H.323 Trunk-to-Trunk routing with CUCM 9-10 - Best way to program?

Looking for some expert guidance on the best way to implement trunk-to-trunk call routing in CUCM.  By "best" I mean user friendly/manageable/efficient etc...
Here's the application:  Inbound TDM calls will be hitting a large, H.323 gateway with 10-digit DNIS.  I need CUCM to recognize the dialed number, then immediately route the call back out one of several smaller, H.323 gateways (depending on the dialed number) while passing the original dialed number to the system at the receiving end of the call.
The inventory of dialed numbers is large, and routing changes to individual numbers will be made by non-technical staff; so I would prefer to maintain the routing rules in CUCM rather than in the gateways.
One other layer of complexity:  I need the ability to route the calls to destinations that may include only a single gateway, or span multiple gateways.
Any suggestions for a best way to implement this in CUCM would be greatly appreciated.

Simply make sure whatever dialed digits you get from GW A, via the inbound CSS, match a RP in CUCM which might point to an individual GW or a RL.
If necessary, adjust the dialed digits by adding/removing digits.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

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    Thks,

  • No ringbacktone for inbound calls with cucm 8.6

    Hi,
    we have this problem from many days...
    we have two branches with cucm cluster(Publisher and Subscriber) at Head office and cisco untiy.The branches are connected to Head office through MPLS vpn and all the ip phones are registred to publisher located at headoffice.
    our setup is like below
    HO and BR2 having SIP lines and BR1 has PSTN Lines.
    we implement greetings for head office and 2 branches at Headoffice Unity.
    when any call comes to headoffice gateway the greetings will be played and call will be diverted to the appropriate extension.everything is fine.
    But the problem is when the call comes to Branch gateway and the greetings will be played and the call gets diverted to the IP phone to which the caller dialed the extension. but the caller is not hearing the ringback tone while the extension is ringing. and the caller cannot know whether the extension is ringing or the call got disconnected.
    i tried to change the " Send h225 User Information Message"  in service parameters from "Use ANN for Ring Back" to H225 Info for call Progress Tone"
    whenever i am changing to  "H225 Info for call Progress Tone" then the branches problem getting solved but Headoffice getting the same problem.
    please can anyone help............................

    Hi Carlo,
    Thankyou for the Response...
    here is the Runn config for BR1 Connected to PSTN lines....
    voice-card 0
    dspfarm
    dsp services dspfarm
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    h323
    voice class codec 1
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    codec preference 3 g729r8
    codec preference 4 g729br8
    voice class h323 1
    h225 timeout tcp establish 3
    interface Tunnel100
    description " Tunnel JED-RYD "
    bandwidth 2048
    ip address 10.10.0.1 255.255.255.252
    tunnel source 172.31.217.202
    tunnel destination 172.31.3.18
    interface FastEthernet0/0
    description DAMMAM Local LAN
    no ip address
    duplex auto
    speed auto
    interface FastEthernet0/0.20
    description JEDDAH Local LAN
    encapsulation dot1Q 20
    ip address 192.168.20.5 255.255.255.0
    interface FastEthernet0/0.21
    description JEDDAH VOICE VLAN
    encapsulation dot1Q 21
    ip address 192.168.21.5 255.255.255.0
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 192.168.21.5
    interface FastEthernet0/1
    ip address 172.31.217.202 255.255.255.252
    duplex auto
    speed auto
    router eigrp 200
    network 10.10.0.0 0.0.0.3
    network 192.168.20.0
    network 192.168.21.0
    no auto-summary
    router bgp 65412
    no synchronization
    bgp log-neighbor-changes
    neighbor 172.31.217.201 remote-as 65000
    no auto-summary
    ip forward-protocol nd
    ip route 0.0.0.0 0.0.0.0 192.168.20.1
    ip route 192.168.20.50 255.255.255.255 192.168.20.1
    ip http server
    ip http access-class 23
    ip http authentication local
    ip http secure-server
    ip http timeout-policy idle 60 life 86400 requests 10000
    access-list 23 permit 10.10.10.0 0.0.0.7
    control-plane
    voice-port 0/0/0
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/0/1
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/0/2
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/0/3
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/2/0
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    connection plar 2022
    shutdown
    impedance complex2
    description STC
    voice-port 0/2/1
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    shutdown
    impedance complex2
    description STC
    voice-port 0/3/0
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/3/1
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/3/2
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/3/3
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    sccp local FastEthernet0/0.21
    sccp ccm 192.168.12.190 identifier 1 priority 1 version 5.0.1
    sccp ccm 192.168.12.189 identifier 2 priority 2 version 5.0.1
    sccp
    sccp ccm group 1
    associate ccm 1 priority 1
    associate ccm 2 priority 2
    associate profile 1 register CONFJEDRAW
    associate profile 2 register TRNJED
    associate profile 3 register MTPJED
    switchover method immediate
    switchback method immediate
    switchback interval 15
    dspfarm profile 2 transcode
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    maximum sessions 2
    associate application SCCP
    dspfarm profile 1 conference
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    shutdown
    dspfarm profile 3 mtp
    codec g729r8
    maximum sessions software 250
    associate application SCCP
    shutdown
    dial-peer voice 1 pots
    dial-peer voice 1000 voip
    description To CallManager - SBWPMPUB
    destination-pattern [1-5]...
    progress_ind progress enable 8
    voice-class codec 1
    voice-class h323 1
    session target ipv4:192.168.12.190
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 9001 pots
    description ** 02-6140294(outgoing) **
    destination-pattern [^2].T
    port 0/0/1
    dial-peer voice 9002 pots
    description ** 02-6140295(outgoing) **
    destination-pattern [^2].T
    port 0/0/2
    dial-peer voice 9003 pots
    description ** 02-6140296(outgoing) **
    destination-pattern [^2].T
    port 0/0/3
    dial-peer voice 9004 pots
    description ** 02-6140293(outgoing) **
    destination-pattern [^2].T
    port 0/0/0
    dial-peer voice 290 pots
    incoming called-number .
    direct-inward-dial
    dial-peer voice 9006 pots
    description ** 02-6529323(local) **
    destination-pattern [^0].T
    port 0/3/0
    dial-peer voice 9010 pots
    description ** 02-6578249(local) **
    destination-pattern [^0].T
    port 0/3/1
    dial-peer voice 9011 pots
    description "to pstn service"
    shutdown
    destination-pattern 0.T
    port 0/3/3
    dial-peer voice 9009 pots
    description "to pstn service"
    shutdown
    destination-pattern [^0].T
    port 0/3/2
    dial-peer voice 9005 pots
    destination-pattern .T
    dial-peer voice 1001 voip
    description To CallManager - Subscriber
    destination-pattern [1-5]...
    progress_ind progress enable 8
    voice-class codec 1
    voice-class h323 1
    session target ipv4:192.168.12.189
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 1002 voip
    description " TO Unity Greetings"
    destination-pattern 2050
    progress_ind progress enable 8
    voice-class codec 1
    voice-class h323 1
    session target ipv4:192.168.12.190
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 1003 voip
    description " TO Unity Greetings"
    destination-pattern 2050
    progress_ind progress enable 8
    voice-class codec 1
    voice-class h323 1
    session target ipv4:192.168.12.189
    dtmf-relay h245-alphanumeric
    no vad

  • Error message when syncing CUACEE 9.0.1.2 with CUCM 9.1.1

    Hi all,
    When I set up the CTI template, I chose owner userid as "none" but then when syncing the AC, I get the message on the sync report for every device of "Error -206 The specified table (owneruserid) is not in the database."
    If I set the template to use a specific owner userid and resync, then it works correctly, but I don't want these CTI ports associated with a userid.
    Is there any reason why this would be happening, as I have been able to do it in the past with "none" on 8.6

    Sean,
    Yes, this is a known defect:  CSCue69477 - CTI Route Port Synch Issue with CUCM.  As per the bug: 
    "This issue will occur if the "SUBSCRIBE Calling Search Space" under Protocol Specific Information in the Template for the CTI Route Ports is configured. To work around this, the "Owner User ID" under Device Information for the Template needs to be configured."
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