Sometimes you may want to get back to UCCX initial page. You can do this by the following step.
1. From the UCCX server, Start> Run and type cet. When you get the warning select No.
2. Select com.cisco.crs.cluster.config.AppAdminSetupConfig, then double click in Row 0 and select com.cisco.crs.cluster.config.AppAdminSetupConfig tab and change Setup State from "Done" to "FRESH_INSTALL"
Sunday, July 25, 2010
UCCX Initial Configuration
This article is based on UCCX7.0.
After you have installed UCCX, the first thing to do is initial the configuration.
1. Use browser to access http://{UCCX IP Address}/appadmin.
The default login is Administrator, ciscocisco
2. Since this is standalone server I'll use Single Node.
3. Enter CM IP Address that has enabled AXL Service (Normally is Publisher) and user account that has Standard AXL API Access Roles in Callmanager.
4. Upload your license files.
5. Wait for System Components Activation and click next.
6. Enter user and password for Telephony Subsystem and RmCm Subsystem. These users will be created in Callmanager Application User automatically.
Note:
6.1 User in Telephony Subsystem is used to control CTI Ports and CTI Route Point in Callmanager.
6.2 User in RmCm Subsystem is used to control Agent Phones.
7. Enter System Parameters Configuration.
Number of HR session licenses = The maximum sessions that HR client can connect to server
Recording Count = The maximum concurrent on demand recording.
Number of Outbound seats = The number of outbound agent.
Codec = Codec that will be used in UCCX.
8. Select language you want to use (can choose more than 1) and check Country Specific if you want to customize your system audios for that language.
9. Select end user(Sync from Callmanager) to be used as an administrator in UCCX.
10. See Cisco Unified CCX Setup Result Information if these is any error.
After you have installed UCCX, the first thing to do is initial the configuration.
1. Use browser to access http://{UCCX IP Address}/appadmin.
The default login is Administrator, ciscocisco
2. Since this is standalone server I'll use Single Node.
3. Enter CM IP Address that has enabled AXL Service (Normally is Publisher) and user account that has Standard AXL API Access Roles in Callmanager.
4. Upload your license files.
5. Wait for System Components Activation and click next.
6. Enter user and password for Telephony Subsystem and RmCm Subsystem. These users will be created in Callmanager Application User automatically.
Note:
6.1 User in Telephony Subsystem is used to control CTI Ports and CTI Route Point in Callmanager.
6.2 User in RmCm Subsystem is used to control Agent Phones.
7. Enter System Parameters Configuration.
Number of HR session licenses = The maximum sessions that HR client can connect to server
Recording Count = The maximum concurrent on demand recording.
Number of Outbound seats = The number of outbound agent.
Codec = Codec that will be used in UCCX.
8. Select language you want to use (can choose more than 1) and check Country Specific if you want to customize your system audios for that language.
9. Select end user(Sync from Callmanager) to be used as an administrator in UCCX.
10. See Cisco Unified CCX Setup Result Information if these is any error.
Call Detail Records (CDR)
Call Detail Records (CDR) logs call activity in the system and can be sent out or be queried to/from billing application.
Before you begin make sure the following services have been activated.
1. Cisco CAR Web Service ---> For access CDR details in Publisher web page.
2. Cisco SOAP - CDRonDemand Service ---> For billing application that query CDR directly from Publisher
Enable Service Parameters
1. Enable CDR in Callmanager Service Parameters (Enable all server in the cluster).
2. If you want to include FAC in CDR, set Display FAC in CDR = True.
To access CDR details in Publisher web page use the following URL.
https://{Publisher IP Address}:8443/car/Logon.jsp
User has to be assigned in Standard CAR Admin Users Group to access this page.
To send CDR details to external server using FTP/SFTP.
1. In Cisco Unified Serviceability Page, Select Tools> CDR Management
2. In Billing Application Server Parameters, click Add new
3. Enter IP Address of FTP/SFTP server, account to login that server and Directory Path.
4. If you've got the error message, make sure that server already start FTP/SFTP service.
Before you begin make sure the following services have been activated.
1. Cisco CAR Web Service ---> For access CDR details in Publisher web page.
2. Cisco SOAP - CDRonDemand Service ---> For billing application that query CDR directly from Publisher
Enable Service Parameters
1. Enable CDR in Callmanager Service Parameters (Enable all server in the cluster).
2. If you want to include FAC in CDR, set Display FAC in CDR = True.
To access CDR details in Publisher web page use the following URL.
https://{Publisher IP Address}:8443/car/Logon.jsp
User has to be assigned in Standard CAR Admin Users Group to access this page.
To send CDR details to external server using FTP/SFTP.
1. In Cisco Unified Serviceability Page, Select Tools> CDR Management
2. In Billing Application Server Parameters, click Add new
3. Enter IP Address of FTP/SFTP server, account to login that server and Directory Path.
4. If you've got the error message, make sure that server already start FTP/SFTP service.
csim start command
csim start {number} is a hidden command to initiate simulated calls. It allows debugging for call routing from voice gateways and CUCME.
When command was issued it will looks into dial-peer configured on the router and send call to destination-pattern configured in dial-peer.
Note that this command also bypass COR configured in dial-peer.
Router#csim start 7101 ---> this case dial-peer was found and success to call
Router#csim start 8001 ---> this case dial-peer was not found
When command was issued it will looks into dial-peer configured on the router and send call to destination-pattern configured in dial-peer.
Note that this command also bypass COR configured in dial-peer.
Router#csim start 7101 ---> this case dial-peer was found and success to call
csim: called number = 7101, loop count = 1 ping count = 0
csim err csimDisconnected recvd DISC cid(4885)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0
Router#csim start 8001 ---> this case dial-peer was not found
csim: called number = 8001, loop count = 1 ping count = 0
csim err:csim_do_test Error peer not found
Remote SPAN (RSPAN) for Voice Recorder
Some Voice Recorders require the copy of signal and RTP from IP Phone to theirs interfaces.
This example will show you how to configure remote span session to send the copy of signal and RTP from IP Phone to Voice Recording Server.
Access Switch Configuration
Access-Switch(config)#vlan 999
Access-Switch(config-vlan)#remote-span
Access-Switch(config-vlan)#exit
Access-Switch(config)#monitor session 1 source vlan 100
Access-Switch(config)#monitor session 1 destination remote vlan 999
Access-Switch(config)#int gi0/1
Access-Switch(config-if)#switchport mode trunk
Core Switch Configuration
Core-Switch(config)#vlan 999
Core-Switch(config-vlan)#remote-span
Core-Switch(config-vlan)#exit
Core-Switch(config)#monitor session 1 source remote vlan 999
Core-Switch(config)#monitor session 1 destination interface gi0/1
Core-Switch(config)#int gi0/3
Core-Switch(config-if)#switchport mode trunk
This example will show you how to configure remote span session to send the copy of signal and RTP from IP Phone to Voice Recording Server.
Access Switch Configuration
Access-Switch(config)#vlan 999
Access-Switch(config-vlan)#remote-span
Access-Switch(config-vlan)#exit
Access-Switch(config)#monitor session 1 source vlan 100
Access-Switch(config)#monitor session 1 destination remote vlan 999
Access-Switch(config)#int gi0/1
Access-Switch(config-if)#switchport mode trunk
Core Switch Configuration
Core-Switch(config)#vlan 999
Core-Switch(config-vlan)#remote-span
Core-Switch(config-vlan)#exit
Core-Switch(config)#monitor session 1 source remote vlan 999
Core-Switch(config)#monitor session 1 destination interface gi0/1
Core-Switch(config)#int gi0/3
Core-Switch(config-if)#switchport mode trunk
Voice Vlan and Auto QoS
Implementation of IP Telephony, voice vlan should be used to separate data and voice traffic.
Auto QoS Advantage.
1. Easy to implement.
2. Follow Cisco Best Practices for QoS.
This will show you the example configuration for voice vlan and QoS in switch.
Switch(config)#int fa0/1
Switch(config-if)#switchport mode access
Switch(config-if)#switchport access vlan 200
Switch(config-if)#switchport voice vlan 100
Switch(config-if)#spanning-tree portfast
Switch(config-if)#auto qos voip cisco-phone ---> this will generate many commands
Note:
1. CDP need to be enabled on switch to allow switch to detect Cisco IP Phones.
2. For switch trunk port use "auto qos voip trust"
Auto QoS Advantage.
1. Easy to implement.
2. Follow Cisco Best Practices for QoS.
This will show you the example configuration for voice vlan and QoS in switch.
Switch(config)#int fa0/1
Switch(config-if)#switchport mode access
Switch(config-if)#switchport access vlan 200
Switch(config-if)#switchport voice vlan 100
Switch(config-if)#spanning-tree portfast
Switch(config-if)#auto qos voip cisco-phone ---> this will generate many commands
Note:
1. CDP need to be enabled on switch to allow switch to detect Cisco IP Phones.
2. For switch trunk port use "auto qos voip trust"
Saturday, July 24, 2010
Understanding Dial Peer
Dial peers are the primary used to implement a dial plan on Cisco voice gateways. Dial peers are used to determine which calls are routed and the path a call should take.
POTS dial peers are used to process calls that are connected to traditional telephony equipment such as PBX, Analog Phone and PSTN.
VOIP dial peers are used to process calls to or from the data network.
Pots Dial Peers
Configuration
Router(config)#dial-peer voice 1 pots
Router(config-dial-peer)#destination-pattern 2222
Router(config-dial-peer)#port 0/0/0
VOIP Dial Peers
Configuration
Router(config)#dial-peer voice 100 voip
Router(config-dial-peer)#destination-pattern 3333
Router(config-dial-peer)#session-target ipv4:192.168.1.1
Dial Peer Matching
Inbound and Outbound Dial Peers.
Inbound Dial Peer Matching
Priority1 "incoming called-number" ---> Defines the called number or DNIS string
Priority2 "answer-address" ---> Defines the originating calling number or ANI string
Priority3 "destination-pattern" ---> Uses the calling number (originating or ANI string) to match the incoming call leg to an inbound dial peer.
Priority4 "port" ---> Attempts to match the configured dial-peer port to the voice port associated with the incoming call (POTS dial peers only).
Priority5 "default" dial-peer or dial peer 0, is matched when no other inbound dial peer is used. It is better to configure a specific dial peer to match incoming calls.
Outbound Dial Peer Matching
For outbound dial peer matching, the called number is matched against the configured destination-pattern in the dial peers with the longest matched.
Consider the following scenario.
R1 has 4 dial-peers configuration, 2 pots and 2 voips.
When phone 2222 call 3333,
dial-peer 2 pots will match as an inbound dial-peer because "incoming called-number" is the 1st priority when matching inbound dial-peer and the call will be processed as one-stage dialing(you will not get the second dial tone to enter additional digits).
dial-peer 101 voip will match as an outgoing dial-peer because 3333 is the longest matched and codec g.729r8 will be used for RTP media stream over ip network.
POTS dial peers are used to process calls that are connected to traditional telephony equipment such as PBX, Analog Phone and PSTN.
VOIP dial peers are used to process calls to or from the data network.
Pots Dial Peers
Configuration
Router(config)#dial-peer voice 1 pots
Router(config-dial-peer)#destination-pattern 2222
Router(config-dial-peer)#port 0/0/0
VOIP Dial Peers
Configuration
Router(config)#dial-peer voice 100 voip
Router(config-dial-peer)#destination-pattern 3333
Router(config-dial-peer)#session-target ipv4:192.168.1.1
Dial Peer Matching
Inbound and Outbound Dial Peers.
Inbound Dial Peer Matching
Priority1 "incoming called-number" ---> Defines the called number or DNIS string
Priority2 "answer-address" ---> Defines the originating calling number or ANI string
Priority3 "destination-pattern" ---> Uses the calling number (originating or ANI string) to match the incoming call leg to an inbound dial peer.
Priority4 "port" ---> Attempts to match the configured dial-peer port to the voice port associated with the incoming call (POTS dial peers only).
Priority5 "default" dial-peer or dial peer 0, is matched when no other inbound dial peer is used. It is better to configure a specific dial peer to match incoming calls.
Outbound Dial Peer Matching
For outbound dial peer matching, the called number is matched against the configured destination-pattern in the dial peers with the longest matched.
Consider the following scenario.
R1 has 4 dial-peers configuration, 2 pots and 2 voips.
When phone 2222 call 3333,
dial-peer 2 pots will match as an inbound dial-peer because "incoming called-number" is the 1st priority when matching inbound dial-peer and the call will be processed as one-stage dialing(you will not get the second dial tone to enter additional digits).
dial-peer 101 voip will match as an outgoing dial-peer because 3333 is the longest matched and codec g.729r8 will be used for RTP media stream over ip network.
ISDN PRI Gateway Configuration
Environment
1. Cisco Router 3825 with VWIC2-1MFT-T1/E1 card in router HWIC0 slot.
2. PVDM2-32
3. Link to PSTN with ISDN PRI E1 EURO Protocol.
Router(config)#card type e1 0 0
Router(config)#network-clock-participate wic 0
Router(config)#network-clock-select 1 E1 0/0/0
Router(config)#isdn switch-type primary-net5
Router(config)#controller E1 0/0/0
Router(config-controller)#pri-group timeslots 1-31
Router(config-controller)#interface serial 0/0/0:15
Router(config-if)#isdn switch-type primary-net5
Router(config-if)#isdn incoming-voice voice
Router(config-if)#end
Router#show isdn status
Global ISDN Switchtype = primary-net5
ISDN Serial0/0/0:15 interface
dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0xFFFF7FFF
Number of L2 Discards = 0, L2 Session ID = 1
Now use dial-peer to create the numbering plan.
1. Cisco Router 3825 with VWIC2-1MFT-T1/E1 card in router HWIC0 slot.
2. PVDM2-32
3. Link to PSTN with ISDN PRI E1 EURO Protocol.
Router(config)#card type e1 0 0
Router(config)#network-clock-participate wic 0
Router(config)#network-clock-select 1 E1 0/0/0
Router(config)#isdn switch-type primary-net5
Router(config)#controller E1 0/0/0
Router(config-controller)#pri-group timeslots 1-31
Router(config-controller)#interface serial 0/0/0:15
Router(config-if)#isdn switch-type primary-net5
Router(config-if)#isdn incoming-voice voice
Router(config-if)#end
Router#show isdn status
Global ISDN Switchtype = primary-net5
ISDN Serial0/0/0:15 interface
dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0xFFFF7FFF
Number of L2 Discards = 0, L2 Session ID = 1
Now use dial-peer to create the numbering plan.
MGCP Gateway Auto Configuration
MGCP was created for a centralized architecture, where most of the configuration and call-control intelligence resides on a call agent, such as Cisco Callmanager.
Configuration on IOS Router,
MGCP-Gateway(config)#ccm-manager mgcp
MGCP-Gateway(config)#ccm-manager music-on-hold
MGCP-Gateway(config)#ccm-manager config server
MGCP-Gateway(config)#ccm-manager config
---> After this command Gateway will download configuration from Callmanager.
Configuration Callmanager
Note:
1. Domain Name must be the same with router hostname.
2. If you have command "ip domain-name" in router configuration for example "ip domain-name uclab.com" in Callmanager Gateway Configuration Page use Domain Name = MGCP-Gateway.uclab.com
3. If you have command "no ip domain-name" in router configuration just use MGCP-Gateway in Callmanager Gateway Configuration Page.
Configuration on IOS Router,
MGCP-Gateway(config)#ccm-manager mgcp
MGCP-Gateway(config)#ccm-manager music-on-hold
MGCP-Gateway(config)#ccm-manager config server
MGCP-Gateway(config)#ccm-manager config
---> After this command Gateway will download configuration from Callmanager.
Configuration Callmanager
Note:
1. Domain Name must be the same with router hostname.
2. If you have command "ip domain-name" in router configuration for example "ip domain-name uclab.com" in Callmanager Gateway Configuration Page use Domain Name = MGCP-Gateway.uclab.com
3. If you have command "no ip domain-name" in router configuration just use MGCP-Gateway in Callmanager Gateway Configuration Page.
Friday, July 23, 2010
Local Route Groups
Local Route Groups(LRG) feature reduce the complexity of the numbering plan when you have a large number of locations.
Consider the following scenario.
In the picture, we have Callmanager at HQ and phone from each branch register to Callmanager.
Each branch has their own PSTN gateway, so in Route Pattern configuration we need 5 Route Patterns with 5 Partitions and phones from each branch have CSS to access their own PT and then to their own gateway.
With Local Route Group we can simplify this route plan. Consider the following picture
As you can see, each branch has their own Device Pool associated with their own LRG and then their own gateway.
With this configuration, we achieve the same result with only 1 Route Pattern.
When a phone initiate a call to outside and then match Route Pattern --> Route List
The Route List contain special route group that is named Standard Local Route Group.
At this point system will check the phone's Device Pool and see which Route Group associated.
Then call will be sent out to gateway associated to that Route Group.
Detail Configuration,
Step1. Create 5 Route Groups and associate gateways to Route Groups.
Step2. Create 5 Device Pools and associate Local Route Group in Device Pool with Route Group created in Step1.
Step3. Create Route List and associate Standard Local Route Group to this Route List.
Step4. Create Route Pattern and associate Route List to this Route Pattern.
Step5. Associate each Device Pool to phones.
Step6. If call sending out to each local gateway require different CLID, we need to use Calling Party Transformation Patterns to manipulate Calling Number for each barnch.
For example BR1 with DID number 021267XXX and extension number 7XXX.
Step6.1 Create Calling Party Transformation Patterns (Partition is required)
Step6.2 Create CSS that can access Calling Party Transformation Patterns Partition in Step6.1 in this case CSS_BR1_CLID
Step6.3 Associate CSS_BR1_CLID to Calling Party Transformation CSS in Device Pool or Gateway.
If you associate CSS_BR1_CLID to Gateway make sure to uncheck Use Device Pool Calling Party Transformation CSS.
Consider the following scenario.
In the picture, we have Callmanager at HQ and phone from each branch register to Callmanager.
Each branch has their own PSTN gateway, so in Route Pattern configuration we need 5 Route Patterns with 5 Partitions and phones from each branch have CSS to access their own PT and then to their own gateway.
With Local Route Group we can simplify this route plan. Consider the following picture
As you can see, each branch has their own Device Pool associated with their own LRG and then their own gateway.
With this configuration, we achieve the same result with only 1 Route Pattern.
When a phone initiate a call to outside and then match Route Pattern --> Route List
The Route List contain special route group that is named Standard Local Route Group.
At this point system will check the phone's Device Pool and see which Route Group associated.
Then call will be sent out to gateway associated to that Route Group.
Detail Configuration,
Step1. Create 5 Route Groups and associate gateways to Route Groups.
Step2. Create 5 Device Pools and associate Local Route Group in Device Pool with Route Group created in Step1.
Step3. Create Route List and associate Standard Local Route Group to this Route List.
Step4. Create Route Pattern and associate Route List to this Route Pattern.
Step5. Associate each Device Pool to phones.
Step6. If call sending out to each local gateway require different CLID, we need to use Calling Party Transformation Patterns to manipulate Calling Number for each barnch.
For example BR1 with DID number 021267XXX and extension number 7XXX.
Step6.1 Create Calling Party Transformation Patterns (Partition is required)
Step6.2 Create CSS that can access Calling Party Transformation Patterns Partition in Step6.1 in this case CSS_BR1_CLID
Step6.3 Associate CSS_BR1_CLID to Calling Party Transformation CSS in Device Pool or Gateway.
If you associate CSS_BR1_CLID to Gateway make sure to uncheck Use Device Pool Calling Party Transformation CSS.
Thursday, July 22, 2010
Single Number reach(SNR) and Enterprise Feature Access(EFA)
SNR associates IP Phone number with the destination number(Up to 10) such as mobile phone. Incoming calls to a user IP Phone also ring to mobile phone at the same time.
With EFA user can originate calls from the remote destination phone through the Callmanager and instead of sending the mobile number to the called party, the user enterprise number gets sent to the called party.
Users can also use EFA to enable or disable SNR (Mobile connect).
Example configuration.
In this example incoming call to IP Phone DN 1001 will also ring to mobile phone number 0891111111.
Note:
1. In this example Route Pattern for mobile number will be 9.08XXXXXXXX with pre-dot discard digit.
2. CLID from PSTN to Callmanager is 0891111111.
3. You'll see in the example that application dial rule need to be used to make both SNR and EFA feature working.
Before you begin make sure that Cisco Unified Mobile Voice Access Service already activated.
Step1. Create User Account(user1001 in this example), associate physical phone or EM profile to this user, check Enable Mobility and assign Standard CCM End User Role.
Step2. Enable EFA in Service Parameter.
Step3. Define EFA Number in Call Routing> Mobility Configuration.
Step4. Create Remote Destination Profile, Select Device> Device Settings> Remote Destination Profile and Add New.
For Calling Search Space enter CSS that can access EFA number Partition in Step3.
For Rerouting Calling Search Space enter CSS that can access PSTN Mobile Call.
Step5. Associate DN 1001 to this RDP, Note that this DN already associated with physical phone.
Step6. Add a New Remote Destination in RDP.
In Destination Number enter mobile number in this case 0891111111. This number must match CLID from PSTN to Callmanager, otherwise EFA will fail.
Delay Before Ringing Timer is the delay time before ring to mobile phone after IP Phone ring.
Define Schedule for this Remote Destination such as you may want to enable SNR only on weekends.
When receiving a call during the above ring schedule is the list of caller that you want to send call to mobile phone(Need to create Access List in Call Routing> Class of Control> Access List)
Save and check Line Associate in Association Information(Important!!!)
Step7. Go to Phone Configuration, assign softkey template that has Mobility softkey and assign Owner User ID(For EM assign this user to Device Profile).
Step8. Since we have to use 9 for access code but in Remote Destination has only 0891111111(why? see Step6 for reason to not include access code). So we need to use Application Dial Rule to help.
Select Call Routing> Dial Rule> Application Dial Rules and Add New.
With this rule system will prepend the digit 9 to number begin with 08 and digit length=10.
Step9. Test call to number 1001, call should be ring both IP Phone and mobile phone.
If you answer call from IP Phone, press Mobility Softkey and select send call to mobile phone, call should be sent to mobile phone.
If you answer call from mobile phone, IP Phone line status will show busy(red) and when you drop the call from mobile phone system still hold the line for a few second before disconnect the call so during this time you can pick up the call from IP Phone.
When you use mobile phone(Same number in Remote Destination) call in to EFA number and press PIN#1#Number to Call# this will use Callmanager to call out to destination and CLID will be user enterprise number(IP Phone) not mobile number.
With EFA user can originate calls from the remote destination phone through the Callmanager and instead of sending the mobile number to the called party, the user enterprise number gets sent to the called party.
Users can also use EFA to enable or disable SNR (Mobile connect).
Example configuration.
In this example incoming call to IP Phone DN 1001 will also ring to mobile phone number 0891111111.
Note:
1. In this example Route Pattern for mobile number will be 9.08XXXXXXXX with pre-dot discard digit.
2. CLID from PSTN to Callmanager is 0891111111.
3. You'll see in the example that application dial rule need to be used to make both SNR and EFA feature working.
Before you begin make sure that Cisco Unified Mobile Voice Access Service already activated.
Step1. Create User Account(user1001 in this example), associate physical phone or EM profile to this user, check Enable Mobility and assign Standard CCM End User Role.
Step2. Enable EFA in Service Parameter.
Step3. Define EFA Number in Call Routing> Mobility Configuration.
Step4. Create Remote Destination Profile, Select Device> Device Settings> Remote Destination Profile and Add New.
For Calling Search Space enter CSS that can access EFA number Partition in Step3.
For Rerouting Calling Search Space enter CSS that can access PSTN Mobile Call.
Step5. Associate DN 1001 to this RDP, Note that this DN already associated with physical phone.
Step6. Add a New Remote Destination in RDP.
In Destination Number enter mobile number in this case 0891111111. This number must match CLID from PSTN to Callmanager, otherwise EFA will fail.
Delay Before Ringing Timer is the delay time before ring to mobile phone after IP Phone ring.
Define Schedule for this Remote Destination such as you may want to enable SNR only on weekends.
When receiving a call during the above ring schedule is the list of caller that you want to send call to mobile phone(Need to create Access List in Call Routing> Class of Control> Access List)
Save and check Line Associate in Association Information(Important!!!)
Step7. Go to Phone Configuration, assign softkey template that has Mobility softkey and assign Owner User ID(For EM assign this user to Device Profile).
Step8. Since we have to use 9 for access code but in Remote Destination has only 0891111111(why? see Step6 for reason to not include access code). So we need to use Application Dial Rule to help.
Select Call Routing> Dial Rule> Application Dial Rules and Add New.
With this rule system will prepend the digit 9 to number begin with 08 and digit length=10.
Step9. Test call to number 1001, call should be ring both IP Phone and mobile phone.
If you answer call from IP Phone, press Mobility Softkey and select send call to mobile phone, call should be sent to mobile phone.
If you answer call from mobile phone, IP Phone line status will show busy(red) and when you drop the call from mobile phone system still hold the line for a few second before disconnect the call so during this time you can pick up the call from IP Phone.
When you use mobile phone(Same number in Remote Destination) call in to EFA number and press PIN#1#Number to Call# this will use Callmanager to call out to destination and CLID will be user enterprise number(IP Phone) not mobile number.
Wednesday, July 21, 2010
Extension Mobility
Cisco Extension Mobility(EM) allow users login to other phones and get their profiles such as DN, Services, speed dial. So users can use any phone in the network just like their own phones.
Configuration
Before you begin make sure that EM service already enabled in Serviceability Page.
1. Create Phone Service, Select Device> Device Settings> Phone Services and Add New
In Service URL enter
http://{EM Server}:8080/emapp/EMAppServlet?device=#DEVICENAME#
2. Create Device Profile, Select Device> Device Settings> Device Profile and Add New enter your device type and protocol.
Enter the other option like Softkey Template, Add DN, etc.. just like you're configuring Phone (Imagine that your Device Profile is your Physical Phone that will be loaded when you logged in).
3. Subscribe EM Service that you created in step1 to Device Profile. At Related Links select Subscribe/Unsubscribe Services, then click Go
4. Create User for login, Select User Management> End User and Add New.
Enter User ID, Password, PIN, Lastname
In Extension Mobility Section select your device profile created in step2 to Controlled Profiles
Click Save, then assisgn role to this user
5. Go to phone configuration, enable EM and subscribe EM service to physical phone
6. Go back to Device Profile Configuration and associate user to Device Profile.
7. Test login, Press Service button on phone enter User ID, PIN.
8. In Phone Configuration you'll see the details for logged in user.
9. You may want to tuning some EM Paramter. Go to Service Parameter and see.
Configuration
Before you begin make sure that EM service already enabled in Serviceability Page.
1. Create Phone Service, Select Device> Device Settings> Phone Services and Add New
In Service URL enter
http://
2. Create Device Profile, Select Device> Device Settings> Device Profile and Add New enter your device type and protocol.
Enter the other option like Softkey Template, Add DN, etc.. just like you're configuring Phone (Imagine that your Device Profile is your Physical Phone that will be loaded when you logged in).
3. Subscribe EM Service that you created in step1 to Device Profile. At Related Links select Subscribe/Unsubscribe Services, then click Go
4. Create User for login, Select User Management> End User and Add New.
Enter User ID, Password, PIN, Lastname
In Extension Mobility Section select your device profile created in step2 to Controlled Profiles
Click Save, then assisgn role to this user
5. Go to phone configuration, enable EM and subscribe EM service to physical phone
6. Go back to Device Profile Configuration and associate user to Device Profile.
7. Test login, Press Service button on phone enter User ID, PIN.
8. In Phone Configuration you'll see the details for logged in user.
9. You may want to tuning some EM Paramter. Go to Service Parameter and see.
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