SIP to SIP Connectivity

Hi,

 

We have a CCM 7.1.3 and Voice Gateway and a Public SIP Server

 

Now Need to create a SIP Truck from CCM To VG and VG to Public SIP Server. Both CCM and VG are in Private Network and Public Server Link is directly connected to VG and all private IP is NAT to public IP.

 

Can you let me know how to configure the same. Tried to do that, but call is not going through.  I am getting a call reject from VG side to CCM when i used Wireshark to identify the problem.

When I do Dailed number Anayasis From SIP Truck, It showing as pattern " Block this pattern". Can you help in gettitng this configure.

but In route pattern, It is not blocked!

 


  • Results Summary

    • Calling Party Information

      • Calling Party
        = 52400

      • Partition
        =

      • Device CSS
        =

      • Line CSS
        =

      • AAR Group Name
        =

      • AAR CSS
        =

    • Dialed Digits
      = 20016094096950

    • Match Result
      = BlockThisPattern

    • Called Party Number
      =

    • Matched Pattern Information

      • Pattern
        =

      • Partition
        =

    • Pattern Type
      =

    • Time Zone
      =

    • Outside Dial Tone
      = NO

  • Call Flow

    • TranslationPattern :Pattern=

      • Partition =

      • Positional Match List =

      • Calling Party Number = 52400

      • PreTransform Calling Party Number =

      • PreTransform Called Party Number =

      • Calling Party Transformations

        • External Phone Number Mask = NO

        • Calling Party Mask =

        • Prefix =

        • CallingLineId Presentation =

        • CallingName Presentation =

        • Calling Party Number = 52400

      • ConnectedParty Transformations

        • ConnectedLineId Presentation =

        • ConnectedName Presentation =

      • Called Party Transformations

        • Called Party Mask =

        • Discard Digits Instruction =

        • Prefix =

        • Called Number =

  • Alternate Matches

    • Note: Information Not Available



NOTE: The analysis results are purely based on configurations
available in the Cisco Communications Manager database. For Gateway
outbound calls, call details might differ depending on the Gateway's
settings.

 

 

Comments

  • To start:

    I dont now your CUCM setup, but reasons why DNA isnt working:

    You didnt set a device or line css

    You didnt to prefix a 9 to match your outgoing route patterns? You have to put the same digits that cucm receives from the phone.

     

    For the CUBE problem, can you include the voice configuration of the cube and a 'debug ccsip messages' of an outgoing call?

  • Press new call and start dialing.

    Do you get reorder tone at some point or the message "your call cannot be compelted as dialed"? It seems that there is no pattern to match. Check if there's a partition or CSS misconfiguration.

  • hi,

     

    I am getting a busy tone...  Above provided analysis is the one that take the anayasis from Trunk.

     

    If i am doing the DNA using Phone (TAB), I am getting correct pattern allowing....

     

    For breifing, below are the configuration, i have done.

     

    CUCM IP: 172.25.168.11, 172.25.168.12, 172.25.168.13

     

    VG IP: 172.25.32.50

     

    VOIP SIP Server: Public IP: 7.7.7.7 ( not the actual Public IP)

     

    IP address are changed and configuration is posted.

    Configuration in CUCM:

    1) Established the SIP Trunk(to 172.25.32.50)

    2) Route Pattern to VG(172.25.32.50)

    0.00! ---> Route to VGSIP Truck ( discarding predot)

    ------------------------------------------------------------------------------

    Configuration in VG:

     

    voice service voip
     allow-connections sip to sip
     fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
    !
    application
     global
      service alternate Default
     !

    interface GigabitEthernet0/0
     ip address 172.25.32.50 255.255.255.0
     ip nat inside
     ip virtual-reassembly
     duplex auto
     speed auto
    !

    interface GigabitEthernet0/2
     ip address 6.6.18.250 255.255.255.252
     ip nat outside
     ip virtual-reassembly
     duplex auto
     speed auto
    !

    ip nat pool Voip 11.11.48.153 11.11.48.153 netmask 255.255.255.248
    ip nat inside source list 135 pool Bang_Voip overload
    ip nat inside source static 172.25.168.13 11.11.48.154
    ip nat inside source static 172.25.168.11 11.11.48.155
    ip nat inside source static 172.25.168.12 11.11.48.156
    ip nat inside source static 172.25.32.50 11.11.48.157
    ip route 172.25.0.0 255.255.0.0 172.25.32.1
    ip route 7.7.7.7 255.255.255.255 6.6.18.249
    !

    dial-peer voice 10 voip
     description ****For VOIP Server****
     destination-pattern 00T
     session protocol sipv2
     session target ipv4:7.7.7.7
     dtmf-relay h245-alphanumeric
    !
    dial-peer voice 11 voip
     destination-pattern .....
     session protocol sipv2
     session target ipv4:172.25.168.11
    !
    !
    sip-ua
     sip-server ipv4:7.7.7.7
    !
    !
    !

  • can you give us:

    debug voip dialp all

    and then another call with

    debug ccsip messages

  • Try binding the SIP traffic to a source address under "voice service voip."

    Sent from my iPhone

    On Nov 26, 2011, at 11:36 AM, KajaHanumantharao <[email protected]> wrote:

    hi,

     

    I am getting a busy tone...  Above provided analysis is the one that take the anayasis from Trunk.

     

    If i am doing the DNA using Phone (TAB), I am getting correct pattern allowing....

     

    For breifing, below are the configuration, i have done.

     

    CUCM IP: 172.25.168.11, 172.25.168.12, 172.25.168.13

     

    VG IP: 172.25.32.50

     

    VOIP SIP Server: Public IP: 7.7.7.7 ( not the actual Public IP)

     

    IP address are changed and configuration is posted.

    Configuration in CUCM:

    1) Established the SIP Trunk(to 172.25.32.50)

    2) Route Pattern to VG(172.25.32.50)

    0.00! ---> Route to VGSIP Truck ( discarding predot)

    ------------------------------------------------------------------------------

    Configuration in VG:

     

    voice service voip
     allow-connections sip to sip
     fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
    !
    application
     global
      service alternate Default
     !

    interface GigabitEthernet0/0
     ip address 172.25.32.50 255.255.255.0
     ip nat inside
     ip virtual-reassembly
     duplex auto
     speed auto
    !

    interface GigabitEthernet0/2
     ip address 6.6.18.250 255.255.255.252
     ip nat outside
     ip virtual-reassembly
     duplex auto
     speed auto
    !

    ip nat pool Voip 11.11.48.153 11.11.48.153 netmask 255.255.255.248
    ip nat inside source list 135 pool Bang_Voip overload
    ip nat inside source static 172.25.158.13 11.11.48.154
    ip nat inside source static 172.25.158.11 11.11.48.155
    ip nat inside source static 172.25.158.12 11.11.48.156
    ip nat inside source static 172.25.32.50 11.11.48.157
    ip route 172.25.0.0 255.255.0.0 172.25.32.1
    ip route 7.7.7.7 255.255.255.255 6.6.18.249
    !

    dial-peer voice 10 voip
     description ****For VOIP Server****
     destination-pattern 00T
     session protocol sipv2
     session target ipv4:7.7.7.7
     dtmf-relay h245-alphanumeric
    !
    dial-peer voice 11 voip
     destination-pattern .....
     session protocol sipv2
     session target ipv4:172.25.168.11
    !
    !
    sip-ua
     sip-server ipv4:7.7.7.7
    !
    !
    !




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