SIP ITSP on CUCM 10.5.2 (No CUBE) Incoming calls fail, outgoing are fine

Hi,
I am in the process of upgrading a customer who is on 8.0.3. They have an ITSP terminating SIP Trunk directly on the CCM Server
I upgraded the system to 10.5.2. During cutover I was able to make outgoing calls but all incoming calls were failing.
After reverting back to the old system, everything is working fine again, and I dont understand what could be the possible issue that it doesnt work on 10.5.2 but it works well on 8.0.3.
I checked almost everything and dont find anything that stands out, which may be contributing to the issue.
Any idea what could be missing here?
Thanks

Thanks for all your tips.
It was turned out that, the URI was a FQDN and during the first install of the 8.0.3 (in the sandbox) I had not bothered to get the DNS Services replicated and then didnt check if the ITSP was sending the invite on URI based on FQDN or IP Address
Thanks

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  • SIP incoming call, won't work (CME)

    Hi all,
    I'm facing a weird problem and the sip-provider can't help. I suppose there is a problem with the dial-peer/translation-rule but I can't figure it out...
    There is a CME (c2800nm-ipvoice-mz.124-11.XW10.bin, CME Version 4.2(0)) with a
    SIP trunk.
    Outgoing calls are working (DID).
    Incoming calls (all DID) are ringing on the same internal number.
    The situation:
    - external  call on 0815440097 is ringing on the internal nr. 296 (should be 297)
    - external call on 0815440096 is ringing on the internal nr. 296
    Here the config:
    ================================
    voice service voip
    allow-connections sip to sip
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    sip
    no update-callerid
    voice translation-rule 40
    rule 2 /\(.*\)/ /9\1/
    voice translation-rule 190
    rule 1 /^0\(.*\)/ /\1/
    rule 2 /^9\(.*\)/ /\1/
    voice translation-rule 191
    rule 2 /296/ /0815440096/
    rule 3 /297/ /0815440097/
    voice translation-rule 192
    rule 2 /^0815440097/ /297/
    rule 3 /^0815440096/ /296/
    voice translation-profile TP_IN_SIP
    translate calling 40
    translate called 192
    voice translation-profile TP_OUT_SIP
    translate calling 191
    translate called 190
    dial-peer voice 2000 voip
    description *** SIP-TRUNK (IN/OUT) ***
    translation-profile incoming TP_IN_SIP
    translation-profile outgoing TP_OUT_SIP
    destination-pattern 0.T
    b2bua
    session protocol sipv2
    session target dns:sip12.e-fon.ch
    session transport udp
    incoming called-number 0815440096
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    sip-ua
    credentials username 0815440096 password 7 xxxx realm sip12.e-fon.ch
    keepalive target dns:sip12.e-fon.ch
    authentication username 0815440096 password 7 xxxx
    calling-info pstn-to-sip from number set 0815440096
    no remote-party-id
    retry invite 2
    retry response 2
    retry bye 2
    retry register 2
    retry options 1
    registrar dns:sip12.e-fon.ch expires 69
    sip-server dns:sip12.e-fon.ch
    reason-header override
    connection-reuse
    host-registrar
    sh sip-ua register status
    Line                              peer        expires(sec)  registered
    ================================  ==========  ============  ==========
    0815440096                        20005       18            yes
    Here the CCSIP MESSAGE debug (looks ok):
    (call from 0000000000 to 0815440097)
    ===============================
    Mar  8 21:55:10.469 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Record-Route: <sip:212.55.198.132;lr=on;ftag=as00cd0e7f>
    Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0
    Via: SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
    Max-Forwards: 69
    From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
    To: <sip:[email protected]:5060>
    Contact: <sip:[email protected]:5061>
    Call-ID: [email protected]
    CSeq: 102 INVITE
    User-Agent: e-fon
    Date: Thu, 08 Mar 2012 20:55:10 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    X-IPCONNECT: 0815440097
    X-Number: 0815440097
    Content-Type: application/sdp
    Content-Length: 415
    v=0
    o=root 770254981 770254981 IN IP4 212.55.198.134
    s=Asterisk PBX 1.6.1.20
    c=IN IP4 212.55.198.134
    t=0 0
    m=audio 11886 RTP/AVP 8 9 111 3 18 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    Mar  8 21:55:10.481 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0,
    Via:SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
    From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
    To: <sip:[email protected]:5060>
    Date: Thu, 08 Mar 2012 20:55:10 GMT
    Call-ID: [email protected]
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Content-Length: 0
    Here is the VOICE DIAL-PEER debug (call from 0000000000 to 0815440097):
    =============================================
    Mar  8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
    Calling Number=0815440096, Called Number=0815440096, Peer Info
    Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
    Match Rule=DP_MATCH_DEST; Called Number=0815440096
    Mar  8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
    Result=Success(0) after DP_MATCH_DEST
    Mar  8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
    Result=SUCCESS(0)
    List of Matched Outgoing Dial-peer(s):
    1: Dial-peer Tag=20005
    2: Dial-peer Tag=2000
    Mar  8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
    Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
    Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
    Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
    Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
    Mar  8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
    Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
    Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
    Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
    Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
    Mar  8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
    Calling Number=0000000000, Called Number=0815440096, Voice-Interface=0x0,
    Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
    Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
    Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
    Mar  8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
    Calling Number=0815440096, Called Number=0815440096, Peer Info
    Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
    Match Rule=DP_MATCH_DEST; Called Number=0815440096
    Mar  8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
    Result=Success(0) after DP_MATCH_DEST
    Mar  8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
    Result=SUCCESS(0)
    List of Matched Outgoing Dial-peer(s):
    1: Dial-peer Tag=20006
    2: Dial-peer Tag=2000
    Mar  8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
    Calling Number=0815440096, Called Number=, Voice-Interface=0x0,
    Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
    Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
    Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=2000
    Mar  8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
    Calling Number=, Called Number=0815440096, Peer Info
    Type=DIALPEER_INFO_SPEECH
    Mar  8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
    Match Rule=DP_MATCH_DEST; Called Number=0815440096
    Mar  8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
    Result=Success(0) after DP_MATCH_DEST
    Mar  8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersMoreArg:
    Result=SUCCESS(0)
    List of Matched Outgoing Dial-peer(s):
    1: Dial-peer Tag=20006
    2: Dial-peer Tag=2000
    show dial-peer voice summary:
    dial-peer hunt 0
    AD                                    PRE PASS                OUT
    TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT
    PORT
    555    voip  up   up             555                0  syst loopback:rtp
    20001  pots  up   up             296$               0                          50/0/1
    20002  pots  up   up             297$               0                          50/0/2
    2000   voip  up   up             0.T                0  syst dns:sip12.e-fon.ch
    20005  pots  up   up             0815440096$        0                     50/0/150
    20006  pots  up   up             0815440097$        9                     50/0/2
    voip translation debugging (call from 0794142975 to 0815440097):
    =========================================
    Mar  8 22:35:26.145 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=1
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=0
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0794142975 type=unknown plan=unknown numbertype=calling
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/sed_subst: Successful substitution; pattern=0794142975 matchPattern=(.*) replacePattern=9\1 replaced pattern=90794142975
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: xlt_number=90794142975 xlt_type=unknown xlt_plan=unknown
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=3
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: No match found
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number=0815440096 type=unknown plan=unknown
    Mar  8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=1
    Mar  8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: No profile found in peer 20005 for outgoing direction
    Mar  8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: calling_number=90794142975 calling_octet=0x0
            called_number=0815440096 called_octet=0x0
            redirect_number= redirect_type=0 redirect_plan=0        redirect_PI=-1 redirect_SI=-1
    Mar  8 22:35:26.181 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=2
    Thanks,
    Norbert

    Hi Alex,
    Thank you for the reply.
    After changing the "incoming called-number" I got the same output.
    The weird think is, why the dial-peer debug shows the 0815440096 number, despite the right "to: number" in the SIP-Message.
    Is there a problem with the "voice service voip" or "sip-ua"?
    on the voice translation debug I see:
    Match Rule=DP_MATCH_TO_URI; URI=sip:0815440097
    Match Rule=DP_MATCH_FROM_URI; URI=sip:0819262424
    But I guess the translation rule is maching this one:
    Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
    So how can the voice translation rule be set to map the entry DP_MATCH_TO_URI; URI=sip:0815440097
    Thanks for the help.
    Regards,
    Norbert
    voip translation debugging (call from 0819262424 to 0815440097):
    ===================================================
    Mar  9 07:45:16.371 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=1
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=0
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0819262424 type=unknown plan=unknown numbertype=calling
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0819262424 matchPattern=(.*) replacePattern=9\1 replaced pattern=90819262424
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=90819262424 xlt_type=unknown xlt_plan=unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0815440096 matchPattern=^0815440096 replacePattern=296 replaced pattern=296
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=296 xlt_type=unknown xlt_plan=unknown
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=1
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: No profile found in peer 20001 for outgoing direction
    Mar  9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: calling_number=90819262424 calling_octet=0x0
            called_number=296 called_octet=0x0
            redirect_number= redirect_type=0 redirect_plan=0        redirect_PI=-1 redirect_SI=-1
    Mar  9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing direction
    Mar  9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: calling_number=90819262424 calling_octet=0x0
            called_number=296 called_octet=0x0
            redirect_number= redirect_type=0 redirect_plan=0
    Mar  9 07:45:18.195 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=2
    debug voice dialpeer detail
    =====================
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
      Dial String=0815440096, Expanded String=0815440096, Calling Number=0815440096T
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=2000 Is Matched
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=20005 Is Matched
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=0819262424T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=0819262424T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Incoming Dial-peer=2000 Is Matched
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=0819262424T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=0819262424T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Incoming Dial-peer=2000 Is Matched
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
      Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected]:5060
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
      Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected]:5061
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=0815440096, Expanded String=0815440096, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=0819262424T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=0819262424T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.576 METD: //-1/D8245087848D/DPM/MatchNextPeer:
       Result=Success(0); Incoming Dial-peer=2000 Is Matched
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=296, Expanded String=296, Calling Number=296T
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=20001 Is Matched
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ANSWER; Calling Number=296
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=296T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=296T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Incoming Dial-peer=20001 Is Matched
    Mar  9 07:49:25.584 METD: //-1/D8245087848D/DPM/dpMatchCore:
       Dial String=296, Expanded String=296, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.584 METD: //-1/D8245087848D/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=20001 Is Matched
    Mar  9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=90819262424, Expanded String=90819262424, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=0819262424, Expanded String=0819262424, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=2000 Is Matched
    Mar  9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ANSWER; Calling Number=296
    Mar  9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=296T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=296T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Incoming Dial-peer=20001 Is Matched
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=296, Expanded String=296, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=20001 Is Matched
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=90819262424, Expanded String=90819262424, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=90819262424, Expanded String=90819262424, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=296, Expanded String=296, Calling Number=
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    Mar  9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
       Result=Success(0); Outgoing Dial-peer=20001 Is Matched

  • Third Party Phone over SIP Trunk with CUCM 9.x

    Hi all,
    I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
    I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
    Cisco Phone: INVITE sip.60xxxx%23@ipadress
    Third Party SIP Phone:  INVITE sip:[email protected]
    It seems the Cisco phones gets some extra configured the Third Party ones dont...
    Thanks in advance for any help.
    //Per

    Thanks for the answer
    Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
    When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
    When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty.  The termination Cause Code is that the number requested is Unallocated/Unassigned..
    In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
    Unfortunatley i dont have the meens to attach the trace...
    Thanks again for any help/advice
    With regards, Per.

  • 3rd Party SIP phone to CUCM via SIP Proxy

    Hi all,
    This is the scenario i'm currently working on :
    3rd party SIP phone <--> Internet <--> SIP Proxy <--> LAN <--> CUCM
    The SIP proxy basically terminates everything (REGISTER, INVITE, etc), including the RTP stream.
    I can register the 3rd party SIP phone to CUCM and in CUCM and  i can see SIP Proxy IP Address as the registered address of the phone.
    Calls from the 3rd party SIP phone to internal Cisco or internal 3rd party SIP phone and vice versa work like charm.
    The only (fatal) problem is i can only register 1 3rd party SIP phone to CUCM via this SIP proxy.
    Since this SIP Proxy always use its internal IP Address and port 5060 (TCP) as its source of registration, CUCM sees multiple registrations for multiple extensions (users) come from a single IP and port, and rejects the second registration request.
    It seems that CUCM binds a digest user to an IP address and port, therefore cannot accept multiple registrations from a single IP and port.
    Can anyone clarify this?  Or is there any way around this?
    I'm using CUCM 8.6.2 and CUCM 9.X (both do not work).
    Regards,
    Christian

    This is most likely because of the following...
    Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
    The REGISTER message includes the following header:
    Authorization: Digest username="xxxxxxxxxx",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
    The username, xxxxxxxxxxx, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
    See the following document.
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
    Also Try this bug CSCef88775

  • CP-6921-C-K9 as SIP phone on CUCM 6?

    Can a CP-6921-C-K9 phone be used as a third party SIP device on CUCM 6?
    We will be upgrading soon to a new version of CUCM where 6921 would be supported but for now can we get them to work on CUCM 6 as SIP phones?
    Regards.

    It looks like the SIP support is even later in the 7.1.x line.  No documented operation as 3rd party SIP that I can find.
    http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-6921/data_sheet_c78-541534.html
    The phones are supported in Cisco Unified Communications Manager and  Cisco Unified Communications Manager Business Edition Versions 7.1.2 and  later using Skinny Client Control Protocol (SCCP) or SIP with Cisco  Unified Communications Manager and Cisco Unified Communications Manager  Business Edition Versions 7.1.5 and later.

  • How to create multiple sip trunks between cucm and cisco unified sip proxy

    Dear Expert,
    Is there a way to create multiple sip trunks between CUCM and Cisco Unified SIP Proxy (CUSP)? How to achieve it without creating multiple IP interfaces on the CUSP module.
    CUCM: 8.5.1.10000-9
    CUSP: 8.5.2
    Thank you,
    .wan

    Hello Michael,
    This SIP trunk is part of UCCE solution, which used between CVP, CUSP, and CUCM.
    The requirements:
    1) To have different codecs for different type of calls, as the phones are at few countries
    2) To pass different number of digits from CUSP to CUCM for different call treatments
    .wan

  • Modify calling number in SIP invite on CUCM 10.5

    Hello,
    I am working at a customer with CUCM 10.5 who uses MGCP gateways to access the PSTN via T1 PRI ISDN.
    They use four digit DNs internally and need to prefix these with 713657 to make the outbound CLID work ok - i.e. a call to the PSTN from extension 1000 needs to send 7136571000 to the ISDN provider.
    I configured this using Calling Party Transformations and this works fine e.g.
    A Calling Party Transformation for 1XXX would prefix 713657.
    The problem I have is that the customer has a NICE active recording system which communicates with the CUCM cluster using a SIP trunk.
    The invites that CUCM sends via the SIP trunk show the full ten digits rather than the four digit extension which will not work according to company deploying the recording system.
    If I remove the Calling Party Transformation then the SIP invite shows four digits and the call recording works but the outbound CLID does not work.
    Can anyone suggest a way to fix this? The customer does not want to change the gateway protocol from MGCP to H323 which would be my favoured choice. Any change of calling party setting on CUCM (e.g. ticking the use external mask for calling party on route pattern) affects the SIP invite.
    Ideally I need a way to modify the number in the SIP invite but I cannot find any example of how to do this.
    Any suggestions are welcome.
    Thanks

    Hi, thanks for your response.
    The Calling Party Transformation CSS is not applied to the SIP trunk but is applied to the T1 port of the MGCP gateway.
    The Transformations are still applied to the SIP Invite messages via the trunk so I guess this is a quirk of the calling recording profile setup on CUCM.
    I did try creating another Calling Party Transformation setup which stripped the unwanted digits and applied it to the SIP trunk but it had no effect.

  • SIP incoming call with G722-64 codec not working

    Hi, Guys.
    Have setup cube sip trunk to ITSP, incoming and outgoing calls are working. Except for an incoming call with g722 codec and video h263 (just need voice call). The called number does not even ring. The caller informed that his using polycom phone.
    Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar)  but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.
    Thanks

    I have looked at your logs and here are my observations..
    1. When you disabled fast start on CUCM, I asked you to enable early offer on your CUBE, however I dont see this in your logs..
    This is the INVITE sent to your ITSP, as you can see, this doesnt contain any SDP, that suggest you are doing delayed offer..
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC862F
    From: <sip:[email protected]>;tag=AE7F464-1B0F
    To: <sip:[email protected]>
    Date: Thu, 03 Apr 2014 07:33:09 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 0011462194-3037647155-0083893506-2887478836
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1396510389
    Contact: <sip:[email protected]:5060>
    History-Info: <sip:[email protected]:5060>;index=1,<sip:[email protected]:5060>;index=2
    Expires: 300
    Allow-Events: telephone-event
    Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
    Content-Length: 0
    2. If you are doing DO, then your CUBE needs to send an answer to what your ITSP is offering in its ACK..but this is not happening
    Here is what I see..Your CUBE sends SDP in its PRACK
    Sent:
    PRACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC9232B
    From: <sip:[email protected]>;tag=AE7F464-1B0F
    To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
    Date: Thu, 03 Apr 2014 07:33:09 GMT
    Call-ID: [email protected]
    CSeq: 103 PRACK
    RAck: 323009643 102 INVITE
    Allow-Events: telephone-event
    Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]2.147.134.21:5060;transport=udp",response="a9d772d988ec971cdad556fd4a992bd0",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="58621262",qop=auth,algorithm=MD5,nc=00000002
    Max-Forwards: 70
    Content-Type: application/sdp
    Content-Length: 293
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1079 7198 IN IP4 172.21.8.134
    s=SIP Call
    c=IN IP4 172.21.8.134
    t=0 0
    m=audio 17082 RTP/AVP 8 96 100
    c=IN IP4 172.21.8.134
    a=rtpmap:8 PCMA/8000
    a=rtpmap:96 telephone-event/8000
    a=fmtp:96 0-16
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 192-194
    a=ptime:20
    ###Here is your ACK to the 200 OK from ITSP###
    On the ACK...Your CUBE doesnt include any SDP in its ACK, hence your ITSP disconnected the call immediately
    ACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKCA1335
    From: <sip:[email protected]>;tag=AE7F464-1B0F
    To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
    Date: Thu, 03 Apr 2014 07:33:09 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
    Allow-Events: telephone-event
    Content-Length: 0
    017151: Apr  3 07:33:11.089 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    BYE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 202.147.134.21:5060;branch=z9hG4bKahsb3f108gbhq8pbn5k1sdj0gkbf0.1
    From: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
    To: <sip:[email protected]>;tag=AE7F464-1B0F
    Call-ID: [email protected]
    CSeq: 323009054 BYE
    Max-Forwards: 9
    Content-Length: 0
    I have two suggestions..
    1. Downgrade or upgrade your CUBE IOS. Something is not quite right with this behaviour
    2. Send your full sh run
    On your inbound call issue, you need to send me the logs for a call to another of your DDI..

  • Prefixing a 9 and 91 to incoming calls from SIP provider for callback

    I am wondering what would be the best options  for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
    callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
    I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
    would this work?
    voice-translation rule 1
    rule 1 // /9/
    voice-translation profile prefix_9
    translate calling 1
    dial-peer voice 101 voip
    destination-pattern ???????...$
    voice-class codec 1
    session protocol sipv2
    session target ipv4: to callmanager
    incoming called-number .
    dtmf-relay rtp-nte
    dial-peer voice 1001 voip
    translation profile incoming prefix_9
    destination-pattern T
    session protocol sipv2
    session target ipv4: to sip provider
    incoming called-number ???????...$
    dtmf-relay rtp-nte

    Your config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
    Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
    HTH,
    Chris

  • Incoming calls issue in Third Party SIP Phone

    Hi,
    Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
    Thanks

    Dear Manish,
    Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI  trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
    CallingPartyNumber=5033
    |DialingPartition=
    |DialingPattern=5030
    |FullyQualifiedCalledPartyNumber=5030
    |DialingPatternRegularExpression=(5030)
    |DialingWhere=
    |PatternType=Enterprise
    |PotentialMatches=NoPotentialMatchesExist
    |DialingSdlProcessId=(0,0,0)
    |PretransformDigitString=5030
    |PretransformTagsList=SUBSCRIBER
    |PretransformPositionalMatchList=5030
    |CollectedDigits=5030
    |UnconsumedDigits=
    |TagsList=SUBSCRIBER
    |PositionalMatchList=5030
    |VoiceMailbox=
    |VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
    |VoiceMailPilotNumber=7103
    |RouteBlockFlag=RouteThisPattern
    |RouteBlockCause=0
    |AlertingName=Syed Ahmer
    |UnicodeDisplayName=Syed Ahmer
    |DisplayNameLocale=1
    |OverlapSendingFlagEnabled=0
    12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
    [23928282,NET]
    INVITE sip:[email protected]:5062 SIP/2.0
    Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
    From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
    To:
    Date: Thu, 30 Jan 2014 07:17:38 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.5
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence
    Send-Info: conference, x-cisco-conference
    Alert-Info:
    Contact:
    Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
    Max-Forwards: 70
    Content-Length: 0
    |14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
    12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*

  • No incoming calls CUCM

    I am trying to configure a CUCM with a SIP trunk to a 2811 and a voice GW to my SIP trunk provider.
    CUCM8.6 <SIP>2811<SIP> Callcentric.
    I am able to make outgoing calls but am failing miserably with incoming.
    I suspect it is my incoming dial peer.
    The incoming calls hit my 2811 but do not seem to go to my CUCM.
    I have attached an output from my "debug ccsip calls"
    Anything help would be greatly appreciated.

    Robert,
    surely that is not all the sip debug information, I am missing the INVITES, TRYING etc SIP messages, can you re-attach and maybe also debug your dial peers to see what gets hit (if anything at all ) when making an inbound call.
    Cheers
    =============================
    Please remember to rate useful posts, by clicking on the stars below. 
    =============================

  • Cisco 2911 Voice Gateway SIP PSTN Calls Fail

    Hello All,
        I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway.  2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy.  Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below).  does anyone have any insight on how to correct this?  Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call.  Thanks in advance for any help!!
    From: <sip:[email protected]>:tag=6166CDC4-882
    To: <sip:[email protected]>
    Shawn C. Smith

    i have same problem my cucm ip is 192.168.200.53
    my Voice Gateway is SIP by ip 192.168.200.86 for internal
    and 172.29.7.94
    and my SIP Server is 10.208.9.69
    if its oky can yuo take a look at my problem please
    this is the syslog from debug
    May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
    Session-Expires:  1800
    P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
    Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
       cc_api_call_setup_ind_common:
       cisco-username=2217156
       ----- ccCallInfo IE subfields -----
       cisco-ani=2217156
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=1
       dest=90555769123
       cisco-desttype=0
       cisco-destplan=0
       cisco-rdie=FFFFFFFF
       cisco-rdn=
       cisco-rdntype=0
       cisco-rdnplan=0
       cisco-rdnpi=-1
       cisco-rdnsi=-1
       cisco-redirectreason=-1   fwd_final_type =0
       final_redirectNumber =
       hunt_group_timeout =0
    May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
       Interface=0x30CF41D4, Call Info(
       Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=90555769123(TON=Unknown, NPI=Unknown),
       Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
       Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :cc_get_feature_vsa malloc success
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288:  cc_get_feature_vsa count is 1
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
       Set Up Event Sent;
       Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=90555769123(TON=Unknown, NPI=Unknown))
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
       Event=0x2B82D890
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
       Try with the demoted called number 90555769123
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
       Context=0x2ABC2E44
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
       >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
       Progress Indication=NULL(0)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
       Destination=, Calling IE Present=TRUE, Mode=0,
       Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
       In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
       Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
       Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
       Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=0555769123(TON=Unknown, NPI=Unknown),
       Redirect Number=, Display Info=Aysar Mohamed
       Account Number=2217156, Final Destination Flag=TRUE,
       Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
       ccCallSetupRequest:
       cisco-username=2217156
       ----- ccCallInfo IE subfields -----
       cisco-ani=2217156
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=1
       dest=0555769123
       cisco-desttype=0
       cisco-destplan=0
       cisco-rdie=FFFFFFFF
       cisco-rdn=
       cisco-rdntype=0
       cisco-rdnplan=0
       cisco-rdnpi=-1
       cisco-rdnsi=-1
       cisco-redirectreason=-1   fwd_final_type =0
       final_redirectNumber =
       hunt_group_timeout =0
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
       Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
       Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
       Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :cc_get_feature_vsa malloc success
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288:  cc_get_feature_vsa count is 2
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
    May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
       SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
    May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
       Context=0x2ABC1984
    May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
       Outgoing Dial-peer=802
    May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
       Interface=0x30CF41D4, Progress Indication=NULL(0)
    May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
    From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1401481174
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: kpml, telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Length: 0
    May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Call-ID: [email protected]
    From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>
    CSeq: 101 INVITE
    Content-Length: 0
    May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
    Call-ID: [email protected]
    From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
    CSeq: 101 INVITE
    Contact: <sip:[email protected]:5060;user=phone>
    Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
    Content-Length: 328
    Content-Type: application/sdp
    v=0
    o=- 17192647 17192647 IN IP4 10.208.9.69
    s=SBC call
    c=IN IP4 10.208.9.69
    t=0 0
    m=audio 39910 RTP/AVP 8 0 102 102 18 116
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:102 AMR/8000
    a=rtpmap:102 AMR/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:116 telephone-event/8000
    a=ptime:5
    a=fmtp:116 0-15
    a=fmtp:18 annexb=yes
    May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
       Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
       Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
       Modem=0x0, Codec Bytes=160, Signal Type=2)
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
       Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
       Playout Max=1000(ms), Fax Nom=300(ms))
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
       Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
       Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
       Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
       Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
       Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
       Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event=170, Call Id=466
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event Is Sent To Conferenced SPI(s) Directly
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event=98, Call Id=466
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event Is Sent To Conferenced SPI(s) Directly
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
       Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
       Cause Value=0
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
       Call Entry(Responsed=TRUE)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
       Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
       Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
       Call Entry(Responsed=TRUE)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
       (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
    May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
       (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
       Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
    May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
    May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                        ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                        tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                        tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
       delay media to slow start case, codec negotation is not done
    May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
       Destination Call Id=466, Disposition=0x0, Tag=0x0
    May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
       Call Entry(Conference Id=0x16, Destination Call Id=466)
    May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
       Call Entry(Conference Id=0x16, Destination Call Id=465)
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
       Conference Id=0x16, Call Id1=465, Call Id2=466
    May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>;tag=739628-1BDB
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 233
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
    s=SIP Call
    c=IN IP4 192.168.200.86
    t=0 0
    m=audio 18288 RTP/AVP 8 0 19
    c=IN IP4 192.168.200.86
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:19 CN/8000
    May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 500 Server Internal Error
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
    Call-ID: [email protected]
    From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
    CSeq: 101 INVITE
    Reason: Q.850;cause=127;text="interworking unspecified"
    Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
    Content-Length: 0
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
       Cause Value=41, Interface=0x30CF41D4, Call Id=466
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
       Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
       release reserved xcoding resource.
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
       Accounting=0, Call Id=466
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
       Conference Id=0x16, Tag=0x0
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
       Destination Call Id=466, Disposition=0x0, Tag=0x0
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: kpml, telephone-event
    Content-Length: 0
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
       Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Call Disconnect Event Sent
    May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
    May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.684:  vsacount in free is 1
    May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
    May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
    May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 503 Service Unavailable
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>;tag=739628-1BDB
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=41
    Content-Length: 0
    May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>;tag=739628-1BDB
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Length: 0
    May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
       Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
    May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Call Disconnect Event Sent
    May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
    May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.688:  vsacount in free is 0
    May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    OPTIONS sip:172.29.7.94:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
    Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
    From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
    To: <sip:172.29.7.94>
    CSeq: 1 OPTIONS
    Max-Forwards: 70
    Content-Length: 0
    May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
    From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
    To: <sip:172.29.7.94>;tag=739BBC-1CE2
    Date: Fri, 30 May 2014 20:19:36 GMT
    Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 1 OPTIONS
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Accept: application/sdp
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Content-Type: application/sdp
    Content-Length: 446
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
    s=SIP Call
    c=IN IP4 172.29.7.94
    t=0 0
    m=audio 0 RTP/AVP 18 0 8 9 4 2 15
    c=IN IP4 172.29.7.94
    m=image 0 udptl t38
    c=IN IP4 172.29.7.94
    a=T38FaxVersion:0
    a=T38MaxBitRate:9600
    a=T38FaxFillBitRemoval:0
    a=T38FaxTranscodingMMR:0
    a=T38FaxTranscodingJBIG:0
    a=T38FaxRateManagement:transferredTCF
    a=T38FaxMaxBuffer:200
    a=T38FaxMaxDatagram:320
    a=T38FaxUdpEC:t38UDPRedundancy
    My SIP GW internal ip address is 192.168.200.86
    and the Public IP is : 172.29.7.94
    My CUCM is 192.168.200.53
    my GW Config is :
    voice service voip
     allow-connections h323 to h323
     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
     sip
      registrar server
    voice class codec 1
     codec preference 1 g711alaw
     codec preference 2 g711ulaw
     codec preference 3 g729r8
     codec preference 4 g729br8
    voice translation-rule 3
     rule 1 /^9\(\)/ /\1/
    voice translation-rule 4
     rule 4 /^22217/ /7/
     rule 5 /^2217/ /7/
     rule 6 /^022217/ /7/
     rule 7 /^0122217/ /7/
    voice translation-rule 5
     rule 1 /^5/ /905/
     rule 2 /^1/ /901/
     rule 3 /^2/ /902/
     rule 4 /^3/ /903/
     rule 5 /^4/ /904/
     rule 6 /^6/ /906/
     rule 7 /^7/ /907/
     rule 8 /^8/ /908/
     rule 10 /^00/ /900/
     rule 11 /'+'/ /900/
    voice translation-profile OUT
     translate called 3
    voice translation-profile REDIAL
     translate calling 5
    voice translation-profile SIP-NEW
     translate called 4
    application
     service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
     service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
    license udi pid CISCO2921/K9 sn FCZ164960G0
    hw-module pvdm 0/0
    hw-module pvdm 0/1
    interface Embedded-Service-Engine0/0
     no ip address
     shutdown
    interface GigabitEthernet0/0
     description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
     ip address 192.168.200.86 255.255.255.0
     duplex auto
     speed auto
    interface GigabitEthernet0/1
     ip address 172.29.7.94 255.255.255.252
     duplex auto
     speed auto
    ip http server
    ip http access-class 23
    ip http authentication local
    no ip http secure-server
    ip http timeout-policy idle 60 life 86400 requests 10000
    ip route 0.0.0.0 0.0.0.0 192.168.200.1
    ip route 10.208.9.0 255.255.255.0 172.29.7.93
    access-list 23 permit 10.10.10.0 0.0.0.7
    control-plane
    mgcp profile default
    sccp local GigabitEthernet0/0
    sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
    sccp
    sccp ccm group 1
     associate ccm 1 priority 1
     associate profile 2 register NAGHI-MTP
    dspfarm profile 2 mtp
     codec g711alaw
     maximum sessions hardware 25
     associate application SCCP
    dial-peer voice 802 voip
     description ** SIP TO STC **
     translation-profile outgoing OUT
     destination-pattern 9T
     session protocol sipv2
     session target ipv4:10.208.9.69:5060
     session transport udp
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay sip-notify rtp-nte sip-kpml
     no vad
    dial-peer voice 811 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 022217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 812 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 22217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 813 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 2217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 814 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 022217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 815 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 22217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 816 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 2217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 817 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 0122217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 818 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 0122217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    Please i need ur help ASAP

  • CUCME Not Incoming Calls, Outgoing calls ok

    Hello everybody,
    i am configuring a CUCME with SIP trunk, i can make calls to outside but i can´t recieve any from outside, this is my second time a configure with SIP
    i´ve used the command debug voice dialpeer all to check was going on, but i can´t find the problem.
    this is my config:
    ip host sip-server A.B.C.D
    voice service voip
    ip address trusted list
      ipv4 A.B.C.D 255.255.255.252
      voice translation-rule 1
    rule 1 /325277\(\)/ /1\1/
    voice translation-profile IN
    translate called 1
    dial-peer voice 1 voip
    description **Incoming Call from SIP Trunk**
    translation-profile incoming IN
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    ephone-dn  1
    number 100
    description RECEPTION
    ephone  2
    mac-address AAAA.BBBB.CCCC
    ephone-template 1
    type 7942
    keep-conference
    button  1:1
    NOTE: IP Address are hidden, just for security
    These are the output of my debug/tests:
    #test voice translation-rule 1 32527700
    Matched with rule 1
    Original number: 32527700       Translated number: 100
    Original number type: none      Translated number type: none
    Original number plan: none      Translated number plan: none
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=32527700
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=32527700, Expanded String=32527700, Calling Number=32527700T
       Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
       dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=NO_MATCH(-1)
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=59513212, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ANSWER; Calling Number=59513212
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=59513212T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_ORIGINATE; Calling Number=59513212
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=59513212T
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=NO_MATCH(-1) After All Match Rules Attempt
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
       dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
    *Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:exit@6704
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_VIA_URI; URI=sip:A.B.C.D:5060
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060;user=phone
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected];user=phone
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected];user=phone
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Dial String=, Expanded String=, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Result=-1
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=32527700
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
       Is Incoming=TRUE, Number Expansion=FALSE
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
       Dial String=32527700, Expanded String=32527700, Calling Number=
       Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:
       Result=Success(0); Incoming Dial-peer=1 Is Matched
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:
       dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
    *Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:exit@6655
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    Thanks in Advance!!!

    Thanks, these are the output
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:32527700@(WAN):5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
    Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
    From: ;tag=6e8b9968-CC-25
    To:
    CSeq: 1 INVITE
    Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
    Max-Forwards: 70
    Supported: 100rel,timer
    User-Agent: Huawei SoftX3000 V300R601
    Session-Expires: 300
    Min-SE: 90
    Contact:
    Content-Length: 376
    Content-Type: application/sdp
    v=0
    o=HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)
    s=Sip Call
    c=IN IP4 (SIP_SERVER)
    t=0 0
    m=audio 11554 RTP/AVP 8 0 18 4 2 98 98 98
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:98 G726-40/8000
    a=rtpmap:98 G726-32/8000
    a=rtpmap:98 G726-24/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=32527700
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
       dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=NO_MATCH(-1)
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=59513212, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=NO_MATCH(-1) After All Match Rules Attempt
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
       dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
    *Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
       Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
    *Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpMatchSafModulePlugin:
       dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 422 Session Timer too small
    Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
    From: ;tag=6e8b9968-CC-25
    To: ;tag=4CD1E84-2094
    Date: Wed, 29 Jan 2014 22:53:19 GMT
    Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Min-SE:  1800
    Server: Cisco-SIPGateway/IOS-15.2.4.M3
    Content-Length: 0
    *Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:32527700@(WAN):5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
    Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
    From: ;tag=6e8b9968-CC-25
    To: ;tag=4CD1E84-2094
    CSeq: 1 ACK
    Max-Forwards: 70
    Content-Length: 0
    *Jan 29 16:53:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    REGISTER sip:(SIP_SERVER):5060 SIP/2.0
    Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
    From: ;tag=4CD4D7C-1634
    To:
    Date: Wed, 29 Jan 2014 22:53:31 GMT
    Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
    Max-Forwards: 70
    Timestamp: 1391036011
    CSeq: 66 REGISTER
    Contact:
    Expires:  3600
    Supported: path
    Content-Length: 0
    *Jan 29 16:53:31: //973/000000000000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 400 Bad Request
    Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
    Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
    From: ;tag=4CD4D7C-1634
    To: ;tag=f2056e8e
    CSeq: 66 REGISTER
    Content-Length: 0
    I´ve replaced the IP Adress for (SIP_SERVER) / (WAN) / SIP_SERVER_INTERNAL
    Thank you

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    Hi Guys,
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    service timestamps debug datetime msec
    service timestamps log datetime msec
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    boot-end-marker
    logging message-counter syslog
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    allow-connections sip to h323
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    name SIP
    label SIP
    voice register pool  1
    id mac 000F.902B.40E0
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    number 1 dn 1
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    ip nat outside
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    speed auto
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    no ip address
    ip nat inside
    ip virtual-reassembly
    duplex auto
    speed auto
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    description DATA
    encapsulation dot1Q 10
    ip address 10.10.10.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
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    description VOICE
    encapsulation dot1Q 20
    ip address 192.168.1.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    ip forward-protocol nd
    ip route 0.0.0.0 0.0.0.0 194.12.0.221
    ip http server
    ip http authentication local
    no ip http secure-server
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    access-list 1 permit 10.10.10.0 0.0.0.255
    tftp-server flash:P003-8-12-00.bin
    tftp-server flash:P003-8-12-00.sbn
    tftp-server flash:P0S3-8-12-00.loads
    tftp-server flash:P0S3-8-12-00.sb2
    tftp-server flash:P003-8-12-00
    tftp-server flash:P003-8-12-00.loads
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    control-plane
    mgcp behavior g729-variants static-pt
    dial-peer cor custom
    dial-peer voice 2 voip
    description Outgoing Geographic
    translation-profile outgoing DiscardDigit9
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    session protocol sipv2
    session target dns:sip.cloudcalling.co.uk
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 1 voip
    description IncomingSIP
    translation-profile incoming IncomingSIP
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.cloudcalling.co.uk
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    You my friend are a star! worked straight away, many thanks.  Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
    The new working config is below with your suggestion, which works!
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    service timestamps debug datetime msec
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    boot-start-marker
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    logging message-counter syslog
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    allow-connections sip to h323
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    codec preference 2 g711ulaw
    codec preference 3 g711alaw
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    max-dn 144
    max-pool 42
    load 7960-7940 P0S3-8-12-00
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    create profile sync 0015244443466064
    voice register dn  1
    number 6999
    allow watch
    name SIP
    label SIP
    voice register pool  1
    id mac 000F.902B.40E0
    type 7960
    number 1 dn 1
    dtmf-relay sip-notify
    username cisco password cisco
    codec g711ulaw
    voice translation-rule 1
    rule 1 /^6...$/ /4143*002/
    voice translation-rule 3
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    rule 2 /^1133501788$/ /6999/
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    translate called 3
    voice translation-profile Translatetrunk
    translate calling 1
    voice-card 0
    no dspfarm
    username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
    archive
    log config
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    interface FastEthernet0/0
    ip address 194.12.0.222 255.255.255.252
    ip nat outside
    ip virtual-reassembly
    duplex auto
    speed auto
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    no ip address
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    duplex auto
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    interface FastEthernet0/1.10
    description DATA
    encapsulation dot1Q 10
    ip address 10.10.10.1 255.255.255.0
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    encapsulation dot1Q 20
    ip address 192.168.1.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    ip forward-protocol nd
    ip route 0.0.0.0 0.0.0.0 194.12.0.221
    ip http server
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    no ip http secure-server
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    access-list 1 permit 192.168.1.0 0.0.0.255
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    tftp-server flash:P003-8-12-00.bin
    tftp-server flash:P003-8-12-00.sbn
    tftp-server flash:P0S3-8-12-00.loads
    tftp-server flash:P0S3-8-12-00.sb2
    tftp-server flash:P003-8-12-00
    tftp-server flash:P003-8-12-00.loads
    tftp-server flash:P003-8-12-00.sb2
    tftp-server flash:SIP000F902B40E0.cnf.xml
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    dial-peer voice 1 voip
    description IncomingSIP
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    incoming called-number .T
    dtmf-relay sip-notify rtp-nte
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    dial-peer voice 2 voip
    description Outgoing Geographic
    translation-profile outgoing Translatetrunk
    destination-pattern 0[7]........
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.cloudcalling.co.uk
    dtmf-relay rtp-nte
    no vad
    sip-ua
    credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
    authentication username 4143*002 password 7 password
    nat symmetric role passive
    nat symmetric check-media-src
    calling-info sip-to-pstn number set 4143*002
    no remote-party-id
    retry invite 3
    retry register 3
    timers connect 100
    registrar dns:sip.cloudcalling.co.uk expires 60
    sip-server dns:sip.cloudcalling.co.uk
      host-registrar
    gatekeeper
    shutdown
    telephony-service
    load 7960-7940 P0S3-8-12-00
    max-ephones 24
    max-dn 30
    ip source-address 192.168.1.1 port 2000
    max-conferences 8 gain -6
    web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
    transfer-system full-consult
    create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
    line con 0
    line aux 0
    line vty 0 4
    login
    scheduler allocate 20000 1000
    ntp server 85.119.80.232
    end
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