SIP & H323

Dos anyone have a configuration example for running SIP and H323 on the same gateway?
I am trying to use Broadvoice and route calls to another router with FXS ports on it via H323.
Thanks in advance.
Steve

[partial config]
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711alaw
codec preference 4 g711ulaw
voice class codec 2
codec preference 1 g729r8
dial-peer voice 1001 voip
description description -SIP- VOXBONE
application prepaid
voice-class codec 2
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 2000 voip
destination-pattern 001.T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 100
session target ipv4:1.1.1.10
dtmf-relay h245-signal h245-alphanumeric
dial-peer voice 2001 voip
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 100
session target ipv4:1.1.1.1
dtmf-relay h245-signal h245-alphanumeric
dial-peer voice 2002 voip
destination-pattern 0057.T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 100
session target ipv4:1.1.1.2
dtmf-relay h245-signal h245-alphanumeric
dial-peer voice 2003 voip
destination-pattern 0056.T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 102
session target ipv4:1.1.1.3
dtmf-relay h245-signal h245-alphanumeric
sip-ua
credentials username 5512041842 password 107758300B4F405A05572722090A671A2D rea
lm SIP_Auth_Realm
registrar ipv4:1.1.1.1 expires 60
If you need specific info let me know

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    number 600
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    number 700
    name test
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    voice register pool  1
    id mac B8BE.BF23.5242
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    template 1
    username test password test
    camera
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    type 9971
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    camera
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    voice register pool  3
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      30820241 308201AA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
      31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
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      53A6E339 4978D451 21F051BE B21F8AD5 86B952DC 1ECCE371 3E094B54 26A41E14
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      947EA591 81CA5489 1B868239 E835EC7C 0AA7651A 22D6E47F 545EBEF3 A172C9A3
      5A0D0203 010001A3 69306730 0F060355 1D130101 FF040530 030101FF 30140603
      551D1104 0D300B82 09564F49 502D3338 3435301F 0603551D 23041830 1680146C
      934AD072 99DDC600 ECD6F389 8F71E0C2 18EC2E30 1D060355 1D0E0416 04146C93
      4AD07299 DDC600EC D6F3898F 71E0C218 EC2E300D 06092A86 4886F70D 01010405
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    tftp-server flash:gui/Delete.gif
    tftp-server flash:gui/dom.js
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    tftp-server flash:ringtones/Analog2.raw
    tftp-server flash:ringtones/AreYouThere.raw
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    tftp-server flash:ringtones/Bass.raw
    tftp-server flash:ringtones/CallBack.raw
    tftp-server flash:ringtones/Chime.raw
    tftp-server flash:ringtones/Classic1.raw
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    tftp-server flash:ringtones/ClockShop.raw
    tftp-server flash:ringtones/DistinctiveRingList.xml
    tftp-server flash:ringtones/Drums1.raw
    tftp-server flash:ringtones/Drums2.raw
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    tftp-server flash:ringtones/Pop.raw
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    tftp-server flash:ringtones/Sax1.raw
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    tftp-server flash:APPS-1.2.1.SBN
    tftp-server flash:SYS-1.2.1.SBN
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    tftp-server flash:CP7921G-1.2.1.LOADS
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    tftp-server flash:SYSH-1.3.1.SBN
    tftp-server flash:TNUXH-1.3.1.SBN
    tftp-server flash:WLANH-1.3.1.SBN
    tftp-server flash:SCCP11.9-2-1S.loads
    tftp-server flash:Desktops/320x212x12/CampusNight.png
    tftp-server flash:Desktops/320x212x12/CiscoFountain.png
    tftp-server flash:Desktops/320x212x12/MorroRock.png
    tftp-server flash:skern9971.022809R2-9-2-1.sebn
    tftp-server flash:sip9971.9-2-1.loads
    tftp-server flash:sboot9971.031610R1-9-2-1.sebn
    tftp-server flash:rootfs9971.9-2-1.sebn
    tftp-server flash:dkern9971.100609R2-9-2-1.sebn
    tftp-server flash:kern9971.9-2-1.sebn
    tftp-server flash:United_States/g4-tones.xml
    tftp-server flash:English_United_States/gd-sip.jar
    tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
    tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
    tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
    tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
    tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
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    destination-pattern *3709
    session protocol sipv2
    session target ipv4:192.168.13.130
    session transport tcp
    dtmf-relay h245-alphanumeric
    codec g711ulaw
    no vad
    dial-peer voice 2 pots
    incoming called-number .
    dial-peer voice 10 voip
    gatekeeper
    shutdown
    telephony-service
    em logout 0:0 0:0 0:0
    max-ephones 262
    max-dn 400
    ip source-address 192.168.2.1 port 2000
    load 7911 SCCP11.9-2-1S
    max-conferences 12 gain -6
    web admin system name admin secret 5 $1$IKnn$tyKyuBcGqXFl6nhxCSu.z0
    dn-webedit
    time-webedit
    transfer-system full-consult
    transfer-pattern .T
    create cnf-files version-stamp 7960 Oct 29 2011 12:39:25
    ephone-template  1
    softkeys connected  Confrn Endcall Trnsfer Hold
    keep-conference endcall
    ephone-dn  1  dual-line
    number 200
    label test
    name test
    ephone-dn  2  dual-line
    number 300
    label Sepahbod
    name Sepahbod
    ephone-dn  4  dual-line
    number 666
    ephone-dn  5  dual-line
    number 660
    ephone-dn  6  dual-line
    number 670
    ephone-dn  7  dual-line
    number 770
    ephone-dn  8  dual-line
    number 770
    ephone-dn  9  dual-line
    number 999
    ephone  1
    device-security-mode none
    mac-address 18EF.639F.BCB0
    keep-conference endcall
    button  1:1
    ephone  2
    device-security-mode none
    mac-address 0025.8418.B017
    ephone-template 1
    keep-conference endcall
    button  1:2
    ephone  3
    device-security-mode none
    mac-address F04D.A243.3154
    keep-conference endcall
    button  1:4
    ephone  4
    device-security-mode none
    mac-address 6CF0.496A.69E9
    button  1:4
    ephone  5
    device-security-mode none
    mac-address 0015.E987.345F
    keep-conference endcall
    button  1:5
    ephone  6
    device-security-mode none
    mac-address 0024.1DEA.614A
    keep-conference endcall
    button  1:6
    ephone  9
    device-security-mode none
    mac-address 001D.7D4D.4DCB
    button  1:9
    line con 0
    line aux 0
    line vty 0 4
    login local
    transport input telnet
    scheduler allocate 20000 1000
    end
    and Voice Gateway connected two PBX system configuration
    Current configuration : 3486 bytes
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Voice-GW
    boot-start-marker
    boot-end-marker
    card type e1 0 2
    no aaa new-model
    network-clock-participate wic 2
    dot11 syslog
    ip source-route
    ip cef
    no ipv6 cef
    multilink bundle-name authenticated
    isdn switch-type primary-net5
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    h323
    voice-card 0
    crypto pki token default removal timeout 0
    license udi pid CISCO2811 sn FHK1352F0E9
    username admin privilege 15 secret 5 $1$O6AN$1kvvqiLdIl3/ZTHoyYRy0/
    redundancy
    controller E1 0/2/0
    framing NO-CRC4
    pri-group timeslots 1-31
    controller E1 0/2/1
    interface Tunnel14
    ip address 192.168.13.130 255.255.255.252
    tunnel source FastEthernet0/1
    tunnel destination 10.9.160.25
    interface Tunnel17
    ip address 192.168.13.134 255.255.255.252
    tunnel source FastEthernet0/1
    tunnel destination 10.9.160.25
    interface FastEthernet0/0
    ip address 192.168.14.252 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    ip address 10.2.68.25 255.255.255.0
    duplex auto
    speed auto
    interface Serial0/2/0:15
    no ip address
    encapsulation hdlc
    isdn switch-type primary-net5
    isdn overlap-receiving
    isdn incoming-voice voice
    no cdp enable
    router eigrp 201
    network 172.25.10.0 0.0.0.255
    network 192.168.14.0
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 10.9.160.0 255.255.255.0 10.2.68.1
    ip route 10.128.0.69 255.255.255.255 Tunnel14
    ip route 192.168.2.1 255.255.255.255 192.168.13.129
    ip route 192.168.17.0 255.255.255.0 Tunnel14
    tftp-server flash:SCCP11.9-2-1S.loads
    tftp-server flash:jar11sccp.9-2-1TH1-13.sbn
    tftp-server flash:dsp11.9-2-1TH1-13.sbn
    tftp-server flash:cvm11sccp.9-2-1TH1-13.sbn
    tftp-server flash:cnu11.9-2-1TH1-13.sbn
    tftp-server flash:apps11.9-2-1TH1-13.sbn
    control-plane
    voice-port 0/0/0
    caller-id enable
    voice-port 0/0/1
    voice-port 0/0/2
    supervisory disconnect dualtone mid-call
    dial-type pulse
    disc_pi_off
    output attenuation 1
    echo-cancel coverage 32
    timeouts call-disconnect 5
    timeouts wait-release 1
    timing hookflash-out 50
    timing sup-disconnect 50
    connection plar 600
    caller-id enable
    voice-port 0/0/3
    caller-id enable
    voice-port 0/2/0:15
    mgcp profile default
    dial-peer voice 1 pots
    description connection-to-PBX
    destination-pattern 0....
    direct-inward-dial
    port 0/2/0:15
    forward-digits 4
    dial-peer voice 10 voip
    destination-pattern ...
    session target ipv4:192.168.13.129
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 20 pots
    description FXO-K
    destination-pattern 9T
    progress_ind alert enable 8
    progress_ind progress enable 8
    progress_ind connect enable 8
    direct-inward-dial
    port 0/0/2
    prefix 9
    dial-peer voice 30 pots
    description FXO-K2
    destination-pattern 9T
    direct-inward-dial
    port 0/0/1
    prefix 9
    telephony-service
    max-ephones 20
    max-dn 100
    ip source-address 192.168.14.252 port 2000
    cnf-file location flash:
    load 7911 term11.default.loads
    max-conferences 8 gain -6
    transfer-system full-consult
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1
    number 770
    line con 0
    line aux 0
    line 1/0 1/15
    line vty 0 4
    login local
    transport input telnet
    scheduler allocate 20000 1000
    end

    Having looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
    I think you may be able to work around the problem by adding
    " supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
    reference
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
    worth a try
    Adam

  • Sip 500 Internal Server Error Reason: Q.850;cause=16

    Please help in understanding what is wrong in the config .Incoming calls don't work.
    show run:
    voice service voip
    ip address trusted list
      ipv4 87.226.136.164 255.255.255.255
      ipv4 172.16.24.0 255.255.255.0
      ipv4 188.254.68.66 255.255.255.255
      ipv4 188.254.68.67 255.255.255.255
      ipv4 188.254.69.66 255.255.255.255
      ipv4 188.254.69.67 255.255.255.255
      ipv4 46.38.52.68 255.255.255.255
    address-hiding
    allow-connections h323 to h323
    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
    no supplementary-service sip refer
    redirect ip2ip
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
    sip
    voice class codec 1
    codec preference 1 g729br8
    codec preference 2 g729r8
    codec preference 3 g711alaw
    codec preference 4 g711ulaw
    voice class codec 2
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    codec preference 3 g729r8
    codec preference 4 g729br8
    voice translation-rule 1
    rule 1 /XXX5397962/ /1999/
    voice translation-rule 2
    rule 1 /XXX55317577/ /1999/
    voice translation-rule 3
    rule 1 /5555317884/ /1999/
    voice translation-profile ROS
    translate called 1
    voice translation-profile ROS2
    translate called 2
    voice translation-profile ROS3
    translate called 3
    interface FastEthernet0/0
    ip address 178.208.X.X 255.255.255.248
    ip access-group INBOUND in
    no ip unreachables
    ip verify unicast reverse-path
    ip nat outside
    ip inspect IPFW in
    ip inspect IPFW out
    ip virtual-reassembly in
    duplex auto
    speed auto
    no cdp enable
    interface FastEthernet0/1
    no ip address
    ip nat inside
    ip virtual-reassembly in
    duplex auto
    speed auto
    interface FastEthernet0/1.1
    encapsulation dot1Q 1 native
    ip address 10.110.0.200 255.255.255.0
    ip nat inside
    ip virtual-reassembly in
    interface FastEthernet0/1.2
    encapsulation dot1Q 2
    ip address 172.16.24.254 255.255.255.0
    ip nat inside
    ip virtual-reassembly in
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 172.16.24.254
    ip dns server
    ip nat inside source list NAT interface FastEthernet0/0 overload
    ip route 0.0.0.0 0.0.0.0 178.208.X.X
    ip route 192.168.0.0 255.255.0.0 Null0 254
    sccp local FastEthernet0/1.2
    sccp ccm 172.16.24.101 identifier 1 version 7.0
    sccp
    sccp ccm group 1
    associate ccm 1 priority 1
    associate profile 1 register XCODE123456
    keepalive retries 1
    keepalive timeout 10
    switchover method immediate
    switchback method immediate
    dspfarm profile 1 transcode
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 6
    associate application SCCP
    dial-peer voice 10000 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS
    destination-pattern 74955397962
    session protocol sipv2
    session target ipv4:87.226.136.164
    session transport udp
    incoming called-number XXXX5397962
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 10010 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS2
    destination-pattern XXX55317577
    session protocol sipv2
    session target ipv4:87.226.136.164
    session transport udp
    incoming called-number 75555317577
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 10020 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS3
    preference 1
    destination-pattern 5555317884
    session protocol sipv2
    session target ipv4:188.254.68.66
    session transport udp
    incoming called-number 5555317884
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 10021 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS
    preference 2
    destination-pattern 5555317884
    session protocol sipv2
    session target ipv4:188.254.69.66
    session transport udp
    incoming called-number 5555317884
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 2 voip
    tone ringback alert-no-PI
    description to CUCM_PUB
    destination-pattern 1...
    session target ipv4:172.16.24.101
    voice-class codec 2
    dtmf-relay rtp-nte
    debug ccsip all:
    c2801#
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a(STATE_IDLE, SUBSTATE_NONE)
    06:19:26: //-1/xxxxxxxxxxxx/SIP/T3c000c-1
    Call-ID: isbc6994325518770806443-1385214296-16204
    Fransport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 5060
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone MSK to SIP default timezone = GMT
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 929rom:
    <sip:[email protected];user=phone>;tag=sbc09026994325from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 5060
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone MSK to SIP default timezone = GMT
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 518770806443
    ddress_to_bind: return addr 178.208.X.Xone>
    06:19:26: //-1/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 5060
    0
    CSeq: 1 INVITE
    Min-SE: 90
    Session-Expires: 3600;refresher=u6:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Convac
    Contact: <sip:[email protected]:9290;user=phone>
    A //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: rellow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK
    Supported:turn addr 178.208.X.X
    06:19:26: //-1/EE5EC9DD8170/SIP/St timer,100rel
    Diversion: <sip:[email protected]>;privacyate/sipSPIChangeState: 0x6A874E70 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1,
    Sen=off;screen=no;reason=unknown,<sip:[email protected]>;priv6:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponsacy=off;screen=no;reason=unknown
    Max-Forwards: 70
    User-AgenteInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Tra: VCS 5.8.2.56-03
    Content-Length: 393
    Content-Type: applicatnsport 1, SentBy Port 9290ion/sdp
    v=0
    o=- 12060 26053 IN IP4 188.254.68.67
    s=SBC call
    c=IN IP4 188.254.68.67
    t=0 0
    m=audio 24402 RTP/AVP 8 0 18 98 96 97 101
    a=rtpmap:98 G.729a/8000
    a=rtpmap:96 G.729ab/8000
    a=rtpmap:97 G.729b/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=fmtp:18 annexb=no
    a=ptime:10
    a=X-vrzcap:vbd Ver=1 Mode=FaxPr ModemRtpRed=0
    a=X-vrzcap:identification bin=DSR2866 Prot=mgcp App=MG
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x6A874E70) with key=[52] to table
    06:19:26: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 9290
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 178.208.X.X
    06:19:26: //-1/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_NONE, SUBSTATE_NONE)  to
    c2801#L
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x6
    c2801#a
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIMatchSrcIp
    c2801#mat: VIA URL:sip:188.254.68.66:9290, Host:188.254.68.66
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : 9067259847
    06:19:26:ched for incoming call
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported h
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetFromCalledPartyId: P-Called-Party-ID header not found
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetPeerByCalledPartyId: P-Called-Party-ID not found or parse error
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Peer tag 10020 matched for incoming call
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 178.208.X.X
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F
    06:19:26: //-1/EE5EC9DD8170/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: xcoder high-density disabled
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Flow Mode set to FLOW_THROUGH
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Media forking disabled
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: Calling name , number 9067259847, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number
    5555317884, oct3 0x00
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: Requires reliable-provisional support
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIValidateRequestUri: Not Enabled
    06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_IsSDPPassthruEnabled:  - 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:6A01759C
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLExtractSessionExpiresHdr:
    Session-Expires value: 3600 refresher: uac
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLExtractMinSEHdr: Min-SE Duration: 90
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLGetInternalSREvent: E_STSL_INITIAL_SR_REQ
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLInitialSRReqPeerEventGen: sending received session expires to the peer leg
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:3600, SE Refresher:uac, Min-SE Value:1800,
    flags:2001
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLMain:
            SE: 3600;refresher:uac peer refresher:none, flags:2001, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
            Configured SE:1800, Configured Min-SE:1800
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessReplacesHeader: No replaces hdr found
    SIP: Warning: Unrecognized attribute (X-vrzcap)
    SIP: Warning: Unrecognized attribute (X-vrzcap)
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIValidateConnectionAddress: Dest port = 24402
    SIP: (129) Attribute mid, level 1 instance 1 not found.
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling reg_invoke_ip_first_hop()
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling ip_best_local_address()
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: return addr 178.208.X.X
    06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 178.208.X.X
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(98) reserved for codec g729r8
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(98) reserved for codec g729r8
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(96) reserved for codec g729abr8
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(96) could not be reserved
                              as its in use by other codec g729abr8
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (96) is  reserved by another application
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 96
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 99 for Codec:
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(97) reserved for codec g729br8
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(97) could not be reserved
                              as its in use by other codec g729br8
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (97) is  reserved by another application
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 97
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 102 for Codec:
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec No Codec 
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 99
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved
                              as its in use by other codec No Codec 
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (101) is  reserved by another application
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 103
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 101
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 101 for Codec:
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:10
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :10, codecbytes: 80
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :80, ptime: 10
    06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:10, Negotiated ptime:10 Negotiated codec bytes: 80 for codec g711ulaw
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPISetFaxFlags: FAX_PASSTHROUGH = 0, END_FAX_PASSTHROUGH = 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIReserveRtpNtePayload: Reserved the payload type 101 for RTP-NTE
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of events.
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: V150 NSE payload = 0, SSE payload = 0, SPRT payload=0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-
    line:1 and num-a-lines:0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
            payload_type=0, codec_bytes=80, codec=g711ulaw, dtmf_relay=rtp-nte
            stream_type=voice+dtmf (1), dest_ip_address=188.254.68.67, dest_port=24402
    06:19:26: //129/EE5EC9DD8170/SIP/State/sipSPIChangeStreamState: Stream (callid =  -1)  State changed from (STREAM_DEAD) to (STREAM_ADDING)
    06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPIUpdCallWithSdpInfo:
            Preferred Codec        : g711ulaw, bytes :160
            Preferred  DTMF relay  : rtp-nte
            Preferred NTE payload  : 101
            Early Media            : No
            Delayed Media          : No
            Bridge Done            : No
            New Media              : No
            DSP DNLD Reqd          : No
    06:19:26: //129/EE5EC9DD8170/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
    06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 178.208.X.X
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
    callId 129 peer 0 flags 0x201 state STATE_IDLE
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_vcc_ProcessXcoderNeeded: xcoder_attached not yet initialised for this call.
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer: Xcoder not yet used for the call
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    CallID 129, Peer CallID 0, sdp 0x69EC3234 channels 0x6A8763C4
    06:19:26: //129/EE5EC9DD8170/SIP/Info/copy_channels:
    callId 129 size 0 ptr 0x6899F6D4)
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    CCB t38 version 0 ipip_caps version 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    CCB fax rate 2 ipip_caps rate 14400
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: reset the  switch..
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 8 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw
    06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:
    Codec to be matched: 6
    06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found: No match for the codecs found..
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 0 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
    06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:
    Codec to be matched: 5
    06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:  codecs[i] = 5 & codec = 5 are same..
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :80, ptime: 10
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=10,stream->negotiated_codec_bytes=80,
    coverted ptime=10 stream->mline_index=1, media_ndx=1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    failed to update call entry
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Adding codec 5 ptype 0 time 10, bytes 80  as channel 0 mline 1 ss 1 188.254.68.67:24402
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 18 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPISelectCodecVersion: Codec (g729r8) is not in preferred list
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using interoperable codec g729r8
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8
    06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:
    Codec to be matched: 16
    06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found: No match for the codecs found..
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 98 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 96 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 97 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 101 mline 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: setting ipip_caps DTMF to RFC2833: callid = 129, dtmf = 6
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec
    = 5
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec
    = -1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
    callId 129 flags 0x100 state STATE_IDLE
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
    Report initial call media
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0x804000C, ccb->pld.flags_ipip 0x201
    06:19:26: //129/EE5EC9DD8170/SIP/Info/copy_channels:
    callId 129 size 240 ptr 0x69E20A34)
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
    CCSIP: Unable to report channel ind
    06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_update_srtp_caps:  5798: Posting Remote SRTP caps to other callleg.
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer: do cc_api_caps_ind()
    06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPIUpdCallWithSdpInfo:
              Stream type            : voice+dtmf
              Media line             : 1
              State                  : STREAM_ADDING (2)
              Stream address type    : 1
              Callid                 : -1
              Negotiated Codec       : g711ulaw, bytes :80
              Nego. Codec payload    : 0 (tx), 0 (rx)
              Negotiated DTMF relay  : rtp-nte
              Negotiated NTE payload : 101 (tx), 101 (rx)
              Negotiated CN payload  : 0
              Media Srce Addr/Port   : [178.208.X.X]:0
              Media Dest Addr/Port   : [188.254.68.67]:24402
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIHandleInviteMedia:
    Negotiated Codec       : g711ulaw, bytes :80
    Preferred Codec        : g711ulaw, bytes :160
    Preferred  DTMF relay 1 : 6
    Preferred  DTMF relay 2 : 0
    Negotiated DTMF relay   : 6
    Preferred and Negotiated NTE payloads: 101 101
    Preferred and Negotiated NSE payloads: 100 0
    Preferred and Negotiated Modem Relay: 0 0
    Preferred and Negotiated V150.1 Modem Passthrough: 0 0
    Preferred and Negotiated V150.1 Modem Relay: 0 0
    Preferred and Negotiated Modem Relay GwXid: 1 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: QOS negotiation for mline_index 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoStreamQoSNegotiation: Best effort
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17550 for stream 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17550
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17550
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
    SIP update src sdp, negoitated codec 5, payload type 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = isbc6994325518770806443-1385214296-16204
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCPA: No CPA found in inbound container
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessCPA: No x-cisco-cpa content found
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_IsContentPassthruEnabled:  - 0
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_ExtractPassthruContentFromSipContainer: Passthru Content Not Enabled
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 6,
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIShrlCall: Check peer: 10020 for Shared-Line call, callid: 129
    06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_set_bearer_capability:
       Bearer Capability: Speech (0x00)
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 81 to table
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:6A0173E4
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLValidateSessRefreshMsg: Ignoring 1xx response for session timer processing
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPISendInviteResponse: Associated container=0x6A0173E4 to Invite Response 100
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: msg=0x6A5A1A34, addr=188.254.68.66, port=9290, sentBy_port=9290, local_addr=, is_req=0,
    transport=1, switch=0, callBack=0x0
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x6A5A1A34 to default port=9290
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostRequestConnection: Posting UDP conn create request for addr=188.254.68.66, port=9290, context=0x68ABB118
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetConnWaitTimer: Wait timer set for connection=0x68ABCB0C,addr=188.254.68.66, port=9290
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetConnWaitTimer:
    Wait Conn Timer started for 5000 msec
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipCreateConnInstance: Created new initiated conn=0x68ABCB0C, connid=-1, addr=188.254.68.66, port=9290, local_addr=,
    transport=UDP
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:188.254.68.66, rport:9290 with laddr:
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstanceGetConnectionId: Registering gcb=0x6A874E70 with connection=0x68ABCB0C
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Waiting for Connection for sending msg=0x6A5A1A34
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: Deferred sending msg=0x6A5A1A34
    06:19:26: //129/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_RECD_INVITE, SUBSTATE_NONE)
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessContactInfo: Previous Hop 188.254.68.66:9290
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 165)
    06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_event_handler:
    ccsip_event_handler: peer ID 130 chans 0x6780D478 event 165 flags 0x844001C 0x100 0x601 data 0x6780D478
    06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_event_handler:
    ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 130 chans 0x6780D478 event 165 flags 0x844001C 0x100 0x601 data 0x6780D478, type = 1
    06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-H323
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SET_MODE
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLMain:
            SE: 3600;refresher:uac peer refresher:none, flags:2001, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
            Configured SE:1800, Configured Min-SE:1800
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 3
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 58
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWConnectionCreated: context=0x68ABB118
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessConnCreated: gConnTab=0x68ABB118, addr=188.254.68.66, port=9290, local_addr=, connid=3,
    transport=UDP
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessConnCreated: connection instance created for addr:188.254.68.66, port:9290 local_addr=
    local_port=57282
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportStopConnWaitTimer: Wait timer stopped for connection=0x68ABCB0C,addr=188.254.68.66, port=9290
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstanceHandleConnectionCreated: Moving connection=0x68ABCB0C, connid=3 state to established. local_addr=,
    local_port=57282
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportPostInternalMsg: Posting Internal Msg type=0
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 63
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x6A5A1A34, addr=188.254.68.66, port=9290, local_addr=, connId=3 for UDP
    06:19:26: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-1
    From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
    To: <sip:[email protected];user=phone>
    Date: Sat, 23 Nov 2013 13:42:29 GMT
    Call-ID: isbc6994325518770806443-1385214296-16204
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_ALERTING
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_call_service_msg: ccb NULL, unable to update the callinfo ui parameters
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_MEDIA_EVENT
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 5
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIAddCiscoGcid: Fatal Error in parsing CCB/Msg
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIStoreTunnelData: Container /RawMessage Absent
    06:19:26: //129/EE5EC9DD8170/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x6A874E70 key=isbc6994325518770806443-1385214296-1620415B6280-0
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:180, container:6A017B1C
    06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLValidateSessRefreshMsg: Ignoring 1xx response for session timer processing
    06:19:26: //129/EE5EC9DD8170/SIP/Event/sipSPICreateRpid: Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPISendInviteResponse: Associated container=0x6A017B1C to Invite Response 180
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPISendInviteResponse: Sending 180 Response to the Transport Layer
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: msg=0x6A5A1A34, addr=188.254.68.66, port=9290, sentBy_port=9290, local_addr=, is_req=0,
    transport=1, switch=0, callBack=0x618A57B8
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x6A5A1A34 to default port=9290
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:188.254.68.66, rport:9290 with laddr:
    06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x6A5A1A34
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x6A5A1A34, addr=188.254.68.66, port=9290, local_addr=, connId=3 for UDP
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sentInviteResponse18x: Sent a 18x Response
    06:19:26: //129/EE5EC9DD8170/SIP/Info/sentInviteResponse18x: Transaction active. Facilities will be queued.
    06:19:26: //129/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE)  to (STATE_SENT_ALERTING, SUBSTATE_NONE)
    06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 30ty-ID:
    <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    06:19:27: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
    06:19:27: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
    06:19:27: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-2
    From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
    To: <sip:[email protected];user=phone>;tag=15B6280-0
    Date: Sat, 23 Nov 2013 13:42:30 GMT
    Call-ID: isbc6994325518770806443-1385214296-16204
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 2 PRACK
    Content-Length: 0
    06:19:27: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
    Sent:
    UPDATE sip:[email protected]:9290;user=phone SIP/2.0
    Via: SIP/2.0/UDP 178.208.X.X:5060;branch=z9hG4bK120
    From: <sip:[email protected];user=phone>;tag=15B6280-0
    To: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
    Date: Sat, 23 Nov 2013 13:42:30 GMT
    Call-ID: isbc6994325518770806443-1385214296-16204
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Timestamp: 1385214150
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 UPDATE
    Contact: <sip:[email protected]:5060>
    Min-SE:  1800
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
    Content-Length: 0
    06:19:27: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-2
    From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
    To: <sip:[email protected];user=phone>;tag=15B6280-0
    Date: Sat, 23 Nov 2013 13:42:30 GMT
    Call-ID: isbc6994325518770806443-1385214296-16204
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 2 PRACK
    Content-Length: 0
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
    06:19:31: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 500 Internal Server Error
    Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-1
    From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
    To: <sip:[email protected];user=phone>;tag=15B6280-0
    Date: Sat, 23 Nov 2013 13:42:30 GMT
    Call-ID: isbc6994325518770806443-1385214296-16204
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=16
    Content-Length: 0
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-1
    Call-ID: isbc6994325518770806443-1385214296-16204
    From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
    To: <sip:[email protected];user=phone>;tag=15B6280-0
    CSeq: 1 ACK
    Max-Forwards: 70
    Content-Length: 0
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
    06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIFindCcbUASRespTable: *****CCB found in UAS Response table. ccb=0x6A874E70
    06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x6A874E70 key=isbc6994325518770806443-1385214296-1620415B6280-0
    06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
    06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIStopRequestPendingTimer: Stopping Request Pending Timer
    06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
    06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 6A874E70
    06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[52]
    c2801#
    c2801#

    Hi Ahmed,
    Looking at the logs, it seems some important messages are missing..
    can you please again collect the following debugs ?
    - debug voip ccapi inout
    - debug ccsip all
    - debug voice translation
    Thanks,
    Piyush

  • SiP Phone wont dial inbound or outbound

    I have 9971 phone and was dialing sip to sip and sip h323 on the network, but now I get Apr 20 14:17:58.911: %VOICE_IEC-3-GW: Application Framework Core: Internal error
    (Toll fraud call rejected): IEC=1.1.228.3.31.0 on callID 12 GUID=BCBC7FI

    Hi Wharrison,
    can you please provide the call flow and where do you see this error.
    I am guessing the call is from an IP phone regsitered to CUCM --> SIP truk --> CUBE --> provider.. Is this right?
    Please let me know where do you see the error.
    Thanks,
    Manoj

  • CallManager 4.x to SIP ITSP

    Hello,
    I am trying to find out if a CallManager 4.x system can communicate to an IP Telephony Service Provider over SIP to their Sessions Border Controller. From what it looks like, I will need an IP2IP gateway to talk H323 to the CCM and SIP to the ITSP. Has anyone successfully done this before?
    Any help or experiences would be greatly appreciated.

    We have tested this in our lab, and this was working well. An Cisco 2811 with ver. 12.4 IP2IP was used for this test and H323 to SIP, H323 to H323 and SIP to H323 was working well.
    Config Cisco router :
    Building configuration...
    Current configuration : 2983 bytes
    version 12.4
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Router
    boot-start-marker
    boot-end-marker
    enable password cisco
    no aaa new-model
    ip cef
    no ip dhcp use vrf connected
    no ip dhcp conflict logging
    ip dhcp excluded-address 10.193.25.1 10.193.25.65
    ip dhcp excluded-address 172.16.1.1 172.16.1.9
    ip dhcp excluded-address 10.193.25.70 10.193.25.80
    ip dhcp pool 10.193.25.0
    network 10.193.25.0 255.255.255.0
    option 150 ip 10.193.25.113
    default-router 10.193.25.111
    lease infinite
    multilink bundle-name authenticated
    voice-card 0
    no dspfarm
    dsp services dspfarm
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    h323
    call start interwork
    sip
    interface Loopback0
    ip address 10.0.0.1 255.255.255.255
    interface FastEthernet0/0
    ip address 10.193.25.111 255.255.255.0
    duplex auto
    speed auto
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 10.193.25.111
    interface FastEthernet0/1
    no ip address
    shutdown
    duplex auto
    speed auto
    interface Content-Engine1/0
    no ip address
    shutdown
    ip route 0.0.0.0 0.0.0.0 10.193.25.1
    no ip http server
    control-plane
    dial-peer voice 10 voip
    description VoIP to live callmanager
    destination-pattern 3...
    progress_ind connect enable 8
    session target ipv4:10.193.1.5
    dtmf-relay h245-alphanumeric
    codec g711alaw
    dial-peer voice 20 voip
    description VoIP to Test Callmanager
    tone ringback alert-no-PI
    destination-pattern 2...
    progress_ind setup enable 3
    progress_ind progress enable 8
    progress_ind connect enable 8
    session target ipv4:10.193.25.113
    dtmf-relay h245-alphanumeric
    codec g711alaw
    dial-peer voice 30 voip
    description to VoIP/AA at Test Callmanager
    destination-pattern 500.
    session target ipv4:10.193.25.113
    dtmf-relay h245-alphanumeric
    codec g711alaw
    dial-peer voice 1 voip
    description to H323 External GW
    destination-pattern 0T
    session target ipv4:10.193.1.4
    dtmf-relay h245-alphanumeric
    codec g711alaw
    dial-peer voice 200 voip
    description to SIP Soft IP-Phone
    destination-pattern 1999
    session protocol sipv2
    session target ipv4:10.193.10.9
    dtmf-relay rtp-nte
    codec g711alaw
    dial-peer voice 100 voip
    tone ringback alert-no-PI
    description 3th party hardware SIP IPPhone
    destination-pattern 1...
    session protocol sipv2
    session target ipv4:10.193.25.200:5060
    dtmf-relay rtp-nte h245-alphanumeric
    codec g711alaw
    no vad
    sip-ua
    retry options 0
    gatekeeper
    shutdown
    line con 0
    line aux 0
    line 66
    no activation-character
    no exec
    transport preferred none
    transport input all
    transport output pad telnet rlogin lapb-ta mop udptn v120
    line vty 0 4
    password cisco
    login
    scheduler allocate 20000 1000
    end
    Router#

  • Call/video not working between Cisco jabber for Windows and VCS control C40s

    Hello,
    I've been struggling with no luck how to make a call using Cisco Jabber for Windows 9.6.0 registered to CM 8.6.2 with intercluster ICT to another CM 8.6.2 where we have a VCS Control 7.0.2 via GK H225, and all C40s are registered as H.323.
    The VCS has interworking between H323 and SIP, however not sure if there is any problem with that. Assuming it is ok, not sure either if I'm facing any interoperability issue because in my remote site I have C40 (H323 registered at VCS and SIP listening mode) and cisco jabber for windows which is SIP based.
    If is not possible, would I be able to change my C40 from H323 to SIP at VCS, or have both H323/SIP registered at VCS? If so, will I need to change as well instead of GK I'll have to establish a SIP Trunk between the CM and VCS?
    Another thing I do not believe either I would be able to have one VCS connected with two clusters, right?
    I'm just trying to find a solution in case my current topology is not compatible, but feel free if you have any better idea to make it work.
    Anyway here is what is happening:
    When I make a call from my cisco jabber windows to C40 using alias number. The call is being redirected just fine to the C40 and it rings, however when someoene or the auto answer picks it up, the call dropped right away.
    However, if I enabled the MTP in my CSF device, the call gets longer before dropping. I was even able to see my jabber " start video" turns green, before was grayed out all the time and the call dropped faster. I hear a fast busy tone. 
    I'm able to provide SDI traces, logs, diagnostic sip/h323 calls from VCS in order to know for sure if this is an incompatible issue or something I can workaround.
    Let me know if someone of you are interested in read these logs or could point me on the right direction.
    Thanks!

    Ok,
    I have looked at both logs. I have to mentinon though that you didnt
    provide the log that shows the h323 setup between cucm and the VCS. This
    is  most likely because the call originated from a different cucm than
    the ones you provided the logs from.
    The call would have orginated from the first cucm in the cucm group of
    this trunk: Name=RL_TRUNK_VIDEO
    The cucm ip will be : 10.252.53.10.
    This is the VCS log that confirms where the h323 request originated
    from:
    pr 10 22:50:29 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:29,187"
    Module="network.h323" Level="DEBUG":  Src-ip="10.252.53.10"  Src-
    port="54000"
     Received RAS PDU:
    Having said that here is my analysis of the logs that you sent..
    Jabber sent an INVITE to CUCM and advertised all the codecs (audio and
    video it can support)..
    Observer that Jabber says it doesnt support G729 anexB
    21:55:16.576 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message
    from 10.223.20.73 on port 54677 index 90661 with 2220 bytes:
    [862370,NET]
    INVITE sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/TCP 10.223.20.73:54677;branch=z9hG4bK000029d3
    From: "4122107" <sip:[email protected]>;tag=00059a3c78000011000070b0
    -00000e65
    To: <sip:[email protected]>
    Call-ID: [email protected]
    Max-Forwards: 70
    Date: Fri, 11 Apr 2014 01:55:16 GMT
    CSeq: 101 INVITE
    User-Agent: Cisco-CSF/9.4.1
    m=audio 19252 RTP/AVP 0 8 18 105 104 101
    c=IN IP4 10.223.20.73
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:105 G7221/16000
    a=fmtp:105 bitrate=24000
    a=rtpmap:104 G7221/16000
    a=fmtp:104 bitrate=32000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    m=video 28878 RTP/AVP 97
    c=IN IP4 10.223.20.73
    ++++Now lets observer the capabilites exchange during h245 negotiation
    between cucm and VCS++++
    Here CUCM advertises its caps to VCS (afterreceiving caps from VCS)
    Note that G729A, G729AB, G729 is all advertised..
    Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,017"
    Module="network.h323" Level="DEBUG":  Src-ip="10.252.53.10"  Src-
    port="45660"
     Received H.245 PDU:
     value MultimediaSystemControlMessage
    ::= request : terminalCapabilitySet
     capabilityTableEntryNumber 2,
           capability receiveAudioCapability :
    g729wAnnexB : 6
           capabilityTableEntryNumber 3,
       capability receiveAudioCapability : g729AnnexAwAnnexB : 6
           capabilityTableEntryNumber 4,
           capability
    receiveAudioCapability : g729 : 6
    capabilityTableEntryNumber 5,
           capability receiveAudioCapability :
    g729AnnexA : 6
    ++++++
    After doing MSD (master slave determination, we move to the OLC phas e..
    Here we see that the far end..c40 wants to use G729AB for media++++
    Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,783"
    Module="network.h323" Level="DEBUG":  Src-ip="10.224.114.11"  Src-
    port="11163"
     Received H.245 PDU:
     value MultimediaSystemControlMessage
    ::= request : openLogicalChannel :
       forwardLogicalChannelNumber 1,
    forwardLogicalChannelParameters
         dataType audioData :
    g729AnnexAwAnnexB : 20,
         multiplexParameters
    h2250LogicalChannelParameters :
    +++Next VCS sends G729AB as the codec to use to CUCM+++
    Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,784"
    Module="network.h323" Level="DEBUG":  Dst-ip="10.252.53.10"  Dst-
    port="45660"
     Sending H.245 PDU:
     value MultimediaSystemControlMessage
    ::= request : openLogicalChannel :
       forwardLogicalChannelNumber 1,
    forwardLogicalChannelParameters
         dataType audioData :
    g729AnnexAwAnnexB : 20,
         multiplexParameters
    h2250LogicalChannelParameters :
    ++++The next thing we get is an OLC reject from CUCM and this is where
    th call drops++
    Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
    Module="network.h323" Level="DEBUG":  Src-ip="10.252.53.10"  Src-
    port="45660"
     Received H.245 PDU:
     value MultimediaSystemControlMessage
    ::= response : openLogicalChannelReject :
    forwardLogicalChannelNumber 1,
       cause dataTypeNotSupported : NULL
    Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
    Module="network.h323" Level="INFO":  Dst-ip="10.224.114.11"  Dst-
    port="11163"
      Detail="Sending H.245 OpenLogicalChannelRejResponse
    +++We then receive a call release from cucm with cause code of 47:
    resource unavailable++++
    Apr 10 22:50:32 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:32,365"
    Module="network.h323" Level="DEBUG":  Src-ip="10.252.53.10"  Src-
    port="50913"
     Received H.225 PDU:
     Q931
       Message Type: Release
    Complete
       Call reference flag: Message sent from originating side
    Call reference value: 0x7b
       Info Element : Cause
         Location: Usr
       Cause Value: Resource unavailable
       Info Element : User User
       Length = 22
    Suggestions:
    Change the region setting between the ICT trunk to VCS and Jabber to use
    G711 and test again.

  • CISCO Jabber 8.6.2 and CME 8.6

    Hello,
    I want to use Cisco Jabber 8.6.2 with Call manager Express 8.6
    I configured the IPhone on CME and is working ok on local wireless LAN,
    When I’m using the VPN I can place call's inside the network but I can't use on outside line. I have no sound.
    Also if I place a call and the I end the call, the called phone rings and is not stopping. So I think is a call disconnect problem.
    Bellow is a part of the configuration.
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    h323
    call start slow
    sip
    session transport tcp
    registrar server
    voice class codec 1
    codec preference 1 g729r8
    voice class custom-cptone romania
    dualtone busy
    frequency 450
    cadence 170 170
    dualtone disconnect
    frequency 450
    cadence 170 170
    voice register global
    mode cme
    source-address 10.12.4.252 port 5060
    max-dn 10
    max-pool 10
    authenticate register
    hold-alert
    tftp-path flash:
    create profile sync 0000545624458818
    voice register dn 1
    number 5146
    call-forward b2bua all 5108
    call-forward b2bua busy 5160
    call-forward b2bua noan 5160 timeout 18
    name Ioan Stanciu
    shared-line
    label Ioan
    voice register pool 10
    registration-timer max 720 min 660
    id mac 68A8.6D91.3FE0
    session-transport tcp
    type CiscoMobile-iOS
    number 1 dn 1
    username 5146 password 5146
    voice-port 0/0/0
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone romania
    no battery-reversal
    timeouts call-disconnect 2
    timeouts wait-release 2
    connection plar opx immediate 5150
    caller-id enable
    voice-port 0/0/1
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone romania
    no battery-reversal
    timeouts call-disconnect 2
    timeouts wait-release 2
    connection plar opx immediate 5150
    caller-id enable
    dspfarm profile 1 conference
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    codec g722-64
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    ===============================
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    State, 1, 2003, State - PHGreeting.cde!PlayGreeting
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    Display, 1, 2003, No DTMF received
    Display, 1, 2003, Playing greeting for Subscriber:  hq2
    State, 1, 2003, Event is [RecordMsgEvent]
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    Trying 142.100.64.13, 5000 ... Open
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    Application, 1, 911, AttemptForward
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    Application, 1, 911, PHTransfer
    State, 1, 911, State - PHTransfer.cde!LoadInfo
    State, 1, 911, Event is [TrueEvent]
    Application, 1, 911, PHGreeting
    State, 1, 911, State - PHGreeting.cde!PlayGreeting
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    Display, 1, 911, No DTMF received
    Display, 1, 911, Playing greeting for Subscriber:  hq2
    State, 1, 911, Event is [RecordMsgEvent]
    State, 1, 911, State - PHGreeting.cde!RecordMsg
    State, 1, 911, Event is [NULL]
    State, 1, 911, State - PHGreeting.cde!RunEditMsg
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    State, 1, 911, State - MessageEditing.cde!PlayEditMenu
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    State, 1, 911, State - MessageEditing.cde!CheckMsgLength
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    Sounds like one way audio from PSTN to your Unity Connection, couple of things to check:
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    2. Make sure IP routing is OK between Unity and the voice gateway
    HTH,
    Chris

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    Hi Carlo,
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    codec preference 2 g711alaw
    codec preference 3 g729r8
    codec preference 4 g729br8
    voice class h323 1
    h225 timeout tcp establish 3
    interface Tunnel100
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    tunnel destination 172.31.3.18
    interface FastEthernet0/0
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    interface FastEthernet0/0.20
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    interface FastEthernet0/0.21
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    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
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    impedance complex2
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    supervisory disconnect dualtone mid-call
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    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
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    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/0/2
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
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    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
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    caller-id alerting dsp-pre-allocate
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    output attenuation -3
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    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
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    shutdown
    impedance complex2
    description STC
    voice-port 0/2/1
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    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
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    impedance complex2
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    voice-port 0/3/0
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    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
    impedance complex2
    description STC
    caller-id alerting dsp-pre-allocate
    voice-port 0/3/1
    supervisory disconnect dualtone mid-call
    no battery-reversal
    input gain -3
    output attenuation -3
    echo-cancel coverage 32
    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
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    impedance complex2
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    caller-id alerting dsp-pre-allocate
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    output attenuation -3
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    cptone BE
    timeouts initial 5
    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
    timing hookflash-out 500
    timing guard-out 300
    timing sup-disconnect 50
    connection plar opx 2050
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    caller-id alerting dsp-pre-allocate
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    timeouts interdigit 3
    timeouts call-disconnect 3
    timeouts ringing 5
    timeouts wait-release 1
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    timing guard-out 300
    timing sup-disconnect 50
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    codec g729abr8
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    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
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    dial-peer voice 9002 pots
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    destination-pattern [^2].T
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    dial-peer voice 9003 pots
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    voice-class codec 1
    voice-class h323 1
    session target ipv4:192.168.12.189
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    no vad

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