Device Weight: Route Pattern Vs IP Phone

Hi, we are starting the planning for removing two CO class GTD5 switches that route 28,000 DID numbers for the County. The switches are +16 years old and maintenance is expensive. Our Callmanager cluster is designed to replace the existing phone infrastructure (2 CO switches, +60 PBXs, and +20 key systems). Here's a quick overview of our Cluster:
- Publisher only running database services (MCS7845H2)
- Two TFTP servers only running TFTP service and MOH streaming (multicast) (MCS7835H)
- Two subscribers configured as 1-to-1 backup (MCS7845H2)
- CCM version 4.1(3)sr3b
We are planning to add the additional six MCS7845H2 subscribers as we need the capacity.
The first step in our migration is to move all the DID's to new T1/PRI's on a set of six Communication Media Modules (CMM) spread out over four Cat6513 switches.
Essentially our CCM cluster will be acting as a Tandem switch until we get all the trunks moved off the GTD5 CO switches to the CMMs.
Unfornutately all of the 28,000 DIDs are pretty much shot gunned all over the County. So, we will have +20,000 route patterns in the beginning. Over time we will also be converting sites to IP Tel and removing the route patterns as we migrate.
My question: Does a route pattern for one directory number carry the same device weight as an IP phone with the directory number assigned to a line?
We are thinking it does and if it does, then we need to scale up our CCM cluster for 30,000 devices before we start the migration.
Thanks in advance for any advice.
Tom.

Thanks Greg for taking the time to reply. Our cluster design is following the "Cisco Callmanager Best Practices" book. The Publisher and TFTP servers are not running the Callmanager service. This allows 8 subscribers running the Callmanager service in the Cluster.
Our research is trying to understand the cost of a route pattern in terms of the dialing forest and the impact on the subscriber's memory and CPU.
We own two complete prefixes plus another 8,000 DNs from a third prefix. We fortunately do not have an overlapping dialing plan. Each directory number will either be assigned to an IP phone or it will belong to phone on a PBX.
All directory numbers for IP phones belong to the same partition and we also followed the Best Practices book for our dialing plan and use the line/device CSS design.
All +80 remote sites connect back to our GTD5 and the GTD5 routes all the numbers to the remote sites.
We are first migrating all the DID services to our new PRI's handled by the Callmanager cluster. We when start the migration it will be a simple process of a route pattern such as [4-5]XXXX to route the 874-xxxx and 875-xxxx numbers to the GTD5. Then as we disconnect the tie-lines from each remote PBX and re-connect it to the CMM for Callmanager to route, we will need to add all the specific route patterns to route the numbers for the site.
It would be ideal if we did not have to retire the GTD5 switch. We would follow our 5 to 7 year plan to migrate the entire County to IP Tel and leave the GTD5 in place routing the numbers to the remote PBXs. However, we have been directed by management to decommission the GTD5 switch within 12 months.
So we are trying to understand the impact to the subscribers when we begin adding 1,000's of route patterns. We are planning to consolidate as many of the route patterns as possible to reduce the number of route patterns. However, we inherited a design that we refer to as "Number-lose-ability", where individual numbers are routed and not blocks of numbers. Over the years of adds, moves, and changes the numbers have been scattered to all the sites. We have very few sites with consecutive numbers.
Another question that we are trying to answer: what is the cost of a route pattern such as 5555x compared to 10 individual route patterns for the same number range. Again, in terms of memory and CPU on the subscriber doing the digit analysis. We are asking this question because we may have 6 of the 10 numbers going to the same PBX, but the other 4 numbers each going to a different site. To consolidate route patterns we would add the 5555X pattern and the four individual route patterns. What we do not know is how the 5555x is added to the dialing forest. Is it expanded to 10 patterns or just one expression.
Thanks again for any help,

Similar Messages

  • Limit of Route Patterns on CallManager ?

    Hi Pros,
    Wanted to check, I have around 60 RPs on CallManager. My CCM hardware is MCS7825. (1 Pub, 1 Sub)
    Would it cause too much load on it if I add say 10 - 20 more RPs on it ?
    I am also creating around 12 - 16 new Partitions & CSS each on it.
    Regards,
    Pratik

    It depends on what else do you have on your system.
    Maximum Number of Devices per Server Platform:
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    - MCS-7835 --> 2500
    - MCS-7825 --> 1000
    - MCS-7815 --> 300
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    * Unique line appearance --> 5
    * Shared line appearance --> 4
    * Route Pattern --> 2
    * Translation Pattern --> 1

  • How to create 20000 route pattern at the same time?

    I would like to create a partition which contains 20000 different number which a list of ip phone are going to be able to call.
    It would be difficult for me of create route pattern one by one. Please How may I proceed?
    Thanks in advance.
    Best Regards,

    If I am understanding you correctly you're trying to allow a subset of the phones in your organization to dial numbers other phones cannot.
    Assuming I understand you correctly, you would be better off to go about this in the opposite direction. Current SRND recommendations indicate that the Device CSS is capable of reaching all destinations while the Line CSS imposes any dialing restrictions. It may very well be that you need to create 20,000 route or translation patterns; however, the advise is to place these in a partition that is listed on the Directory Number [Line] CSS and blocks the destination from those phones that should not be able to reach it. This becomes especially useful with other features such as Extension Mobility as the unique list of restrictions follow that DN regardless of the device it is being used on.
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    Lastly, be mindful of the capacity limits of the database:
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    A Unified CM cluster with a very large dial plan containing many gateways, route patterns, translation patterns, and partitions, can take an extended amount of time to initialize when the Cisco CallManager Service is first started. If the system does not initialize within the default time, you can modify the system initialization timer (a Unified CM service parameter) to allow additional time for the configuration to initialize. For details on the system initialization time, refer to the online help for Service Parameters in Unified CM Administration.
    You should work with your Cisco Partner who has access to sizing tools before doing this. These tools can calculate the impact that the additional route patterns will have on the cluster.
    For Cisco partners and employees, the tools are available at the following locations:•Cisco Unified Communications Sizing Tool is available athttp://tools.cisco.com/cucst•Cisco Unified CM Capacity Tool is available athttp://www.cisco.com/go/cucmct

  • How do I add a route pattern to CUCM 7.1

    I am currently using CUCM7.1 and need to  add the route pattern 9911 to dial out to emergency dispatch.  I do not want the capability of dialing 9 for an outside line for all users, just when I a calling 911 emergency.  We recently changed to dialing a # for an outside line due to excessive 911 hangup calls.  I tried adding 9911 to the route pattern list but I am missing something.  I received the message stating could not complete call as dialed. Thank you, Cindy

    Are you sure the phone has a CSS that allows you to dial 9911??
    HTH
    java
    If this helps, please rate
    www.cisco.com/go/pdihelpdesk

  • Route pattern to SIP trunk problem

    Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
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    But when I make a call that matches this route pattern all I get is the intermittent beep message from the phone. I cannot route calls succesfully through it.
    I have checked network connectivity and all is fine. The IP address I specfied in CUCM for the SIP trunk is simply one of the interfaces on the 2801 router and it is definitley reachable.
    I also activated "debug ccsip all" on the 2801 gateway router but nothing appears. So it seems like the calls are not even reaching the 2801 gateway ?
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    Do I need to disable CME somehow on the gateway first ?  Or am I not doing something correct in the CUCM config ?
    Thank you kindly for any suggestions.
    ps. I have attached a couple of screenshots of my config.

    Hello, thanks for helping.
    I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
    Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
    As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
    I have included my running config from the gateway router below..
    One addition I made was to add an incoming dial peer. That is "dial peer 5,  description CUCM SIP trunk".
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    Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
    I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
    So why is it not then moving down the SIP trunk to my gateway and getting picked up by the incoming dial peer ?
    Thanks if you guys can offer any more help.
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Router
    boot-start-marker
    boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
    boot-end-marker
    no aaa new-model
    clock timezone nzst 13 0
    dot11 syslog
    ip source-route
    ip dhcp pool DATA_SCOPE
       network 192.168.200.0 255.255.255.0
       default-router 192.168.200.1
       dns-server 8.8.8.8
    ip dhcp pool VOICE_SCOPE
       network 192.168.100.0 255.255.255.0
       default-router 192.168.100.1
       option 150 ip 192.168.2.115
    ip dhcp pool MGMT_SCOPE
       network 192.168.1.0 255.255.255.0
       default-router 192.168.1.99
    ip cef
    ip name-server 4.2.2.2
    no ipv6 cef
    multilink bundle-name authenticated
    voice class codec 1
    codec preference 1 g711alaw
    codec preference 2 g729r8
    codec preference 3 g711ulaw
    codec preference 4 ilbc
    voice translation-rule 1
    rule 1 /^9/ //
    voice translation-profile Strip9ToGetOut
    translate called 1
    voice-card 0
    crypto pki token default removal timeout 0
    crypto pki trustpoint TP-self-signed-2995340181
    enrollment selfsigned
    subject-name cn=IOS-Self-Signed-Certificate-2995340181
    revocation-check none
    crypto pki certificate chain TP-self-signed-2995340181
    certificate self-signed 01
      3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
      31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
      69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
      32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
      4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
      34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
      8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
      AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
      675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
      12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
      9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
      551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
      F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
      396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
      81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
      D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
      7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
      CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
          quit
    license udi pid CISCO2801 sn FTX0947W07M
    username xxx privilege 15 password 0 xxx
    interface FastEthernet0/0
    ip address 192.168.3.50 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    no ip address
    duplex auto
    speed auto
    interface FastEthernet0/1.2
    encapsulation dot1Q 2
    ip address 192.168.2.1 255.255.255.0
    interface FastEthernet0/1.99
    encapsulation dot1Q 99
    ip address 192.168.1.99 255.255.255.0
    interface FastEthernet0/1.100
    description voice_VLAN
    encapsulation dot1Q 100
    ip address 192.168.100.1 255.255.255.0
    interface FastEthernet0/1.200
    description data_VLAN
    encapsulation dot1Q 200
    ip address 192.168.200.1 255.255.255.0
    ip forward-protocol nd
    ip http server
    ip http authentication local
    ip http secure-server
    ip route 0.0.0.0 0.0.0.0 192.168.3.1
    logging esm config
    tftp-server flash:/phone/7940-7960/P00307020200.bin alias P00307020200.bin
    tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
    tftp-server flash:/phone/7940-7960/P00307020200.sb2 alias P00307020200.sb2
    tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
    control-plane
    mgcp fax t38 ecm
    dial-peer voice 1 voip
    description local_7_Digit_Calling
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 9[2-9]......
    session protocol sipv2
    session target ipv4:203.184.16.2
    voice-class codec 1 
    dial-peer voice 2 voip
    description international_calling
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 900T
    session protocol sipv2
    session target ipv4:203.184.16.2
    voice-class codec 1 
    dial-peer voice 3 voip
    description national_calling
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 90[34679].......
    session protocol sipv2
    session target ipv4:203.184.16.2
    voice-class codec 1 
    dial-peer voice 4 voip
    translation-profile outgoing Strip9ToGetOut
    destination-pattern 90[34679].......
    dial-peer voice 5 voip
    description CUCM SIP trunk
    destination-pattern 2...
    session protocol sipv2
    session target ipv4:192.168.2.115
    voice-class codec 1 
    sip-ua
    authentication username xxxxxxxxxx password xxxxxxxx
    060
    telephony-service
    max-ephones 10
    max-dn 20
    ip source-address 192.168.1.99 port 2000
    load 7960-7940 P00307020200
    max-conferences 4 gain -6
    transfer-system full-consult
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1  dual-line
    number 1000
    name Lydia Francis
    ephone-dn  2  dual-line
    number 1001
    name Leah Francis
    ephone-dn  3  dual-line
    number 1002
    n
    ephone-dn  4  dual-line
    number 1003
    ephone  1
    mac-address C80A.A970.01DE
    type CIPC
    button  2:2
    ephone  2
    mac-address 000C.3070.8705
    button  1:1 2:15
    ephone  3
    mac-address 000C.8546.5954
    button  1:3 2:15
    line con 0
    logging synchronous
    line aux 0
    line vty 0 4
    privilege level 15
    login local
    transport input telnet ssh
    scheduler allocate 20000 1000
    ntp server 195.43.74.123
    end

  • CUCM to CVP calls. CTI-RP vs Route Pattern

    CVP 9 or above
    CUCM 9 or above
    Requirement:
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    "... Calls Originated by Unified CM
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    Consultative Warm Transfer: For these calls, a Unified CM agent places the caller on hold and dials in to Unified ICME to reach a second agent .... "
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    Thanks & Regards,
    Kartik

    Kartik,
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    -Jameson

  • 9@ Route Pattern Matched Issues

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    Hi,
    As per the following link
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/5_0_4/ccmsys/ccmsys/a03rp.html#wp1050657
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    https://supportforums.cisco.com/discussion/10698966/9-route-pattern
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  • Route pattern help

    I need a route pattern for
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    87000 ? 87199,
    88470 ? 88519,
    88520 ? 88569
    I am a bit stuck, and don't want to enter loads it, can it be done with 1?
    Thanks

    All you need then is the 8XXXX route pattern pointing to CM B. If phone on CM A dials any extesnion registered with CM A (granted it has access to it's partition) it will be more explicit match. If there is no DN matching on CM 1 then the route pattern will be used, again assuming it's partition is listed in the phone's CSS.
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  • Route Patterne

    Hello experts,
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    When I dial a 1800 number it works but not any other such as 1866, 1877 etc. I just don't get it. Why it works for 1800 numbers and not for the other ones. Could you please let me know if I am missing anything here?
    Thanks,

    Bahman,
    What type of gateway is this ? MGCP or H323 gateway ? Are you having a PRI (T1/E1) ? If yes can you get a debug isdn q931 output of a failed 1866/1877 call ?
    I would also check if the 1877/1866 numbers you are dialing are valid numbers. Try calling them from a cell phone and see if it works.
    Like Chris suggested, your pattern should be 9.18XX format so that either you do predot digit strip for Mgcp gateway or no digit strip for h323 gateway.
    HTH
    Sankar.
    PS: please rate all posts!

  • Translation Pattern for a Route Pattern

    I´m trying to make a translation pattern for a route pattern to add a * or a # in the end of the number I'm dialing for example the route pattern is 9.0414XXXXXXX and I want to change to XXXXXXX*. If I dial 904141309131 I see in the phone 4130913*. It seem to be taking a 4 that belong to the 0414 and it is eliminating the las number that in this example is 1. To me the number that I must see when the translation is made is 1309131* and not 4130913*. Is this the way it shoul be done?

    Martin,
    Did you ever find out how to do this ? I have the same requirement and have tried various transform masks none of which has succeeded.
    Thanks in Advance.
    Mark.

  • Route Pattern / Extension Issue

    Hello! I'm currently having an issue with a new block of extensions in Call Manager 7.1.5.
    We recently ordered a new block of DIDs. The telco sends us the last 4 digits. The last 4 are 9XXX. When dialing these extensions internally, there is an 8-10 second pause of dead air and then it eventually rings through to the phone.
    All of my Route Patterns for each site start with a " 9. "
    My guess is that when dialing the extension, Call Manager tries to match it to a Route Pattern since it starts with a "9" followed by 3 digits. After it cycles through the Route Patterns and doesn't match it, it eventually routes it to the phone. Is this what's happening? And if so, any thoughts to how I alleviate this issue?
    Any recommendations are appreciated.
    Thanks!

    Hi James,
    It appears to be correct. Due to overlapping pattern, the call manager would wait for an inter-digit timeout to expire and route the calls to the correct pattern.
    As a workaround, you can try to lower the T302 Timer(inter-digit timeout) in call manager to 4-5 seconds (Service Parameters -> Cisco Call Manager -> T302 Timer).
    As this issue is experienced with overlapping pattern, the best way to resolve it to design your dial plan efficient so that there is no overlapping.
    HTH,
    Jagpreet Singh Barmi

  • Adding a Route Pattern to a Line Group?

    Hi
    We have an analog device which is patched into our Voice Gateway and has a dial peer using extension 444.
    I have setup a route pattern within CUCM which points at the gateway for 444.
    However I can't seem to add this Route pattern to a line group to be part of a hunt?
    The line group seems to be just internal extensions but I need the route pattern of 444 to be in the same line group as two internal extensions?
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