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