIP signaling
voice class sip-profiles 1000 request ANY sdp-header Connection-Info remove response ANY sdp-header Connection-Info remove ! Start voice service voip
voice service voip ! allow only trusted hosts or network
ip address trusted list ipv4 10.1.1.0 255.255.255.0 ipv4 192.168.10.0 255.255.255.0 ipv4 10.1.10.0 255.255.255.252
! Allow interworking or leg to leg communication
allow-connections allow-connections allow-connections allow-connections
h323 to h323 h323 to sip sip to h323 sip to sip
! Disable SIP redirect response for call forwarding, Disable SIP REFER message for call transfers and enable call-by-call detection of H.450 capabilities
supplementary-service h450.12 no supplementary-service sip moved-temporarily no supplementary-service sip refer ! FAX relay configuration
fax protocol none no fax-relay sg3-to-g3 sip ! enable SIP registrar functionality
registrar server expires max 3600 min 120 no update-callerid sip-profiles 1000 copy running-config startup-config
CCME is now ready to receive SC or SIP phones registration, but phone loads needs to be ed into the flash directory and define the TFTP alias for such loads, the ones needed for this lab are for Cisco Phones 7842 (SIP) and 7975 (SC).
Start TFTP server and set the root path as follows:
Phone loads compressed files for Cisco Phones are: 7841 (SIP) cmterm-78xx.10-2-1-12_REL.tar 7975 (SC) cmterm-7975-sc.9-2-1.tar
Open a SSH session to the SPIAD system if you are not logged in. And follow the next procedure to and uncompress phones firmware files. Create a directory to save the files mkdir flash:phones now create a directory for each model mkdir flash:phones/7800 mkdir flash:phones/7975 validate that your directories were created dir flash:phones
Proceed to and extract files from TFTP server archive tar /xtract tftp://192.168.10.11/cmterm-78xx.10-2-1-12_REL archive tar /xtract tftp://192.168.10.11/cmterm-7975-sc.9-2-1.tar flash:/phones/7975
Once all the files were extracted check that files are on the directories that were created previously.
Now create the TFTP alias for each file that was ed in order to make the file reachable when phone request it.
configure terminal ip tftp source-interface Loopback0 tftp-server tftp-server tftp-server tftp-server
flash:/phones/7800/kern78xx.10-2-1-12.sbn alias kern78xx.10-2-1-12.sbn flash:/phones/7800/rootfs78xx.10-2-1-12.sbn alias rootfs78xx.10-2-1-12.sbn flash:/phones/7800/sboot78xx.10-2-1-12.sbn alias sboot78xx.10-2-1-12.sbn flash:/phones/7800/sip78xx.10-2-1-12.loads alias sip78xx.10-2-1-12.loads
tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server
flash:/phones/7975/apps75.9-2-1TH1-13.sbn alias apps75.9-2-1TH1-13.sbn flash:/phones/7975/cnu75.9-2-1TH1-13.sbn alias cnu75.9-2-1TH1-13.sbn flash:/phones/7975/cvm75sc.9-2-1TH1-13.sbn alias cvm75sc.9-2-1TH1-13.sbn flash:/phones/7975/dsp75.9-2-1TH1-13.sbn alias dsp75.9-2-1TH1-13.sbn flash:/phones/7975/jar75sc.9-2-1TH1-13.sbn alias jar75sc.9-2-1TH1-13.sbn flash:/phones/7975/SC75.9-2-1S.loads alias SC75.9-2-1S.loads flash:/phones/7975/term75.default.loads alias term75.default.loads
end copy running-config startup-config
Phone firmware is now loaded and ready to upgrade phones if it is necessary. In case you have a phone on a pre-8.3.2 firmware and can't do a direct 9.2.1. You should load an interim firmware for instance 8.4.4 on your flash and upgrade it to that load first. Then 9.1.2.
Cisco Unified Communications Manager Express Phone Provisioning In order to start to make and receive calls, you need to the specific IP phones that you want on the CCME system. In this process you set up individual ephone−dns or -dns and then associate each with a button or buttons on one or more ephones. Each ephone−dn or -dn is a virtual line, or extension, on which call connections can be made. Each physical phone must be configured as an ephone or pool in the Cisco CME router in order to receive in the LAN environment. With the use of the ephone−dn command and dual−line keyword you create an ephone−dn in dual−line mode. The reason is to have one voice port and two channels in order to handle two independent calls. This mode enables call transfer, call waiting, and conference options.
ing Cisco SC phones to CCME This procedure s ephones and ephones−dns to Cisco SPIAD:
configure terminal ! Create the ephone directory number with dual-line mode for Jim Smith, in case that phone is busy or no answer call will be forwarded to voice mail extension.
ephone-dn 1 dual-line number 201 no-reg both label 201 description Jim Smith name Jim Smith call-forward busy 399 call-forward noan 399 timeout 10 ! Create the ephone directory number with dual-line mode for Sara Noa, in case that phone is busy or no answer call will be forwarded to voice mail extension.
ephone-dn 2 dual-line number 202 no-reg both label 202 description Sara Noa name Sara Noa call-forward busy 399 call-forward noan 399 timeout 10
! Create the ephone for Jim Smith and assign the ephone-dn 1 to button 1 *replace mac-address for the one of your phone 7975 SC IP phone.
ephone 1 mac-address 0026.99EF.1DEB type 7975 ephone-template 16 name "jsmith" qwer201 button 1:1 ! Create the ephone for Sara Noa and assign the ephone-dn 2 to button 1 *replace mac-address for the one of your Cisco IP Communicator Client (CIPC)
ephone 2 mac-address AAAA.BBBB.0001 type CIPC ephone-template 16 name "snoa" qwer202 button 1:2 ! Go to telephony-service an create CNF files
telephony-service create cnf-files end copy running-config startup-config
Connect your Cisco IP Phone 7975 to the switch, if your phone was upgraded you can see on your console typing: sh phone-load Output shows something like this:
Cisco IP Communicator Client (CIPC) Configuration Sara Noa has no a physical phone assigned, instead IT deparment has installed Cisco IP communicator client which is a SC softphone that offers all features that you could find in a Cisco phone 7975, an ephone and an ephone-dn have been configured previously. IP Communicator Client is available on Cisco Software: https://software.cisco.com//release.html?mdfid=278468661&catid=280789323&s oftwareid=282074237&release=8.6(4)&relind=AVAILABLE&rellifecycle=&reltype=latest When the software is installed, run the application and perform the audio test if needed. On any place of the software interface right click and select menu preferences…
On the new window opened go to network tab, click on use this device name option, if you on the ephone 2 configuration, mac-address AAAA.BBBB.0001 was assigned to this device for Sara Noa. The device name is composed by prefix SEP followed by the mac address in this case is SEPAAAABBBB0001. TFTP server is needed also to load CIPC parameters, TFTP server will be reached on IP address 10.1.1.1.
Click OK and wait until CIPC is reloaded, when finished you will see that it is ed and extension 202 is assigned on button one to Sara Noa. Now Sara Noa should be able to call Jim Smith on extension 201, and he can call her back to extension 202. Try a call from Jim to Sara and validate that basic call between SC phones are working.
ing Cisco SIP phones to CCME This procedure s SIP phones and SIP DNs to CCME on Cisco SPIAD:
configure terminal ! Create Directory Number 203 associated to Emma Smith in case that no answer call will be forwarded to voice mail extension 399
voice dn 1 number 203 call-forward b2bua noan 399 timeout 20 name Emma Smith label Emma Smith ! Prevent that this DN attempts to to external SIP proxy
no-reg ! Create the voice pool for the cisco phone 7841 associated to Emma Smith, replace id mac with the mac address of your phone
voice pool 1 id mac 2C3E.CF86.23C0 type 7841 ! Assign voice dn 1 to button 1
number 1 dn 1 ! Set DTMF relay method
dtmf-relay rtp-nte ! set the name and for this device
name esmith qwer203 ! force codec to G711ulaw
codec g711ulaw ! go to voice global and create configuration files
voice global create profile end copy running-config startup-config Connect your 7841 and place some calls to the other extensions.
Configuring PSTN access via SIP trunk In order to gain PSTN access, in this section we will see the procedure to get a SIP Trunk ed against a simulated Internet Telephony Service Provider. This Are the requirements and SIP trunk information. Inbound SIP Trunk DID (PSTN Call in) Numbers (Use PSTN phone to test) Auto attendant (200) Extension 1 (201) Jim Smith Extension 2 (202) Sara Noa Extension 3 (203) Emma Smith VoiceMail (399)
4085xx1200 4085xx1201 4085xx1202 4085xx1203 4085xx1209
Outbound Numbers to Call (From IP Phones on SPIAD to PSTN Phone) Emergency Number Local Call Long Distance International
9060 952671800 9018182212462 90014085256800
SIP Trunk information; XX corresponds to your POD number SIP Server SIP Outbound Proxy name
4.31.34.33 4.31.34.33 4085xx1200 4085xx1200
In this case SIP Trunk requires authentication for each number, follow the next steps to get registration from ITSP
configure terminal voice service voip sip outbound-proxy ipv4:4.31.34.33:5060 sip-ua authentication name 4085011200 4085011200 credentials name 4085011200 4085011200 realm credentials name 4085011201 4085011200 realm credentials name 4085011202 4085011200 realm credentials name 4085011203 4085011200 realm no remote-party-id retry invite 2 retry 10 timers connect 100 registrar ipv4:4.31.34.33:5060 expires 300 sip-server ipv4:4.31.34.33:5060 host-registrar
4.31.34.33 4.31.34.33 4.31.34.33 4.31.34.33
end copy running start
Validate that SIP trunk was successfully ed, issue: show sip-ua status
In order to make and receive calls some translation rules and profiles are needed, follow these steps to create translation rules for incoming and outgoing calls. Allow sip server signaling IP address into the toll fraud prevention list
configure terminal voice service voip ip address trusted list ipv4 4.31.34.33 255.255.255.255 end copy running-config startup-config Outgoing Calls Setup configure terminal ! This service provider works only with codec G711u
voice class codec 1 codec preference 1 g711ulaw ! Following translation rules remove 9 prefix and translate source extensions and any number into the main DID 4085011200
voice translation-rule 410 rule 1 /^9\(.*\)/ /\1/ rule 15 /^...$/ /4085011200/
voice translation-rule 1111 rule 15 /^.*/ /4085011200/ voice translation-rule 1112 rule 1 /^9/ // ! Following profiles translate calling or called number according to translation rules defined, these profiles will be applied later on voice dial-peers
voice translation-profile CALLER_ID_TRANSLATION_PROFILE translate calling 1111 ! voice translation-profile OUTGOING_TRANSLATION_PROFILE translate called 1112 !
voice translation-profile PSTN_CallForwarding translate redirect-target 410 translate redirect-called 410 voice translation-profile PSTN_Outgoing translate calling 1111 translate called 1112 translate redirect-target 410 translate redirect-called 410
Translation rules were created, the next step is to create voip outgoing voice dial-peers, notice that session pattern includes outgoing prefix 9, session target is pointing to SIP server that was previously configured on SIP trunk section, class-codec and dtmf relay method may vary from carrier to carrier.
! dial-peer voice 1021 voip description **Local Calls - 7, 8 or 10 digits** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 9[1-9]T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip asymmetric payload full dtmf-relay rtp-nte digit-drop no vad ¡
dial-peer voice 1023 voip description ** International Calls** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 900T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! dial-peer voice 1025 voip description **National Long Distance** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 901.......... session protocol sipv2 session target sip-server voice-class codec 1 no vad ! dial-peer voice 1026 voip description **CCA*Mexico*Emergency** translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE preference 1 destination-pattern 060 session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! dial-peer voice 1027 voip description **CCA*Mexico*Emergency** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 9060 session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad end write mem
Outgoing calls should be connecting now, test the numbers provided.
Incoming Calls Setup To receive calls, voice translation rules and translation profiles are needed also; the next procedure shows how to redirect an incoming call to an specific extension or application (autoattendant).
configure terminal ! Sent DID 40850xx200 to AutoAttendant extension 200, *AA will be ready later on the lab, just configure translations
voice translation-rule 6 rule 1 /4085011200/ /200/ voice translation-profile DID_AutoAtt translate called 6 ! Sent DIDs 40850xx201 to 203 to extensions 201 to 203
voice rule rule rule
translation-rule 12 1 /4085011201/ /201/ 2 /4085011202/ /202/ 3 /4085011203/ /203/
voice translation-profile DIDs-SIP_trunk translate called 12 ! In order to get incoming calls properly routed add following voip dial-peers
dial-peer voice 1000 voip permission term description ** Incoming call from SIP trunk ** session protocol sipv2 session target sip-server incoming called-number .% voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad
dial-peer voice 3012 voip description ** DIDs from SIP_trunk ** translation-profile incoming DIDs-SIP_trunk session protocol sipv2 session target sip-server incoming called-number 408501120 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! dial-peer for incoming DIDs
dial-peer voice 3012 voip description ** DIDs from SIP_trunk ** translation-profile incoming DIDs-SIP_trunk session protocol sipv2 session target sip-server incoming called-number 408501120[1-3] voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! AutoAttendant will be enabled later but this is the dial-peer needed
dial-peer voice 3002 voip description DIDAutoAtt-AA translation-profile incoming DID_AutoAtt session protocol sipv2 session target sip-server incoming called-number 4085011200 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad
Now that you can place and receive calls your PBX is setup for basic calls in the next sections we will cover Voice Mail and Autattendant integration, additionally some advanced features will be covered.
TASK 5: DEPLOYING VOICEMAIL AND AUTOATTENDAT Cisco Unified Communications Manager Express provides call processing for Cisco Unified IP Phones. Cisco Unity Express offers voicemail and automated-attendant capabilities for IP phone s connected to Cisco Unified Communications Manager Express. The voicemail and automated-attendant capabilities are fully integrated into the Cisco access router using a network module or services-ready engine (SRE). With this solution, the Cisco portfolio of access routers delivers features similar to those of a key system or hybrid private branch exchange (PBX) plus the rich data and routing capabilities expected on the award winning Cisco integrated services routers(ISRs), Customers can now deliver unified communications to their small sites and branch offices with a solution that is very simple to deploy, ister, and maintain. Cisco Unified Communications Manager Express with Cisco Unity Express offers customers a cost-effective, highly reliable, feature-rich solution for an office deployment. As was described on Internal Service Module network setup; Cisco Unity Express is loaded from factor in ISM 0/0, Voice Mail and Auto attendant Ports licenses are loaded also on Cisco SPIAD. Cisco Unity Express GUI can read s and extension from Communications Manager Express in order to do it graphical interface for CCME should be loaded and configured, follow this procedure to GUI files from TFTP server and apply the CLI commands to enable http access and create and for this task. CCME GUI setup Open a ssh connection to the SPIAD system, and extract GUI files into flash: Issue: archive tar /xtract tftp://192.168.10.11/CME_GUI_10.5.tar flash:
Apply the following commands to create the GUI istrative and to grant http access for such . GUI name: GUI :
spiad
aaa new-model ! ! aaa authentication default local !
ip http server ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ip http path flash:/GUI ! file privilege 0 ! telephony-service web system name secret spiad dn-webedit time-webedit
Open your browser and test CCME GUI access, go to http://192.168.10.1/ccme.html Type name: and spiad
After a successful authentication you should be able to see the istrative web interface for Cisco Unified Communications Manager Express, from here you can create SC phones, SC extensions and configure some telephony-service parameters. Note: SIP phones and extensions cannot be managed on this interface.
Voice Mail Message Waiting Indicator (MWI) As we said before, Cisco Unity Express GUI reads configuration from CCME in order to have all parameters ready for CUE is necessary to create MWI configuration, follow the next steps to create indicator for ON and OFF and dial-peer to get a proper communication to CUE system.
! Create ephone-dn for ON and OFF indicators
ephone-dn 199 number A801... no-reg primary mwi off ! ! ephone-dn 200 number A800... no-reg primary mwi on
! Create voice voip dial-peer to through Inbound Calls for MWI from CUE
dial-peer voice 1005 voip description ** through Inbound Calls for MWI from CUE ** b2bua session protocol sipv2 session target ipv4:10.1.10.1 incoming called-number A80T dtmf-relay rtp-nte codec g711ulaw no vad
Cisco Unity Express Setup With the previous steps done no we are ready to complete CUE setup. Open your browser and go to http://10.1.10.1/ by default name and are “cisco” When , chose Communications Manager Express Integration.
This will trigger a system reload.
Wait for a few minutes until system goes up.
Click on Run Initialization Wizard.
Enter the name and that was created previously to get access to CCME GUI.
If you don’t see the next button click TAB several times until you see it and click on it. CUE GUI has read s and extensions for CCME and can be imported; check the mailboxes for the to create a voice mail box, and choose jsmith as , click next. Left the default values for the mailboxes and click next.
Enter following values to complete setup: Voice Mail Number: 399 Voice Mail Operator Extension: 200 Auto Attendant Access Number: 200 (if no VM box call will be redirected to AA) Auto Attendant Operator Extension: 201 (this is usually reception extension) istration via Telephone Number: 397 SIP MWI Notification Mechanism: by default outcalling MWI ON Number (Outcalling mechanism): read from ccme A800… MWI OFF Number (Outcalling mechanism):read from ccme A801…
Click next.
Review the values that you just entered and commit your setup, checking the box “Finally, save to startup configuration (will take a few minutes more)” and click finish
Once system finish initialization you will see that voicemail boxes were created and configuration was success.
Click .
Creating Mailbox for a SIP SIP s cannot be imported by CUE, so if you want to assign a voicemail box, this has to be done manually, follow the next steps to add and create voice mail box for Emma Smith (esmith) extension 203. to CUE once again, use credentials name: spiad : spiad Go to configure->s and click add. Fill the new open window with the following mandatory values ID: esmith First Name: Emma Last Name: Smith Nick Name: (automatically filled with firstname lastname) Display Name: (automatically filled with firstname lastname) Associated Phone: none Primary Extension: chose other and set it to 203
Left the rest of the parameters as default and scroll down to the button, mark the check box to create mailbox Click on the icon
located on the upper left corner to finishing addition. As mailbox creation was selected in a previous step a new window is showed:
Left default values and click on the add icon Now the new esmith was created and a mailbox was associated to it. You can validate that was added going to configure->s
By default we have created and extension that forward calls to voicemail if there is no answer. Before to test voicemail and auto attendant, go to the next section to configure VOIP routing to Cisco Unity Express.
In order to get access to voicemail and basic auto attendant from SIP or SC clients some dial peers are needed, add such dial-peers as follows:
conf t dial-peer voice 2000 voip description ** cue voicemail pilot number ** translation-profile outgoing XFER_TO_VM_PROFILE destination-pattern 399 b2bua session protocol sipv2 session target ipv4:10.1.10.1 voice-class sip outbound-proxy ipv4:10.1.10.1 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2001 voip description ** through Inbound Calls for Internal Extensions from CUE ** b2bua session protocol sipv2 session target ipv4:10.1.10.1 incoming called-number ^...$ dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2002 voip description ** cue auto attendant number ** translation-profile outgoing PSTN_CallForwarding destination-pattern 200 b2bua session protocol sipv2 session target ipv4:10.1.10.1 voice-class sip outbound-proxy ipv4:10.1.10.1 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2003 voip description ** cue prompt manager number ** translation-profile outgoing PSTN_CallForwarding destination-pattern 397 b2bua session protocol sipv2 session target ipv4:10.1.10.1 voice-class sip outbound-proxy ipv4:10.1.10.1 dtmf-relay rtp-nte codec g711ulaw no vad
Save your configuration, issue:
copy running-config startup-config
Testing Voice Mail and Auto Attendant Now you can test voicemail service dialing from extension 203 (Emma Smith) to 201 (Jim Smith), left a short message and check that message indicator is turned on once you hang up. Retrive your messages pressing the message button you will be prompt for a use 12345 Do the same test now dialing from Jim Smith (201) to Emma Smith (203). To Validate Auto Attendant Operation dial from extension 202 (Sara Noa) to extension 200 you should be able to hear Auto Attendant prompt. Once you are connected chose 1 to dial by telephone number and the dial 201 then # your call should be transferred to Jim Smith. Repeat the test but this time dial to extension 203 then # Whit these tests we have finished the Voice Mail and Basic Auto Attendant deployment, your system is fully working as an IPPBX. You can place and receive call from SIP trunk directly to a specific extension and to a basic auto attendant.
TASK 6: ADVANCED FEATURES In this section we will cover some advance features that can be implemented on Cisco SPIAD, we will cover following implementation: Hunt Groups Pickup Group Call Park Meetme Conference Call restriction by (pin before dial) Extension Mobility
APPENDIX A. Cisco Unity Express Factory Restart
CISCO_SPIAD# service-module ism0/0 session Trying 10.1.10.2, 2067 ... Open se-10-1-10-1# se-10-1-10-1# offline !!!WARNING!!!: If you are going offline to do a backup, it is recommended that you save the current running configuration using the 'write' command, prior to going to the offline state. Putting the system offline will disable management interfaces. Are you sure you want to go offline?[confirm] se-10-1-10-1(offline)# restore factory default !!!WARNING!!!: This operation will cause all configuration and data on the system to be erased. This operation is not reversible. Do you wish to continue?[confirm] Restoring the system. Please wait ... ..done System will be restored to factory default when it reloads. Press any key to reload: System reloading .... MONITOR SHUTDOWN... INIT: Sending processes the TERM signal Sending an RB message to IOS notifying module reboot... Rebooting ... — SNIP — INIT: Entering runlevel: 2 ********** rc.post_install **************** IMPORTANT:: IMPORTANT:: Welcome to Cisco Systems Service Engine IMPORTANT:: post installation configuration tool. IMPORTANT:: IMPORTANT:: This is a one time process which will guide IMPORTANT:: you through initial setup of your Service Engine. IMPORTANT:: Once run, this process will have configured IMPORTANT:: the system for your location. IMPORTANT:: IMPORTANT:: If you do not wish to continue, the system will be halted IMPORTANT:: so it can be safely removed from the router. IMPORTANT:: Do you wish to start configuration now (y,n)?
B. Cisco IP Phone Factory Reset Follow the steps below to successfully Factory reset your Cisco IP phone: 1. Unplug the power cable from the ip phone and then plug it back in. 2. While the phone is powering up, and before the Speaker button flashes on and off, press and hold the hash # key. 3. Continue to hold # until each line button (right of the LCD screen) flashes on and off in sequence in orange color. 4. Now release the hash # key and type the following sequence 123456789*0# After the sequence has been entered the line buttons on the phone flash orange, then green and the phone goes through the factory reset process. This process can take several minutes and the firmware of the IP Phone will be erased. When complete, the IP phone will reboot and the bootloader will try to obtain an IP address via DH. The IP phone also expects the IP address (option 150) or the name (option 66) of the TFTP server to be delivered by the DH server. This is why these DH options are critical at this phase. The phone then tries to obtain the appropriate termXX.default.loads file depending in its model: This "loads" file indicates all the files the IP phone has to from the TFTP server to make up the device firmware. The IP phone should first obtain the “loads” file and then proceed with the individual files. Once complete, the IP phone will install the files and finally reboot C. How to Reset Cisco IP Communicator Follow the steps below to successfully clear settings on Cisco IP Communicator Client: Press Settings Button (looks like a check in a box)
Press**# (to unlock menu) Press Erase
Hands On Technical Training
Lab Guide
Table of Contents
TASK 1:
LAB TOPOLOGY
TASK 2:
LAB OVERVIEW
TASK 3:
NETWORK SETUP
TASK 4:
CONFIGURING CCME AS KEY SYSTEM
TASK 5:
VOICEMAIL & AUTO ATTENDANT
TASK 6:
ADVANCED FEATURES
Disclaimer This lab is primarily intended to be a learning tool. In order to convey specific information, the lab may not necessarily follow best practice recommendation at all times. This exercise is intended to demonstrate one way to configure the network, servers and applications to meet specified requirements for the lab environment. There are various ways that this can be accomplished, depending on the situation and the customer’s goals/requirements. Please ensure that you consult all current official Cisco documentation before proceeding with a production/lab design or installation. By enrolling in this class or having access to this document you acknowledge you are aware of this disclaimer and its implications.
TASK 1: LAB TOPOLOGY
In the lab document, ‘xx’ refers to your Pod number. For example, if you are seated at POD01, you would replace all instances of ‘xx’ in the lab documents with ‘01’. In order to get PSTN access a SIP trunk was set up for each POD, this is to emulate a PSTN connection for the lab environment. Normally, you would plug trunks from the PSTN into the FXO ports or T1/E1 port. Hardware and software requirements: Cisco SPIAD running CME 10.5 LAN Switch (2900, 300 or 500 series, Meraki MS) Serial Console Cable Cisco Phone 78xx, 79xx or any other ed model Windows PC Cisco IP Communicator Client SSH client and terminal emulator (putty client) TFTP server software FTP Server software
WAN IP addressing, Interface GigabitEthernet 0/0: WAN IP Address WAN Subnet Mask WAN Default Router
DH DH DH
LAN IP addressing Interface GigabitEthernet 0/1: LAN_DATA IP Address LAN_DATA Subnet Mask VLAN_DATA
192.168.10.1 255.255.255.0 VLAN 1
LAN_VOICE IP Address LAN_VOICE Subnet Mask VLAN_VOICE
10.1.1.1 255.255.255.0 VLAN 100
ISM Module0/0 IP addressing: ISM IP Address ISM Subnet Mask ISM Default Router
10.1.10.1 255.255.255.252 10.1.10.2
Inbound SIP Trunk DID (PSTN Call in) Numbers (Use PSTN phone to test) Auto attendant (200) Extension 1 (201) Jim Smith Extension 2 (202) Sara Noa Extension 2 (203) Emma Smith VoiceMail (399)
4085xx1200 4085xx1201 4085xx1202 4085xx1203 4085xx1209
Outbound Numbers to Call (From IP Phones on SPIAD to PSTN Phone) Emergency Number Local Call Long Distance International
9060 952671800 9018182212462 90014085256800
SPIAD name and name
spiad spiad
Cisco Unity Express name and name
spiad spiad
Cisco Communications Manager Express GUI name and name
spiad
TASK 2: LAB OVERVIEW Lab Objective: To gain experience configuring the Cisco SPIAD solution, configuring call processing, Key System, PBX, Voicemail, Auto Attendant, hunt groups, and additional features of the Cisco Communications Manager Express
Audience and Prerequisites This document is intended to assist solution architects, sales engineers, field engineers, and consultants in learning many of the features of Cisco Communications Manager Express 10 running on a bundled system (CISCO SPIAD). This document assumes the reader has an architectural and istrative understanding of the CCME and has reviewed the latest istration guide. Basic knowledge of how to install and ister CCME and CUE is recommended however not necessary. All configurations are based on Command Line Interface (CLI) so is highly recommendable that you were familiar with the IOS commands. Please refer to the following links to get further information about operation of Cisco Communications Manager Express and Cisco Unity Express: Cisco CME main page: http://www.cisco.com/go/ccme Cisco CME Install Guides: http://www.cisco.com/c/en/us//unified-communications/unified-communicationsmanager-express/products-installation-guides-list.html Cisco Unity Express: http://www.cisco.com/go/cue Cisco Unity Express Maintain and Operate Guides: http://www.cisco.com/c/en/us//unified-communications/unity-express/productsmaintenance-guides-list.html Cisco IOS compatibility matrix http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/requirements/guide/33matri x.html
TASK 3: NETWORK SETUP In order to get access to the SPIAD system a previous configuration for voice and data VLAN was loaded, as well as DH pool for phones and s.
config terminal hostname CISCO_SPIAD ip dns server name spiad privilege 15 secret spiad ntp server 24.56.178.140 interface Vlan1 description DATA_VLAN no ip address interface Vlan100 description VOICE_VLAN no ip address interface GigabitEthernet 0/1 no ip address no shutdown interface GigabitEthernet0/1.1 description DATA_INTERFACE encapsulation dot1Q 1 native ip address 192.168.10.1 255.255.255.0 interface GigabitEthernet0/1.100 description VOICE_INTERFACE encapsulation dot1Q 100 ip address 10.1.1.1 255.255.255.0
interface GigabitEthernet0/0 description WAN_INTERFACE ip address dh
ip dh excluded-address 192.168.10.1 192.168.10.10 ip dh excluded-address 10.1.1.1 10.1.1.10 ip dh pool data import all network 192.168.10.0 255.255.255.0 default-router 192.168.10.1 dns-server 192.168.10.1 option 150 ip 10.1.1.1 ip dh pool phone import all network 10.1.1.0 255.255.255.0 default-router 10.1.1.1 option 150 ip 10.1.1.1 Assuming that LAN switch has been configured and both VLANs (voice and data) were assigned on designated ports, plug the interface Gigabit Ethernet 0/1 to the uplink switch trunk port. Check that your PC is connected to one of the switch ports and validate DH assignment C:\> ipconfig
Test connectivity to the default gateway assigned by SPIAD. C:\> ping 192.168.10.1
ISM Module Configuration On Cisco SPIAD Cisco Unity Express is loaded from factor in ISM 0/0, Voice Mail and Auto attendant Ports licenses are loaded also, so in order to get access to CUE IP addressing; configuration is needed, this section describes the configuration procedure. Open SSH putty client and create a new session, use the IP address 192.168.10.1 click open and accept the security alert.
When prompt appears, type spiad for and spiad for .
Load the following configuration that is needed to get Internal Service Module reachable:
config terminal ! Create a loopback interface interface Loopback0 ip address 10.1.10.2 255.255.255.252 ip virtual-reassembly in ! according to the lab topology assign the ip address 10.1.10.1/30 and default gateway 10.1.10.2 interface ISM0/0 ip unnumbered Loopback0 ip nat inside ip virtual-reassembly in service-module ip address 10.1.10.1 255.255.255.252 !Application: CUE Running on ISM service-module ip default-gateway 10.1.10.2 no shutdown ! add an static route in order to reach CUE via ISM0/0 ip route 10.1.10.1 255.255.255.255 ISM0/0
Validate that you can reach ISM IP address and its default gateway from the router pinging both IP addresses 10.1.10.1 and 10.1.10.2
Do the same tests, now form your PC.
Cisco Unity Express will be configured later in this lab at this point you have done with ISM configuration.
Internet Connection PSTN access will be provided via SIP trunk in this case internet connection is needed, this section will provide a basic configuration to gain internet access for voice, data VLAN and ISM deploying NAT.
config terminal ! Define NAT inside interfaces interface GigabitEthernet0/1.1 ip nat inside interface GigabitEthernet0/1.100 ip nat inside interface ISM0/0 ip nat inside ! Define NAT outside interfaces interface GigabitEthernet0/0 ip nat outside ! Create an access list to allow Internet access to the following source subnets access-list 1 permit 192.168.10.0 0.0.0.255 access-list 1 permit 10.1.1.0 0.0.0.255 access-list 1 permit 10.1.10.0 0.0.0.3 ! Packets received on inside interfaces and permitted on access list 1 will be translated to the IP address assigned to interface Gigabit Ethernet 0/0 ip nat inside source list 1 interface GigabitEthernet0/0 overload end Don’t forget to save your configuration copying running configuration to startup configuration.
Validate that you can get internet access, open your browser and go to: http://www.cisco.com.
Internet connection is done.
TASK 4: CONFIGURING CCME AS IPPBX SYSTEM In this section CCME will be configured as key system, we will see a couple of examples when CCME is enabled to Cisco Proprietary Skinny Client Control Protocol (SC) phones and Session Initiation Protocol (SIP) phones. The simplest model is the PBX model, in which most of the IP phones in your system have a single unique extension number. Incoming PSTN calls are routed to a receptionist at an attendant console or to an automated attendant. Phone s may be in separate offices or be geographically separated and therefore often use the telephone to each other. For this model, is recommended that you configure directory numbers as dual-lines so that each button that appears on an IP phone can handle two concurrent calls. Dual-line directory numbers enable your configuration to call waiting, call transfer with consultation, and three-party conferencing. This section contains a list of the types of files that must be ed and installed in the router flash memory to use with Cisco Unified CME. The files listed in this section are included in zipped or tar archives that are ed from the Cisco Unified CME software website at: https://software.cisco.com//type.html?mdfid=277641082 To get software a valid CCO ID is needed with and partner level privileges for such section. Now let’s continue with the
Configure the Cisco Unified Communications Manager Express Parameters The following configuration besides to start the telephony service will allow SC phones registration.
configure terminal ! Enter the command telephony−service in order to enter telephone configuration mode.
telephony−service ! Enter the command max−ephones max−num−phones in order to set the maximum number of IP phones to be ed by this platform.
max-ephones 30 ! Enter the command max−dn max−directory−numbers in order to set the maximum number of extensions that can exist in this platform.
max-dn 200
! For security reasons enter the command no auto−reg−ephone in order to prevent the connection of any phone to the system.
no auto-reg-ephone ! Enter the command load phone−type firmware−file in order to identify the firmware file that the IP phone uses to in the system.
load 7975 SC75.9-2-1S ! Set time zone for Mexico Standard/Daylight, display time format as 24 hours and set date format as dd-mm-yy
time-zone 9 time-format 24 date-format dd-mm-yy ! Assign the voice mail extension 399 according to the Lab Topology
voicemail 399 ! Define the call forward and call park behavior
call-park system application call-forward pattern .T call-forward system redirecting-expanded ! Define the call transfer behavior allowing transfer to any number and call transfer method
transfer-system full-consult dss transfer-pattern 9.T transfer-pattern .T ! Create another tone when you dial 9 to place an outside call.
secondary-dialtone 9 ! Enter the command ip source−address ip−address in order to identify the IP address and port number that the Cisco CallManager Express router uses for IP phone registration. The default port is 2000.
ip source-address 10.1.1.1 ! Set the interdigit timeout
timeouts interdigit 5
! Enter the command create cnf−files in order to build the XML configuration files.
create cnf-files end This is a generic phone template that will be used by SC phones load it.
configure terminal ephone-template 16 url services 1 http://10.1.10.1/voiceview/common/.do VoiceviewExpress softkeys remote-in-use Newcall softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback softkeys connected Hold Endcall Trnsfer TrnsfVM Confrn ConfList RmLstC Acct Park Select Don’t forget to save your configuration copying running configuration to startup configuration.
Configure the Cisco Unified Communications Manager Express to allow SIP phones registration. To enter voice global configuration mode in order to set global parameters for all ed Cisco SIP IP phones in a Cisco Unified CME or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) environment, use the voice global command in global configuration mode. To automatically remove the existing DNs, pools, and global dialplan patterns, use the no form of this command.
config terminal voice global ! Enables mode for provisioning SIP phones in Cisco Unified CME
mode cme ! Enter the command source−address ip−address in order to identify the IP address and port number that the Cisco CallManager Express router uses for IP phone registration.
source-address 10.1.1.1 port 5060 ! SIP Outbound Proxy won’t be needed for SIP phones
no outbound-proxy ! Enter the command max−pool max−num−phones in order to set the maximum number of IP phones to be ed by this platform.
max-pool 30 ! Enter the command max−dn max−directory−numbers in order to set the maximum number of extensions that can exist in this platform.
max-dn 200 ! Enter the command load phone−type firmware−file in order to identify the firmware file that the IP phone uses to in the system.
load 7841 sip78xx.10-2-1-12 ! Registration requests from SIP phones in a Cisco Unified CME system must be authenticated
authenticate
! Assign the voice mail extension 399 to the message button according to the Lab Topology
voicemail 399 ! Set a repeating audible alert notification when a call is on hold on all ed SIP phones directly connected in Cisco Unified CME
hold-alert ! Specify the directory to which the configuring files for SIP phones in Cisco Unified CME are written
tftp-path flash: ! Set time zone for Mexico Standard/Daylight, display time format as 24 hours and set date format as dd-mm-yy
timezone 9 time-format 24 date-format D/M/Y ! Enter the command create profile in order to build the XML configuration files.
create profile exit
Configure voice service voip
! Create a SIP profile in order to normalize SIP signaling
voice class sip-profiles 1000 request ANY sdp-header Connection-Info remove response ANY sdp-header Connection-Info remove ! Start voice service voip
voice service voip ! allow only trusted hosts or network
ip address trusted list ipv4 10.1.1.0 255.255.255.0 ipv4 192.168.10.0 255.255.255.0 ipv4 10.1.10.0 255.255.255.252
! Allow interworking or leg to leg communication
allow-connections allow-connections allow-connections allow-connections
h323 to h323 h323 to sip sip to h323 sip to sip
! Disable SIP redirect response for call forwarding, Disable SIP REFER message for call transfers and enable call-by-call detection of H.450 capabilities
supplementary-service h450.12 no supplementary-service sip moved-temporarily no supplementary-service sip refer ! FAX relay configuration
fax protocol none no fax-relay sg3-to-g3 sip ! enable SIP registrar functionality
registrar server expires max 3600 min 120 no update-callerid sip-profiles 1000 copy running-config startup-config
CCME is now ready to receive SC or SIP phones registration, but phone loads needs to be ed into the flash directory and define the TFTP alias for such loads, the ones needed for this lab are for Cisco Phones 7842 (SIP) and 7975 (SC).
Start TFTP server and set the root path as follows:
Phone loads compressed files for Cisco Phones are: 7841 (SIP) cmterm-78xx.10-2-1-12_REL.tar 7975 (SC) cmterm-7975-sc.9-2-1.tar
Open a SSH session to the SPIAD system if you are not logged in. And follow the next procedure to and uncompress phones firmware files. Create a directory to save the files mkdir flash:phones now create a directory for each model mkdir flash:phones/7800 mkdir flash:phones/7975 validate that your directories were created dir flash:phones
Proceed to and extract files from TFTP server archive tar /xtract tftp://192.168.10.11/cmterm-78xx.10-2-1-12_REL archive tar /xtract tftp://192.168.10.11/cmterm-7975-sc.9-2-1.tar flash:/phones/7975
Once all the files were extracted check that files are on the directories that were created previously.
Now create the TFTP alias for each file that was ed in order to make the file reachable when phone request it.
configure terminal ip tftp source-interface Loopback0 tftp-server tftp-server tftp-server tftp-server
flash:/phones/7800/kern78xx.10-2-1-12.sbn alias kern78xx.10-2-1-12.sbn flash:/phones/7800/rootfs78xx.10-2-1-12.sbn alias rootfs78xx.10-2-1-12.sbn flash:/phones/7800/sboot78xx.10-2-1-12.sbn alias sboot78xx.10-2-1-12.sbn flash:/phones/7800/sip78xx.10-2-1-12.loads alias sip78xx.10-2-1-12.loads
tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server tftp-server
flash:/phones/7975/apps75.9-2-1TH1-13.sbn alias apps75.9-2-1TH1-13.sbn flash:/phones/7975/cnu75.9-2-1TH1-13.sbn alias cnu75.9-2-1TH1-13.sbn flash:/phones/7975/cvm75sc.9-2-1TH1-13.sbn alias cvm75sc.9-2-1TH1-13.sbn flash:/phones/7975/dsp75.9-2-1TH1-13.sbn alias dsp75.9-2-1TH1-13.sbn flash:/phones/7975/jar75sc.9-2-1TH1-13.sbn alias jar75sc.9-2-1TH1-13.sbn flash:/phones/7975/SC75.9-2-1S.loads alias SC75.9-2-1S.loads flash:/phones/7975/term75.default.loads alias term75.default.loads
end copy running-config startup-config
Phone firmware is now loaded and ready to upgrade phones if it is necessary. In case you have a phone on a pre-8.3.2 firmware and can't do a direct 9.2.1. You should load an interim firmware for instance 8.4.4 on your flash and upgrade it to that load first. Then 9.1.2.
Cisco Unified Communications Manager Express Phone Provisioning In order to start to make and receive calls, you need to the specific IP phones that you want on the CCME system. In this process you set up individual ephone−dns or -dns and then associate each with a button or buttons on one or more ephones. Each ephone−dn or -dn is a virtual line, or extension, on which call connections can be made. Each physical phone must be configured as an ephone or pool in the Cisco CME router in order to receive in the LAN environment. With the use of the ephone−dn command and dual−line keyword you create an ephone−dn in dual−line mode. The reason is to have one voice port and two channels in order to handle two independent calls. This mode enables call transfer, call waiting, and conference options.
ing Cisco SC phones to CCME This procedure s ephones and ephones−dns to Cisco SPIAD:
configure terminal ! Create the ephone directory number with dual-line mode for Jim Smith, in case that phone is busy or no answer call will be forwarded to voice mail extension.
ephone-dn 1 dual-line number 201 no-reg both label 201 description Jim Smith name Jim Smith call-forward busy 399 call-forward noan 399 timeout 10 ! Create the ephone directory number with dual-line mode for Sara Noa, in case that phone is busy or no answer call will be forwarded to voice mail extension.
ephone-dn 2 dual-line number 202 no-reg both label 202 description Sara Noa name Sara Noa call-forward busy 399 call-forward noan 399 timeout 10
! Create the ephone for Jim Smith and assign the ephone-dn 1 to button 1 *replace mac-address for the one of your phone 7975 SC IP phone.
ephone 1 mac-address 0026.99EF.1DEB type 7975 ephone-template 16 name "jsmith" qwer201 button 1:1 ! Create the ephone for Sara Noa and assign the ephone-dn 2 to button 1 *replace mac-address for the one of your Cisco IP Communicator Client (CIPC)
ephone 2 mac-address AAAA.BBBB.0001 type CIPC ephone-template 16 name "snoa" qwer202 button 1:2 ! Go to telephony-service an create CNF files
telephony-service create cnf-files end copy running-config startup-config
Connect your Cisco IP Phone 7975 to the switch, if your phone was upgraded you can see on your console typing: sh phone-load Output shows something like this:
Cisco IP Communicator Client (CIPC) Configuration Sara Noa has no a physical phone assigned, instead IT deparment has installed Cisco IP communicator client which is a SC softphone that offers all features that you could find in a Cisco phone 7975, an ephone and an ephone-dn have been configured previously. IP Communicator Client is available on Cisco Software: https://software.cisco.com//release.html?mdfid=278468661&catid=280789323&s oftwareid=282074237&release=8.6(4)&relind=AVAILABLE&rellifecycle=&reltype=latest When the software is installed, run the application and perform the audio test if needed. On any place of the software interface right click and select menu preferences…
On the new window opened go to network tab, click on use this device name option, if you on the ephone 2 configuration, mac-address AAAA.BBBB.0001 was assigned to this device for Sara Noa. The device name is composed by prefix SEP followed by the mac address in this case is SEPAAAABBBB0001. TFTP server is needed also to load CIPC parameters, TFTP server will be reached on IP address 10.1.1.1.
Click OK and wait until CIPC is reloaded, when finished you will see that it is ed and extension 202 is assigned on button one to Sara Noa. Now Sara Noa should be able to call Jim Smith on extension 201, and he can call her back to extension 202. Try a call from Jim to Sara and validate that basic call between SC phones are working.
ing Cisco SIP phones to CCME This procedure s SIP phones and SIP DNs to CCME on Cisco SPIAD:
configure terminal ! Create Directory Number 203 associated to Emma Smith in case that no answer call will be forwarded to voice mail extension 399
voice dn 1 number 203 call-forward b2bua noan 399 timeout 20 name Emma Smith label Emma Smith ! Prevent that this DN attempts to to external SIP proxy
no-reg ! Create the voice pool for the cisco phone 7841 associated to Emma Smith, replace id mac with the mac address of your phone
voice pool 1 id mac 2C3E.CF86.23C0 type 7841 ! Assign voice dn 1 to button 1
number 1 dn 1 ! Set DTMF relay method
dtmf-relay rtp-nte ! set the name and for this device
name esmith qwer203 ! force codec to G711ulaw
codec g711ulaw ! go to voice global and create configuration files
voice global create profile end copy running-config startup-config Connect your 7841 and place some calls to the other extensions.
Configuring PSTN access via SIP trunk In order to gain PSTN access, in this section we will see the procedure to get a SIP Trunk ed against a simulated Internet Telephony Service Provider. This Are the requirements and SIP trunk information. Inbound SIP Trunk DID (PSTN Call in) Numbers (Use PSTN phone to test) Auto attendant (200) Extension 1 (201) Jim Smith Extension 2 (202) Sara Noa Extension 3 (203) Emma Smith VoiceMail (399)
4085xx1200 4085xx1201 4085xx1202 4085xx1203 4085xx1209
Outbound Numbers to Call (From IP Phones on SPIAD to PSTN Phone) Emergency Number Local Call Long Distance International
9060 952671800 9018182212462 90014085256800
SIP Trunk information; XX corresponds to your POD number SIP Server SIP Outbound Proxy name
4.31.34.33 4.31.34.33 4085xx1200 4085xx1200
In this case SIP Trunk requires authentication for each number, follow the next steps to get registration from ITSP
configure terminal voice service voip sip outbound-proxy ipv4:4.31.34.33:5060 sip-ua authentication name 4085011200 4085011200 credentials name 4085011200 4085011200 realm credentials name 4085011201 4085011200 realm credentials name 4085011202 4085011200 realm credentials name 4085011203 4085011200 realm no remote-party-id retry invite 2 retry 10 timers connect 100 registrar ipv4:4.31.34.33:5060 expires 300 sip-server ipv4:4.31.34.33:5060 host-registrar
4.31.34.33 4.31.34.33 4.31.34.33 4.31.34.33
end copy running start
Validate that SIP trunk was successfully ed, issue: show sip-ua status
In order to make and receive calls some translation rules and profiles are needed, follow these steps to create translation rules for incoming and outgoing calls. Allow sip server signaling IP address into the toll fraud prevention list
configure terminal voice service voip ip address trusted list ipv4 4.31.34.33 255.255.255.255 end copy running-config startup-config Outgoing Calls Setup configure terminal ! This service provider works only with codec G711u
voice class codec 1 codec preference 1 g711ulaw ! Following translation rules remove 9 prefix and translate source extensions and any number into the main DID 4085011200
voice translation-rule 410 rule 1 /^9\(.*\)/ /\1/ rule 15 /^...$/ /4085011200/
voice translation-rule 1111 rule 15 /^.*/ /4085011200/ voice translation-rule 1112 rule 1 /^9/ // ! Following profiles translate calling or called number according to translation rules defined, these profiles will be applied later on voice dial-peers
voice translation-profile CALLER_ID_TRANSLATION_PROFILE translate calling 1111 ! voice translation-profile OUTGOING_TRANSLATION_PROFILE translate called 1112 !
voice translation-profile PSTN_CallForwarding translate redirect-target 410 translate redirect-called 410 voice translation-profile PSTN_Outgoing translate calling 1111 translate called 1112 translate redirect-target 410 translate redirect-called 410
Translation rules were created, the next step is to create voip outgoing voice dial-peers, notice that session pattern includes outgoing prefix 9, session target is pointing to SIP server that was previously configured on SIP trunk section, class-codec and dtmf relay method may vary from carrier to carrier.
! dial-peer voice 1021 voip description **Local Calls - 7, 8 or 10 digits** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 9[1-9]T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip asymmetric payload full dtmf-relay rtp-nte digit-drop no vad ¡
dial-peer voice 1023 voip description ** International Calls** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 900T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! dial-peer voice 1025 voip description **National Long Distance** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 901.......... session protocol sipv2 session target sip-server voice-class codec 1 no vad ! dial-peer voice 1026 voip description **CCA*Mexico*Emergency** translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE preference 1 destination-pattern 060 session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! dial-peer voice 1027 voip description **CCA*Mexico*Emergency** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 9060 session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad end write mem
Outgoing calls should be connecting now, test the numbers provided.
Incoming Calls Setup To receive calls, voice translation rules and translation profiles are needed also; the next procedure shows how to redirect an incoming call to an specific extension or application (autoattendant).
configure terminal ! Sent DID 40850xx200 to AutoAttendant extension 200, *AA will be ready later on the lab, just configure translations
voice translation-rule 6 rule 1 /4085011200/ /200/ voice translation-profile DID_AutoAtt translate called 6 ! Sent DIDs 40850xx201 to 203 to extensions 201 to 203
voice rule rule rule
translation-rule 12 1 /4085011201/ /201/ 2 /4085011202/ /202/ 3 /4085011203/ /203/
voice translation-profile DIDs-SIP_trunk translate called 12 ! In order to get incoming calls properly routed add following voip dial-peers
dial-peer voice 1000 voip permission term description ** Incoming call from SIP trunk ** session protocol sipv2 session target sip-server incoming called-number .% voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad
dial-peer voice 3012 voip description ** DIDs from SIP_trunk ** translation-profile incoming DIDs-SIP_trunk session protocol sipv2 session target sip-server incoming called-number 408501120 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! dial-peer for incoming DIDs
dial-peer voice 3012 voip description ** DIDs from SIP_trunk ** translation-profile incoming DIDs-SIP_trunk session protocol sipv2 session target sip-server incoming called-number 408501120[1-3] voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! AutoAttendant will be enabled later but this is the dial-peer needed
dial-peer voice 3002 voip description DIDAutoAtt-AA translation-profile incoming DID_AutoAtt session protocol sipv2 session target sip-server incoming called-number 4085011200 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad
Now that you can place and receive calls your PBX is setup for basic calls in the next sections we will cover Voice Mail and Autattendant integration, additionally some advanced features will be covered.
TASK 5: DEPLOYING VOICEMAIL AND AUTOATTENDAT Cisco Unified Communications Manager Express provides call processing for Cisco Unified IP Phones. Cisco Unity Express offers voicemail and automated-attendant capabilities for IP phone s connected to Cisco Unified Communications Manager Express. The voicemail and automated-attendant capabilities are fully integrated into the Cisco access router using a network module or services-ready engine (SRE). With this solution, the Cisco portfolio of access routers delivers features similar to those of a key system or hybrid private branch exchange (PBX) plus the rich data and routing capabilities expected on the award winning Cisco integrated services routers(ISRs), Customers can now deliver unified communications to their small sites and branch offices with a solution that is very simple to deploy, ister, and maintain. Cisco Unified Communications Manager Express with Cisco Unity Express offers customers a cost-effective, highly reliable, feature-rich solution for an office deployment. As was described on Internal Service Module network setup; Cisco Unity Express is loaded from factor in ISM 0/0, Voice Mail and Auto attendant Ports licenses are loaded also on Cisco SPIAD. Cisco Unity Express GUI can read s and extension from Communications Manager Express in order to do it graphical interface for CCME should be loaded and configured, follow this procedure to GUI files from TFTP server and apply the CLI commands to enable http access and create and for this task. CCME GUI setup Open a ssh connection to the SPIAD system, and extract GUI files into flash: Issue: archive tar /xtract tftp://192.168.10.11/CME_GUI_10.5.tar flash:
Apply the following commands to create the GUI istrative and to grant http access for such . GUI name: GUI :
spiad
aaa new-model ! ! aaa authentication default local !
ip http server ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ip http path flash:/GUI ! file privilege 0 ! telephony-service web system name secret spiad dn-webedit time-webedit
Open your browser and test CCME GUI access, go to http://192.168.10.1/ccme.html Type name: and spiad
After a successful authentication you should be able to see the istrative web interface for Cisco Unified Communications Manager Express, from here you can create SC phones, SC extensions and configure some telephony-service parameters. Note: SIP phones and extensions cannot be managed on this interface.
Voice Mail Message Waiting Indicator (MWI) As we said before, Cisco Unity Express GUI reads configuration from CCME in order to have all parameters ready for CUE is necessary to create MWI configuration, follow the next steps to create indicator for ON and OFF and dial-peer to get a proper communication to CUE system.
! Create ephone-dn for ON and OFF indicators
ephone-dn 199 number A801... no-reg primary mwi off ! ! ephone-dn 200 number A800... no-reg primary mwi on
! Create voice voip dial-peer to through Inbound Calls for MWI from CUE
dial-peer voice 1005 voip description ** through Inbound Calls for MWI from CUE ** b2bua session protocol sipv2 session target ipv4:10.1.10.1 incoming called-number A80T dtmf-relay rtp-nte codec g711ulaw no vad
Cisco Unity Express Setup With the previous steps done no we are ready to complete CUE setup. Open your browser and go to http://10.1.10.1/ by default name and are “cisco” When , chose Communications Manager Express Integration.
This will trigger a system reload.
Wait for a few minutes until system goes up.
Click on Run Initialization Wizard.
Enter the name and that was created previously to get access to CCME GUI.
If you don’t see the next button click TAB several times until you see it and click on it. CUE GUI has read s and extensions for CCME and can be imported; check the mailboxes for the to create a voice mail box, and choose jsmith as , click next. Left the default values for the mailboxes and click next.
Enter following values to complete setup: Voice Mail Number: 399 Voice Mail Operator Extension: 200 Auto Attendant Access Number: 200 (if no VM box call will be redirected to AA) Auto Attendant Operator Extension: 201 (this is usually reception extension) istration via Telephone Number: 397 SIP MWI Notification Mechanism: by default outcalling MWI ON Number (Outcalling mechanism): read from ccme A800… MWI OFF Number (Outcalling mechanism):read from ccme A801…
Click next.
Review the values that you just entered and commit your setup, checking the box “Finally, save to startup configuration (will take a few minutes more)” and click finish
Once system finish initialization you will see that voicemail boxes were created and configuration was success.
Click .
Creating Mailbox for a SIP SIP s cannot be imported by CUE, so if you want to assign a voicemail box, this has to be done manually, follow the next steps to add and create voice mail box for Emma Smith (esmith) extension 203. to CUE once again, use credentials name: spiad : spiad Go to configure->s and click add. Fill the new open window with the following mandatory values ID: esmith First Name: Emma Last Name: Smith Nick Name: (automatically filled with firstname lastname) Display Name: (automatically filled with firstname lastname) Associated Phone: none Primary Extension: chose other and set it to 203
Left the rest of the parameters as default and scroll down to the button, mark the check box to create mailbox Click on the icon
located on the upper left corner to finishing addition. As mailbox creation was selected in a previous step a new window is showed:
Left default values and click on the add icon Now the new esmith was created and a mailbox was associated to it. You can validate that was added going to configure->s
By default we have created and extension that forward calls to voicemail if there is no answer. Before to test voicemail and auto attendant, go to the next section to configure VOIP routing to Cisco Unity Express.
In order to get access to voicemail and basic auto attendant from SIP or SC clients some dial peers are needed, add such dial-peers as follows:
conf t dial-peer voice 2000 voip description ** cue voicemail pilot number ** translation-profile outgoing XFER_TO_VM_PROFILE destination-pattern 399 b2bua session protocol sipv2 session target ipv4:10.1.10.1 voice-class sip outbound-proxy ipv4:10.1.10.1 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2001 voip description ** through Inbound Calls for Internal Extensions from CUE ** b2bua session protocol sipv2 session target ipv4:10.1.10.1 incoming called-number ^...$ dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2002 voip description ** cue auto attendant number ** translation-profile outgoing PSTN_CallForwarding destination-pattern 200 b2bua session protocol sipv2 session target ipv4:10.1.10.1 voice-class sip outbound-proxy ipv4:10.1.10.1 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2003 voip description ** cue prompt manager number ** translation-profile outgoing PSTN_CallForwarding destination-pattern 397 b2bua session protocol sipv2 session target ipv4:10.1.10.1 voice-class sip outbound-proxy ipv4:10.1.10.1 dtmf-relay rtp-nte codec g711ulaw no vad
Save your configuration, issue:
copy running-config startup-config
Testing Voice Mail and Auto Attendant Now you can test voicemail service dialing from extension 203 (Emma Smith) to 201 (Jim Smith), left a short message and check that message indicator is turned on once you hang up. Retrive your messages pressing the message button you will be prompt for a use 12345 Do the same test now dialing from Jim Smith (201) to Emma Smith (203). To Validate Auto Attendant Operation dial from extension 202 (Sara Noa) to extension 200 you should be able to hear Auto Attendant prompt. Once you are connected chose 1 to dial by telephone number and the dial 201 then # your call should be transferred to Jim Smith. Repeat the test but this time dial to extension 203 then # Whit these tests we have finished the Voice Mail and Basic Auto Attendant deployment, your system is fully working as an IPPBX. You can place and receive call from SIP trunk directly to a specific extension and to a basic auto attendant.
TASK 6: ADVANCED FEATURES In this section we will cover some advance features that can be implemented on Cisco SPIAD, we will cover following implementation: Hunt Groups Pickup Group Call Park Meetme Conference Call restriction by (pin before dial) Extension Mobility
APPENDIX A. Cisco Unity Express Factory Restart
CISCO_SPIAD# service-module ism0/0 session Trying 10.1.10.2, 2067 ... Open se-10-1-10-1# se-10-1-10-1# offline !!!WARNING!!!: If you are going offline to do a backup, it is recommended that you save the current running configuration using the 'write' command, prior to going to the offline state. Putting the system offline will disable management interfaces. Are you sure you want to go offline?[confirm] se-10-1-10-1(offline)# restore factory default !!!WARNING!!!: This operation will cause all configuration and data on the system to be erased. This operation is not reversible. Do you wish to continue?[confirm] Restoring the system. Please wait ... ..done System will be restored to factory default when it reloads. Press any key to reload: System reloading .... MONITOR SHUTDOWN... INIT: Sending processes the TERM signal Sending an RB message to IOS notifying module reboot... Rebooting ... — SNIP — INIT: Entering runlevel: 2 ********** rc.post_install **************** IMPORTANT:: IMPORTANT:: Welcome to Cisco Systems Service Engine IMPORTANT:: post installation configuration tool. IMPORTANT:: IMPORTANT:: This is a one time process which will guide IMPORTANT:: you through initial setup of your Service Engine. IMPORTANT:: Once run, this process will have configured IMPORTANT:: the system for your location. IMPORTANT:: IMPORTANT:: If you do not wish to continue, the system will be halted IMPORTANT:: so it can be safely removed from the router. IMPORTANT:: Do you wish to start configuration now (y,n)?
B. Cisco IP Phone Factory Reset Follow the steps below to successfully Factory reset your Cisco IP phone: 1. Unplug the power cable from the ip phone and then plug it back in. 2. While the phone is powering up, and before the Speaker button flashes on and off, press and hold the hash # key. 3. Continue to hold # until each line button (right of the LCD screen) flashes on and off in sequence in orange color. 4. Now release the hash # key and type the following sequence 123456789*0# After the sequence has been entered the line buttons on the phone flash orange, then green and the phone goes through the factory reset process. This process can take several minutes and the firmware of the IP Phone will be erased. When complete, the IP phone will reboot and the bootloader will try to obtain an IP address via DH. The IP phone also expects the IP address (option 150) or the name (option 66) of the TFTP server to be delivered by the DH server. This is why these DH options are critical at this phase. The phone then tries to obtain the appropriate termXX.default.loads file depending in its model: This "loads" file indicates all the files the IP phone has to from the TFTP server to make up the device firmware. The IP phone should first obtain the “loads” file and then proceed with the individual files. Once complete, the IP phone will install the files and finally reboot C. How to Reset Cisco IP Communicator Follow the steps below to successfully clear settings on Cisco IP Communicator Client: Press Settings Button (looks like a check in a box)
Press**# (to unlock menu) Press Erase
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