Extension Mobility made easy - 20 minutes

Cisco Extension Mobility allows users to temporarily access their Cisco IP Phone configuration such as line appearances, services, and speed dials from other Cisco IP Phones.

1. Add new phone service , name it "Extention Mobility" - Device/Device Setting/Phone Services
 Use following URL for ext-mobility : http://yourCUCM_ip_addr:8080/emapp/EMAppServlet?device=#DEVICENAME#  . You can get this URL from the CUCM help menu

2. Add new "Device Profile" for your IP phone , let 's name it "EM-7970" under menu Device/Device Setting/Device Profile . This device profile is dependent on the type of IP phone you have , ie. 7970 , 7960 ...  So if you have 2 different types of IP phones, you need to create 2 separate Device profiles, i.e EM-7970, EM-7960 ... Add DN to this Device-Profile, for ex: DN=1002, PT = PT-internal. Also need to subscribe this "Device Profile" to Ex-Mo service (otherwise user cannot log out of Ex-Mo).

3. Configure end-user - under menu User Management/End Users, add new user or open an existing user, select Extension Mobility Available Profile "EM-7970" (added in step 2). Make sure the usual end-user groups (CTI enabled, CCM user ...) are configured, "allow control of device from CTI" BOX is checked and user is associated to a Phone (Controlled Device).

4. Configure IP phone - under menu Device / Phone, select a phone.  Checked on BOX labeled "Enabled Extention Mobility" and select Log out profile to "EM-7970" (configured in step 2).  Subscribe phone to "Extention Mobility" service (configured in step 1).  Reset phone as needed.

5. To test EM, on a same type IP phone , i.e. 7970, select service button. Phone should prompt with UserID and PIN # to log in. Once user is successfully logged in, phone will reset with correct softkey and button template.

ISDN L2 on HQ did not came up after configure MGCP

On this particular instance whereas after I configured ISDN MGCP on the HQ router and ISDN layer2 did not came up. After checking with the obvious CLI such as "show mgcp endpoint , show mgcp status , show ccm-manager host"  and realize everything looked good. Decided to reload the router and that fixed the problem
              
HQ#sh isdn sta
Global ISDN Switchtype = primary-ni
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply
ISDN Serial1/0:23 interface
        dsl 0, interface ISDN Switchtype = primary-ni
        L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED

HQ#sh mgcp end       
Interface T1 1/0
             ENDPOINT-NAME    V-PORT     SIG-TYPE   ADMIN
             S1/ds1-0/1@HQ     1/0:23        none   up   
             S1/ds1-0/2@HQ     1/0:23        none   up   
             S1/ds1-0/3@HQ     1/0:23        none   up   
Interface T1 1/1
             ENDPOINT-NAME    V-PORT     SIG-TYPE   ADMIN
HQ#sh mgcp stat
  .........
 IP address based Call Agents statistics:
 IP address 10.1.200.21, Total msg rx 77,
                  successful 68, failed 9
      ................................ 
HQ#sh ccm-manager hosts 
MGCP Domain Name: HQ
Priority        Status                   Host
============================================================
Primary         Registered               10.1.200.21
First Backup    None                    
Second Backup   None                    

Call-manager-fallback config for SRST and non-SRST modes

For H323 GW interworking with CUCM cluster, User needs to config dial-peers in order to handle in/out calls from PSTN. For ex:

dial-pee voice 1 pots    ==== to handle incoming calls from PSTN
    incoming called .
    direct-inward
    port 1/0:23
dial-p voice 3120 voip    ==== to forward calls to CUCM
    destination-pat 1312301....$
    session target ipv4:10.1.200.21
    pref 1
dial-p v 900 pots
    desc ==== local  ====    note that this dial-peer does NOT have prefix 9 because CM strips digit 9
    destination-pat [2-9]......$
    port 1/0:23

In SRST mode, any local call with 7 digit would fail if call-manager-fallback is configured with secondary-dialtone 9

To get around this issue configure voice translation-rule to strip out digit 9 when dial-out of PSTN

voice translation-rule 100
        rule 1 /^9\(.*\)$/    /\1/        <<<  strip any string of digit beginning with 9
    voice translation-profile SRST-LOCAL
        translate called 100
call-manager-fallback
    translation-profile incoming SRST-LOCAL

Also, add translation-rule to trim incoming 10 digits to 4 digit
voice translation-rule 1
   rule 1 /^312301\(....\)$/   /\1/   ==== strip out all digits except last 4
voice translation-profile DID
   translate called 1
dial-peer voice 1 pots
   translation-profile incoming DID
   incoming called-number .
   direct-inward-dial
    port 1/0:23

In case SRST still did not work without adding digit 9 in the POTS dial-peer, add CLI  "access-code pri 9 direct-inward-dial" into call-manager-fallback config. Below is full config for call-manager-fallback:

voice translation-rule 1
 rule 1 /^1312301\(....\)/ /\1/
 rule 2 /^312301\(....\)/ /\1/
!
voice translation-profile DID
 translate called 1
!
application
 service alternate default
 !
 global
  service alternate default
 !
dial-peer voice 1 pots
 translation-profile incoming DID
 incoming called-number .
 direct-inward-dial
 port 1/0:23
!
dial-peer voice 911 pots
 description ==== 911
 destination-pattern 911
 port 1/0:23
 forward-digits 3
!
dial-peer voice 1000 pots
 description ==== local
 destination-pattern [2-9]......$
 port 1/0:23
!
dial-peer voice 1001 pots
 description ==== LD
 destination-pattern 1[2-9]..[2-9]......$
 port 1/0:23
 prefix 1
!
dial-peer voice 1002 pots
 description ==== INTL
 destination-pattern 011.T
 port 1/0:23
 prefix 011
!
dial-peer voice 2 voip
 description ==== to CUCM pub
 preference 1
 destination-pattern 312301....$
 session target ipv4:10.1.200.21
!
dial-peer voice 3 voip
 description ==== to CUCM sub
 preference 2
 destination-pattern 312301....$
 session target ipv4:10.1.200.20
!        
!
gateway
 timer receive-rtp 1200
!
!
call-manager-fallback
 secondary-dialtone 9
 max-conferences 12 gain -6
 transfer-system full-consult
 ip source-address 10.10.32.2 port 2000
 max-ephones 2
 max-dn 4
 system message primary SRST@BR1
 dialplan-pattern 1 3123012... extension-length 4
 transfer-pattern .T
 access-code pri 9 direct-inward-dial
 voicemail 917752011500
 call-forward pattern .T
 call-forward busy 917752011500
 call-forward noan 917752011500 timeout 10
!

Single Number Reach - aka SNR in 5 minutes

SNR - single number reach in 5 minutes

1. Config End-user hquser3, choose "Enable Mobility" and associate user with device (phone3)
2. On Menu device/device setting/softkey template, copy standard user template to Mobile user template. Click on "configure softkey layout" link. Add Mobility softkey to on-hook and connected states.
3. Config hqphone3 with Mobility softkey-template, and turn "Device Mobility Mode" ON
4. Add new device/device setting/remote destination profile - ex: hquser3.
        - userid = hquser3
        - DP = DP-HQ
        - CSS = CSS-global
5. Add new DN 1003 on remote destination profile ,
        - PT = PT-USA  (same PT as hqphone3/Line1)
    then save
6. Add Associated Remote Destinations
        - check both Mobile Phone & Enable Mobile Connect
        - destination number = 918005553333
7. Make sure Association Information - Line Association BOX is checked
8. Test by dial 1003, external phone 18005553333 should ring. Otherwise, reset  hqphone3 and recheck config

Install CUPC in 10 minutes

    1. on CM create a PresenceSIP trunk with DP-HQ and location.
    2. use same hquser1 (or hqphone1) for Presence, verify that hquser1 is associated to HQphone1 device and belong to Standard CTI enabled group. Also verify HQphone1 device "Owner User ID" is hquser1
    3. add Application User CtiGw with CTI enabled and control of all CTI devices
    4. verify System/Service Parameters/CUP Publish Trunk is set to PresenceSIP trunk name
    5. set CM license capability for users to CUP & CUPC
    6. add presence as an application server in CM
    7. Bring up Presence and add in CM info . Need CM security password
    8. Config CUP system/security/incoming & outgoing ACL to ALL
    9. set CUP system/service parameters  proxy-domain to CUPS.cisco.com
    10. add CM as a Presence gateway under menu Presence/Gateway
    11. configure routing at Presence/Routing to Default Cisco SIP Proxy TCP Listener
    12. set TFTP server under Application/CUPC/Settings with CM IP address and Default Cisco SIP Proxy TCP Listener
    13. under Application/CUPC/User Setting , select hquser1 and sign CTI profile to DP-HQ_cti_tcp_profile_synced_000.  If user hquser1 is not present, verify CM and CUPS sync in previous steps
    14. on Application/Desktop Control/Settings, set status to ON and fill in rest of info for CtiGw user
   
Verify with CUPC Desktop control on the local PC

Calls from PSTN to HQ did not went thru

HQ config #
mgcp
mgcp call-agent 10.1.200.21 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp bind control source-interface GigabitEthernet0/0.1
mgcp bind media source-interface GigabitEthernet0/0.1

Something wrong with the above config, perhaps I should have bind mgcp with Loopback0
Anyway, removed following config out of HQ router made it worked.

Did all config with Loopback interface now
 mgcp bind control source-interface Loop0

 mgcp bind media source-interfaceLoop0

IPMA in 10 minutes

- add PT-MANAGER and CSS-MANAGER. CSS-MANAGER = PT-USA + PT-MANAGER

- add CTI RP RP-IPMA, set device-pool to DP-HQ, CSS to CSS-MANAGER

- add new DN for RP-IPMA with 1XXX, partition = PT-USA

- add new phone service IPMA under device/device settings/phone services with URL:

http://10.1.200.21:8080/ma/servlet/MAService?cmd=doPhoneService&Name=#DEVICENAME#

- on System/service parameters, config IP ManagerAssistant app with correct Pub IP add and CTI route point , RP-IPMA

- add new PT under call routing/intercom/intercom route pratition PT-INTERCOM

- add new Phone Button template, with 2 lines + 1 intercom, IPMA-7970-3+3

- change HQ phone1 setting to

phone button template = IPMA-7970-3+3

softkey template = standard manager

- set line1 of HQphone1 to PT-MANAGER

- set intercom line of HQphone1 to *1001, PT-INTERCOM, CSS = PT-INTERCOM-GEN and speed-dial = *1002

- subscribe HQphone1 to IPMA service

- change HQ phone2 setting to

phone button template = IPMA-7970-3+3

softkey template = standard assistant

- set intercom line of HQphone2 to *1002, PT-INTERCOM, CSS = PT-INTERCOM-GEN and speed-dial = *1001

- add new DN to line2, DN = 1011, CSS = CSS-MANAGER, change display to IPMA

- add following users:

manager with "allow control of device from CTI" , User-Group = Standard CCM end user, Standard CTI allow control of all devices, Standard CTI enabled.  Primary Ext = 1001

assistant with "allow control of device from CTI",  User-Group = Standard CCM end user, Standard CTI allow control of all devices, Standard CTI enabled.  Primary Ext = 1002

- associate phones to manager and assistant

- on menu user management/end-user, click on manager user, under Related Links, select manager configuration. Uncheck "automatic configuration" , set intercom line to *1001, select associated assistans = assistant, selected lines = line 1 - 1001 - PT-manager

- perform same tasks with user assistant. available lines = line 2 - 1011, manager name = manager , manager line = line 1 - 1001 - PT-manager

RESTART Cisco IP Manager Assistant VIA CU Serviceability Web page

[Verification]
manager private number = 1001
manager proxy number  = 1011
manager intercom          = *1002

assistant number            = 1002
assistant intercom          = *1001
assistant proxy              = 1011

1) Check that manager’s phone has IPMA softkey set on it’s screen
2) Install the Cisco IPMA Console Application on Windows PC and log in as “assistant”
3) Place a call to 1001, ensure it get’s routed to the assistant phone, and pick it up from the IPMA console. Forward the call back to manager’s primary line
4) Configure from the manager’s phone to accept all calls and place a call to manager’s primary line once again
5) Press intercom line on either manager or assistant phone should ring the other side