Configure Cisco CME auto-attendant to forward call after ext is dialed

                   I am trying to configure my CME if the auto attendant picks up and a users extension is dialed after hours to forward that call to the number configured for that extension, currently if the extension is dialed the call is not forwarded.
Thank you,
-Tom

Hi,
I have same or similar issue, when a person call my company and press option 3 for example (after hour) the call get forward to a manager mobile number, my question is how can I disable the forwarding from CUCM? the OS i use is
8.6.2.22900-9
Many thanks

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    cptone AU
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    cptone AU
    voice-port 0/0/3
    cptone AU
    voice-port 0/1/0
    cptone AU
    voice-port 0/1/1
    cptone AU
    voice-port 0/2/0
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    connection plar opx 398
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    voice-port 0/2/2
    connection plar opx 398
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    caller-id enable
    voice-port 0/2/3
    connection plar opx 398
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    caller-id enable
    voice-port 0/4/0
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    signal immediate
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    sccp local Loopback0
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    sccp
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    associate profile 1 register confprof1
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    codec g711ulaw
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    translation-profile outgoing PSTN_CallForwarding
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    session protocol sipv2
    session target ipv4:10.1.10.1
    voice-class sip outbound-proxy ipv4:10.1.10.1
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    session protocol sipv2
    session target ipv4:10.1.10.1
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    dial-peer voice 90 pots
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    description International number
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    dial-peer voice 30 pots
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    night-service day Fri 17:00 09:00
    night-service day Sat 17:00 09:00
    create cnf-files version-stamp 7960 Dec 23 2013 10:55:20
    ephone-template 15
    softkeys remote-in-use Newcall
    softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
    softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
    softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
    button-layout 7931 2
    ephone-template 16
    softkeys remote-in-use Newcall
    softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
    softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
    softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
    ephone-template 17
    softkeys remote-in-use CBarge Newcall
    softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
    softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
    softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
    ephone-template 18
    softkeys remote-in-use CBarge Newcall
    softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
    softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
    softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
    button-layout 7931 2
    ephone-dn 5 dual-line
    number 301 no-reg primary
    label 301
    description PhoneA Analog
    name PhoneA Analog
    ephone-dn 6 dual-line
    number 302 no-reg primary
    label 302
    description PhoneB Analog
    name PhoneB Analog
    ephone-dn 7 dual-line
    number 303 no-reg primary
    label 303
    description PhoneC Analog
    name PhoneC Analog
    ephone-dn 8 dual-line
    number 304 no-reg primary
    label 304
    description PhoneD Analog
    name PhoneD Analog
    ephone-dn 9
    number BCD no-reg primary
    description MoH
    moh out-call ABC
    ephone-dn 10 dual-line
    number 201 no-reg primary
    pickup-group 1
    label 201
    description Extension 201
    name Receptionist Receptionist
    mobility
    call-forward busy 300
    call-forward noan 300 timeout 20
    ephone-dn 11 dual-line
    number 207 no-reg primary
    label 207
    description Extension 207
    name None None
    ephone-dn 12 dual-line
    call-waiting ring
    number 203 no-reg primary
    pickup-group 1
    label 203
    description Extension 203
    name Peter Steve
    call-forward busy 300
    call-forward noan 300 timeout 15
    huntstop channel
    ephone-dn 13 dual-line
    call-waiting ring
    number 204 no-reg primary
    pickup-group 1
    label 204
    description Extension 204
    name Tim OConnor
    call-forward busy 300
    call-forward noan 300 timeout 20
    huntstop channel
    ephone-dn 14 dual-line
    number 205 no-reg primary
    pickup-group 1
    label 205
    description 205
    name 205
    ephone-dn 15 dual-line
    number 206 no-reg primary
    pickup-group 1
    label 206
    description 206
    name 206
    ephone-dn 16 dual-line
    call-waiting ring
    number 202 no-reg primary
    pickup-group 1
    label 202
    description Extension 202
    name David Holmes
    call-forward busy 300
    call-forward noan 300 timeout 15
    huntstop channel
    ephone-dn 17 dual-line
    number 208 no-reg primary
    label 208
    description 208
    name 208
    ephone-dn 18 dual-line
    number 209 no-reg primary
    label 209
    description 209
    name 209
    ephone-dn 19 dual-line
    number 210 no-reg primary
    label 210
    description 210
    name 210
    ephone-dn 43 octo-line
    number 771 no-reg primary
    conference meetme
    preference 3
    ephone-dn 44 octo-line
    number 771 no-reg primary
    conference meetme
    preference 2
    no huntstop
    ephone-dn 45 octo-line
    number 771 no-reg primary
    conference meetme
    preference 1
    no huntstop
    ephone-dn 46 octo-line
    number 771 no-reg primary
    conference meetme
    no huntstop
    ephone-dn 49 octo-line
    number C001 no-reg primary
    conference ad-hoc
    preference 3
    ephone-dn 50 octo-line
    number C001 no-reg primary
    conference ad-hoc
    preference 2
    no huntstop
    ephone-dn 51 octo-line
    number C001 no-reg primary
    conference ad-hoc
    preference 1
    no huntstop
    ephone-dn 52 octo-line
    number C001 no-reg primary
    conference ad-hoc
    no huntstop
    ephone-dn 55
    number A801... no-reg primary
    mwi off
    ephone-dn 56
    number A800... no-reg primary
    mwi on
    ephone 1
    device-security-mode none
    mac-address 4142.4DB8.0000
    ephone-template 16
    max-calls-per-button 2
    type anl
    button 1:5
    ephone 2
    device-security-mode none
    mac-address 4142.4DB8.0001
    ephone-template 16
    max-calls-per-button 2
    type anl
    button 1:6
    ephone 3
    device-security-mode none
    mac-address 4142.4DB8.0002
    ephone-template 16
    max-calls-per-button 2
    type anl
    button 1:7
    ephone 4
    device-security-mode none
    mac-address 4142.4DB8.0003
    ephone-template 16
    max-calls-per-button 2
    type anl
    button 1:8
    ephone 5
    device-security-mode none
    mac-address 0024.97AA.E811
    ephone-template 15
    max-calls-per-button 2
    username "Receptionist" password receptionist
    type 7931
    button 1:10
    --More--                    !
    ephone 6
    device-security-mode none
    mac-address 0024.C4FC.4013
    ephone-template 16
    username "None"
    type 7911
    button 1:11
    ephone 7
    device-security-mode none
    video
    mac-address 000F.34FA.168B
    ephone-template 16
    username "steve" password petersteve
    speed-dial 1 xxx label "Peter - Home"
    speed-dial 2 xxx label "David - Mobile"
    speed-dial 3 xxx label "Tim - Mobile AUS"
    speed-dial 4 xxx label "Tim - Mobile USA"
    type 7960
    button 1:12
    ephone 8
    device-security-mode none
    video
    mac-address A40C.C394.B1F0
    ephone-template 16
    username "tim" password timoconnor
    speed-dial 1 xxx label "David - Mobile"
    speed-dial 2 xxx label "Peter - Mobile"
    speed-dial 3 xxx label "Clare - Mobile"
    type 7911
    button 1:13
    ephone 9
    device-security-mode none
    mac-address 0024.C4FC.5425
    ephone-template 16
    type 7911
    button 1:14
    ephone 10
    device-security-mode none
    mac-address 0024.C4FD.E27C
    ephone-template 16
    type 7911
    button 1:15
    ephone 11
    device-security-mode none
    video
    mac-address 0007.5098.1AB6
    ephone-template 16
    username "holmes" password davidholmes
    speed-dial 1 xx label "David - Home"
    speed-dial 2 xxxl abel "Sue - Mobile"
    speed-dial 3 xxx label "Peter - Mobile"
    speed-dial 4 xxx label "Tim - Mobile USA"
    speed-dial 5 xxx label "Tim - Mobile AUS"
    type 7960
    button 1:16
    ephone-hunt 1 sequential
    pilot 501
    list 202, 203, 204
    final 300
    timeout 8, 8, 8
    no-reg pilot
    statistics collect
    description Sales
    alias exec cca_vm_notification schedule from_time=00 to_time=24
    banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Nov 15 22:54:23 EST 2013^C
    line con 0
    no modem enable
    line aux 0
    line 2
    no activation-character
    no exec
    transport preferred none
    transport input all
    line vty 0 4
    transport input all
    line vty 5 100
    transport input all
    ntp master
    end
    UC520#

    I was configure custom disconnect tone refer to this site:
    http://ciscoflair.blogspot.com/2009/05/cisco-fxo-disconnect-issue.html
    And the tone is in the attachment, and the custom disconnect tone configuration like below:
    voice class custom-cptone Disconnect
    dualtone disconnect
      frequency 420 420
      cadence 251 255 245 250 249 250 250 250
    and the port configuration was like below:
    voice-port 0/1/2
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone Disconnect
    cptone NL
    timeouts interdigit 4
    timeouts call-disconnect 5
    timeouts wait-release 5
    timing hookflash-out 500
    connection plar 334
    impedance complex2
    caller-id enable
    caller-id alerting line-reversal
    caller-id alerting dsp-pre-allocate
    but it was not working and the phone still ringing after the PSTN caller disconnect.
    but i was read about "dualtone-detect-params", and i was add the below command and i do not understand it, but it was solve hte problem:
    voice class dualtone-detect-params 1
    freq-max-deviation 20
    cadence-variation 50
    so what it is and how to determine this parameters.

  • Cisco CME and Calls through SIP provider

    Hello, friends.
    There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
    Telephones connected to SCCP, registered SIP from the provider.
    When I try to call to test number 4444 through sip in debug I see:
    *Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
    From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
    To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
    Server: kamailio (4.0.3 (x86_64/linux))
    Content-Length: 0
    *Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
    From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
    To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
    Date: Sun, 09 Feb 2014 21:51:25 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Cisco при этом зарегана у провайдера SIP
    DC#show sip-ua register status
    Line peer expires(sec) registered P-Associ-URI
    Configuration:
    voice service voip
    ip address trusted list
      ipv4 178.16.26.122 255.255.255.255
      ipv4 144.76.42.108 255.255.255.255
      ipv4 176.9.145.115 255.255.255.255
      ipv4 5.9.108.25 255.255.255.255
      ipv4 78.46.95.118 255.255.255.255
      ipv4 89.249.23.194 255.255.255.255
      ipv4 178.16.26.124 255.255.255.255
      ipv4 176.9.85.133 255.255.255.255
      ipv4 46.4.53.86 255.255.255.255
      ipv4 5.9.84.165 255.255.255.255
      ipv4 78.16.26.122 255.255.255.255
      ipv4 77.235.62.222 255.255.255.255
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    sip
      registrar server
    voice class codec 1
    codec preference 1 g711ulaw
    codec preference 2 g729r8
    codec preference 3 g711alaw
    voice register global
    max-dn 10
    max-pool 10
    voice register dn  1
    number 150
    voice register dn  2
    number 151
    voice translation-rule 9
    rule 1 /^95/ //
    voice translation-rule 1020
    rule 1 /^.$/ /40232/
    voice translation-profile outgoing
    translate calling 1020
    translate called 9
    mgcp fax t38 ecm
    mgcp profile default
    dial-peer voice 2 voip
    translation-profile outgoing outgoing
    destination-pattern 95....
    session protocol sipv2
    session target sip-server
    voice-class codec 1
    no voice-class sip outbound-proxy
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay rtp-nte
    no vad
    sip-ua
    credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
    authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
    registrar dns:sip.zadarma.com:5060 expires 3600
    sip-server dns:sip.zadarma.com:5060
    connection-reuse
    host-registrar
    DC#show sip-ua register status
    Line                             peer       expires(sec) registered P-Associ-URI
    ================================ ========== ============ ========== ============
    150                              40001      12           no
    40232                            -1         550          yes
    SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
    Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
    Should be so sip:40232@<my ip>
    Please help me!

    Yes, I behind nat.
    *Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
    Max-Forwards: 70
    Contact:
    To: "954444"
    From: "150";tag=7b409f06
    Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: X-Lite release 1104o stamp 56125
    Content-Length: 314
    v=0
    o=- 2 2 IN IP4 192.168.11.14
    s=CounterPath X-Lite 3.0
    c=IN IP4 192.168.11.14
    t=0 0
    m=audio 5724 RTP/AVP 107 0 8 101
    a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
    a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
    a=fmtp:101 0-15
    a=rtpmap:107 BV32/16000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    *Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
    From: "" >;tag=169E6BC4-1E16
    To: [email protected]>
    Date: Mon, 10 Feb 2014 14:11:53 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1392041513
    Contact: outside ip cisco cme:5060>
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 262
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 18534 RTP/AVP 0 8 101
    c=IN IP4 92.63.108.115
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    *Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
    From: "150";tag=7b409f06
    To: "954444"
    Date: Mon, 10 Feb 2014 14:11:53 GMT
    Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    *Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
    From: "" ;tag=169E6BC4-1E16
    To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
    Server: kamailio (4.0.3 (x86_64/linux))
    Content-Length: 0
    *Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
    From: "150" [email protected]>;tag=169E6BC4-1E16
    To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
    Date: Mon, 10 Feb 2014 14:11:53 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    *Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
    From: "" ;tag=169E6BC4-1E16
    To: [email protected]>
    Date: Mon, 10 Feb 2014 14:11:53 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Timestamp: 1392041513
    Contact: :5060>
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
    Max-Forwards: 69
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 262
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 18534 RTP/AVP 0 8 101
    c=IN IP4 92.63.108.115
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    *Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 100 trying -- your call is important to us
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
    From: "" ;tag=169E6BC4-1E16
    To: [email protected]>
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Server: kamailio (4.0.3 (x86_64/linux))
    Content-Length: 0
    *Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
    Record-Route:
    From: "k40232" ;tag=169E6BC4-1E16
    To: [email protected]>;tag=as7e8de8e5
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Server: Zadarma Voip
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact:
    Content-Type: application/sdp
    Content-Length: 281
    v=0
    o=root 1942395501 1942395501 IN IP4 178.16.26.124
    s=Asterisk PBX
    c=IN IP4 178.16.26.124
    t=0 0
    m=audio 12164 RTP/AVP 8 0 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    *Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
    From: "150" [email protected]>;tag=169E6BC4-1E16
    To: [email protected]>;tag=as7e8de8e5
    Date: Mon, 10 Feb 2014 14:11:53 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 102 ACK
    Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
    Allow-Events: telephone-event
    Content-Length: 0
    *Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
    From: "150";tag=7b409f06
    To: "954444";tag=169E6F78-88E
    Date: Mon, 10 Feb 2014 14:11:53 GMT
    Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
    CSeq: 1 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: :5060;transport=tcp>
    Supported: replaces
    Server: Cisco-SIPGateway/IOS-12.x
    Supported: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 193
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 17190 RTP/AVP 8
    c=IN IP4 92.63.108.115
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    *Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
    Max-Forwards: 70
    Contact:
    To: "954444";tag=169E6F78-88E
    From: "150";tag=7b409f06
    Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
    CSeq: 1 ACK
    User-Agent: X-Lite release 1104o stamp 56125
    Content-Length: 0

  • Cisco CME: calls through SIP-provider again

    Hello,friends!
    I have already published a discussion here https://supportforums.cisco.com/discussion/12089656/cisco-cme-and-calls-through-sip-provider and you helped me, everything works well for Russian numbers.
    When I tried to add the configuration for calls to Belarus, again, there was a problem. I do not understand why, although the configuration ideintichnaya.
    My config:
    voice service voip
     ip address trusted list
      ipv4 178.16.26.122 255.255.255.255
      ipv4 144.76.42.108 255.255.255.255
      ipv4 176.9.145.115 255.255.255.255
      ipv4 5.9.108.25 255.255.255.255
      ipv4 78.46.95.118 255.255.255.255
      ipv4 89.249.23.194 255.255.255.255
      ipv4 178.16.26.124 255.255.255.255
      ipv4 176.9.85.133 255.255.255.255
      ipv4 46.4.53.86 255.255.255.255
      ipv4 5.9.84.165 255.255.255.255
      ipv4 78.16.26.122 255.255.255.255
      ipv4 77.235.62.222 255.255.255.255
      ipv4 81.88.86.11 255.255.255.255
      ipv4 192.168.1.50 255.255.255.255
      ipv4 217.150.198.44 255.255.255.255
      ipv4 178.63.96.3 255.255.255.255
      ipv4 178.63.96.28 255.255.255.255
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     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service sip moved-temporarily
     sip
      registrar server
    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g729r8
     codec preference 3 g711alaw
    voice class sip-profiles 20
     request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:[email protected]>"
    voice translation-rule 9
     rule 1 /^98/ /7/
    voice translation-rule 10
     rule 1 /^9/ //
    voice translation-rule 1020
     rule 1 /^.*$/ /141756/
    voice translation-rule 1030
     rule 1 /^.*/ /141756/
    voice translation-rule 1040
     rule 1 /^.*$/ /21/
    voice translation-profile incoming
     translate called 1040
    voice translation-profile outgoing
     translate calling 1030
     translate called 9
    voice translation-profile outgoing-mezhdunarod
     translate calling 1030
     translate called 10
    voice-card 0
    dial-peer voice 2 voip
     description TO-RUSSIA
     translation-profile outgoing outgoing
     preference 1
     destination-pattern 98..........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     no voice-class sip outbound-proxy
     voice-class sip profiles 20
     voice-class sip bind control source-interface FastEthernet0/0
     voice-class sip bind media source-interface FastEthernet0/0
     dtmf-relay rtp-nte sip-notify
     no vad
    dial-peer voice 3 voip
     translation-profile incoming incoming
     incoming called-number 141756
     voice-class codec 1
     voice-class sip bind control source-interface FastEthernet0/0
     voice-class sip bind media source-interface FastEthernet0/0
     dtmf-relay rtp-nte
     no vad
    dial-peer voice 4 voip
     description To-Belarus
     translation-profile outgoing outgoing-mezhdunarod
     destination-pattern 9375.........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     no voice-class sip outbound-proxy
     voice-class sip profiles 20
     voice-class sip bind control source-interface FastEthernet0/0
     voice-class sip bind media source-interface FastEthernet0/0
     dtmf-relay rtp-nte sip-notify
     no vad
    sip-ua
     credentials username 141756 password 7<pass> realm sip.zadarma.com
     authentication username 141756 password 7 <pass>
     no remote-party-id
     registrar 1 dns:sip.zadarma.com expires 3600
     sip-server dns:sip.zadarma.com
     connection-reuse
     host-registrar
    DEBUG ccsip message:
    Jun 17 14:23:09.033: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
    From: "" <sip:[email protected]>;tag=40FCB218-23D7
    To: <sip:[email protected]>
    Date: Tue, 17 Jun 2014 09:23:09 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE: 1800
    Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1402996989
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 309
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 18252 RTP/AVP 0 18 8 101
    c=IN IP4 92.63.108.115
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Jun 17 14:23:09.089: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65;rport=5060
    From: "" <sip:[email protected]>;tag=40FCB218-23D7
    To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1", qop="auth"
    Server: kamailio (4.1.2 (x86_64/linux))
    Content-Length: 0
    Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
    From: "Vankuver" <sip:[email protected]>;tag=40FCB218-23D7
    To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
    Date: Tue, 17 Jun 2014 09:23:09 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
    From: "" <sip:[email protected]>;tag=40FCB218-23D7
    To: <sip:[email protected]>
    Date: Tue, 17 Jun 2014 09:23:09 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE: 1800
    Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1402996989
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 309
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 18252 RTP/AVP 0 18 8 101
    c=IN IP4 92.63.108.115
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Jun 17 14:23:09.637: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
    From: "" <sip:[email protected]>;tag=40FCB218-23D7
    To: <sip:[email protected]>
    Date: Tue, 17 Jun 2014 09:23:09 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE: 1800
    Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1402996989
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 309
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 18252 RTP/AVP 0 18 8 101
    c=IN IP4 92.63.108.115
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Jun 17 14:23:10.621: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
    From: "" <sip:[email protected]>;tag=40FCB218-23D7
    To: <sip:[email protected]>
    Date: Tue, 17 Jun 2014 09:23:10 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE: 1800
    Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1402996990
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A
    All possible debugging has been turned off
    DC#231",qop=auth,algorithm=md5,nc=00000001
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 309
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
    s=SIP Call
    c=IN IP4 92.63.108.115
    t=0 0
    m=audio 18252 RTP/AVP 0 18 8 101
    c=IN IP4 92.63.108.115
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Debug voice ccapi inout:
     Destination Pattern=9375........., Called Number=375298911396, Digit Strip=FALSE
    Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccCallSetupRequest:
       Calling Number=141756(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
       Called Number=375298911396(TON=Unknown, NPI=Unknown),
       Redirect Number=, Display Info=Vankuver
       Account Number=, Final Destination Flag=FALSE,
       Guid=13366763-F540-11E3-AF35-FAC82C2E981E, Outgoing Dial-peer=4
    Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/cc_api_display_ie_subfields:
       ccCallSetupRequest:
       cisco-username=
       ----- ccCallInfo IE subfields -----
       cisco-ani=141756
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=0
       dest=375298911396
       cisco-desttype=0
       cisco-destplan=0
       cisco-rdie=FFFFFFFF
       cisco-rdn=
       cisco-rdntype=0
       cisco-rdnplan=0
       cisco-rdnpi=0
       cisco-rdnsi=0
       cisco-redirectreason=0   fwd_final_type =0
       final_redirectNumber =
       hunt_group_timeout =0
    Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
       Interface=0x6968AA04, Interface Type=3, Destination=, Mode=0x0,
       Call Params(Calling Number=141756,(Calling Name=Vankuver)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
       Called Number=375298911396(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
       Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=4, Call Count On=FALSE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
    Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    Jun 17 15:22:13.073: :cc_get_feature_vsa malloc success
    Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    Jun 17 15:22:13.077:  cc_get_feature_vsa count is 2
    Jun 17 15:22:13.077: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    Jun 17 15:22:13.077: :FEATURE_VSA attributes are: feature_name:0,feature_time:1819298856,feature_id:3371
    Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
       SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
    Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccCallSetContext:
       Context=0x6C726BF4
    Jun 17 15:22:13.077: //14425/13366763AF35/CCAPI/ccSaveDialpeerTag:
       Outgoing Dial-peer=4
    Jun 17 15:22:13.085: //14427/13366763AF35/CCAPI/cc_api_call_proceeding:
    Please help me... I don't know what to do!

    You need to contact service provider for this , after authentication challenge your sip provider is not sending any response.
    Contact them and ask whether they had received INVITE with proxy authentication details or not.

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    Hi Guys,
    In CCA i get the error:java.lang.NullPointerException and it is only when selecting "Auto Attendant".
    No Configuration has been done via CLI
    Cisco IOS Software, UC500 Software (UC500-ADVIPSERVICESK9-M), Version 15.1(4)M5, RELEASE SOFTWARE (fc1)
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    Attaching CCA logs
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    http://wwws.sun.com/software/sundev/jde/index.html

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    Hi, Does anyone know if there is a way to get call data for forwarded calls. Specifically calls forwarded to an extension from the Cisco Attendant Console.

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