PRI Utilization in SIP Dialer

Hi All - Is there any way to achieve Utilization of Multiple PRI in Single Gateway with CUSP Deployment.
Let's say i have 5 PRI'S in same VG, Suppose if one of the PRI goes down is there any option to utilize remaining 4 PRI'S in the same VG.
I checked SRND, where i can see IF VG is overloaded and loses it's WAN connectivity with PSTN it will return error code as 503
In CUSP we can configure Failure response code as 503 so that CUSP will take alrenate element and route it accordingy.
If that is the case, We are not utilizing remaining 4 PRI'S in the first VG.
Is there any way to achieve the remaining 4 PRI'S without jumping to alternate element in CUSP.
SIVANESAN R       

Hello Sivanesan,
If you have 5 PRI's. Do you have 5 different dial-peers in the Voice Gateway for the same destination pattern ? Do you have Preference command given in the dial-peer ? Basically If a Call matches more than one outbound dial peer, the router itseld hunt them in order as per your configuation. If the Call setup failed for some reason, the next dial-peer will be attempted. Its based on the hunt order you specify.
In the voicegateway global config mode use the below command, you should see 7 options, based on your configuration the dial-peer will be hunted. This happens within the Gateway/Router before sending the response to CUSP
VG3845(config)#dial-peer hunt ?
  <0-7>  Dial-peer hunting choices, listed in hunting order within each choice:
  0 - Longest match in phone number, explicit preference, random selection.
  1 - Longest match in phone number, explicit preference, least recent use.
  2 - Explicit preference, longest match in phone number, random selection.
  3 - Explicit preference, longest match in phone number, least recent use.
  4 - Least recent use, longest match in phone number, explicit preference.
  5 - Least recent use, explicit preference, longest match in phone number.
  6 - Random selection.
  7 - Least recent use.
Regards,
Senthil

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    SIVANESAN R

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  • SIP Dialer Gateway

    Hello,
    I am building a design for one of our customers who is intersted to have the SIP dialer.
    I read about the SIP dialer and it looks like that the rel1xx on the SIP config should be enabled but for the CVP it needs to be disabled.
    Is this mean that i have to had a sperate GW for the SIP dialer or there is some workaround for this.
    Thanks
    Amer

    Amer,
         Everyone is right... You need one Dial-peer for the SIP Dialer w/ REL1xx enabled, however what was not mentioned is you cannot hand the call off to CVP with out a translational routing. Your CVP Dial-Peer will actually need to be configured to match the did range you defined for tranlation routes, in my environment it 12[1-2].. for my first CVP Call Server and follow suit for the remaining 3. When the Dialer hands the call off to CVP it will use the trans route which will need to match a a Dialpeer for the CVP Call Server as the Voice Gateway will be the one performing the actual transfer.
    Also see my post regarding Dissabling Ring Back for SIP Dialer Calls.
    https://supportforums.cisco.com/docs/DOC-21998
    Make sure you adhere to ALL the IOS revision requirements defined in the UCCE SRND and CVP SRND ** A2Q(Cisco BU) will will not certify the design otherwise and if they overlook the IOS versions then TAC will definatly tell you the first time you open a TAC case that you have to modify the IOS version to match the SRND.
    My CC
    CVP 8.5
    ICM 8.5.2
    CUCM 8.5.1
    CUSP 8.5
    IOS 15.1.3T2

  • SIP Dialer codec

    Hello,
    There is one question to community:
    Where we can setup preferable codec in Cisco SIP dialer?
    It is needed to change g711ulaw which is default SIP Dialer codec to g711alaw. 
    Thanks in advance,
    Regards,
    Alex

    Chintan,
    I see that INVITE from dialer comes with one codec g711ulaw. Please see below:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 10.13.81.3:58800;branch=z9hG4bK-d8754z-57145a04a377e42c-1---d8754z-;rport
    Max-Forwards: 70
    Require: 100rel
    Contact: <sip:[email protected]:58800>
    To: <sip:[email protected]>
    From: <sip:[email protected]>;tag=23395838
    Call-ID: eb13700c-00095a52-50032441-92184f3b
    CSeq: 1 INVITE
    Session-Expires: 1800
    Min-SE: 90
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS
     --More--        
    Content-Type: Multipart/mixed;boundary=uniqueBoundary
    Supported: timer, resource-priority, replaces
    User-Agent: Cisco-SIPDialer/UCCE8.0
    Content-Length: 608
    Remote-Party-ID: <sip:@10.13.81.27>;party=calling;screen=no;privacy=off
    --uniqueBoundary
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2884 2524 IN IP4 172.19.155.41
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 19994 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    --uniqueBoundary
     --More--        
    Content-Type: application/x-cisco-cpa
    Content-Disposition: signal;handling=optional
    Events=FT,Asm,AsmT,Sit,Piano
    CPAMinSilencePeriod=608
    CPAAnalysisPeriod=2500
    CPAMaxTimeAnalysis=3000
    CPAMaxTermToneAnalysis=30000
    CPAMinValidSpeechTime=112
    Regards,
    Alex

  • Unable to generate CUSSM Trunk/PRI Utilization Report

    Dear All,
    We want to generate Voice Gateways E1 Utilization report from CUSSM 8.5. I have configured CUOM 8.5 and CUSM 8.5 but i am unable to get the date for number of active calls and % utilization/PRI.
    Thanks in Advance.
    Hassan

    Well i have done a little reading and found out that CUSSM 8.5 gives trunk utilizaition but only for MGCP Controlled Gateways. Reverting back to version 1.3
    Hassan

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