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  1. Senior Member
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    #1

    Default Incoming Dial-Peer-Translation pattern question

    I came across an article yesterday and it showed the steps how to fix Missed Call/Received Call numbers so that you can dial them from the menu correctly (auto-add a 9, etc.)?

    I tried it this morning and came up with this translation pattern:

    voice translation-rule 6
    rule 1 /^201\(.*\)/ /8\1/
    rule 2 /\(..........\)/ /81\1/

    voice translation-profile filter_Incoming
    translate calling 6

    This translation pattern rule 1 adds the dial out character 8 and strips 201 for local calls. Rule 2 adds dial out character 8 and adds 1 for long distance. The purpose of this translation rule is when the ephone receives the phone call the characters 8 and 1 are added so when you quickly need to redial you do not have to edit the number and add 8 for each call.

    I tested the translation-rule:
    ROUTER-2911#test voice translation-rule 6 9082121231

    Matched with rule 2
    Original number: 9082121231 Translated number: 819082121231
    Original number type: none Translated number type: none
    Original number plan: none Translated number plan: none


    ROUTER-2911#test voice translation-rule 6 2019121231
    Matched with rule 1
    Original number: 2019121231 Translated number: 89121231
    Original number type: none Translated number type: none
    Original number plan: none Translated number plan: none

    ROUTER-2911#

    Issue is I am not sure with my inbound call leg if it can even work. We dial out through the SIP Trunk and the incoming is translated to the AutoAttendant on Cisco Unity Express.



    voice translation-rule 1
    rule 1 /2015552100/ /2003/


    voice translation-profile CUE_Voicemail/AutoAttendant
    translate called 1


    dial-peer voice 9 voip
    description **Incoming Call from SIP Trunk**
    translation-profile incoming CUE_Voicemail/AutoAttendant
    call-block translation-profile incoming BLOCKED-INCOMING
    call-block disconnect-cause incoming call-reject
    session protocol sipv2
    session target dns:nd01-04.fs.SIPPROVIDER.net
    incoming called-number .%
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad


    Can what I am trying to do be done with my current setup?
    In life you have to make your own opportunities. Don't let anyone stop you from your dreams to many negative people want you to fail because they can't succeed.
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  3. Senior Member
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    #2
    So call calls come in to unity express first?
    Currently Reading

    CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related
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  4. Senior Member
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    #3
    Yes all inbound calls designated for the 201 office number are translated to pilot #2003 which forwards to the AA.
    Last edited by dmarcisco; 11-06-2014 at 04:39 PM.
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  5. Senior Member
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    #4
    so then they calls are being sent from the AA to the phones. In that cause you have the virtual dial peers that CME makes for the ephones which you can't manipulate. So you are kinda stuck where are you are at unless you hand out DID's
    Currently Reading

    CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related
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  6. Senior Member
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    #5
    Yea, thats what I figured earlier that it should work using DIDs. I just wanted to make sure there wasn't a way to rig it up before I drop the idea. Thanks for taking the time though, your still an O.G.
    In life you have to make your own opportunities. Don't let anyone stop you from your dreams to many negative people want you to fail because they can't succeed.
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