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  1. Member
    Join Date
    Sep 2010
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    MCSA, CCNA:V, CCNP:V, some day CCIE:V
    #1

    Default CME - outgoing calls did not work

    Hi,

    ě`m a little bit new to the cme world. I have purchased a nice 2811 Router and want now to
    test this in my lab. I have a external voice provider. This is what i have configured at the moment and incoming calls work but outgoing not:


    voice translation-rule 3
    rule 1 /^00/ /+/ type any international
    !
    !
    voice translation-profile SIP-Outgoing
    translate called 3
    !


    !
    dial-peer voice 1 voip
    description "INCOMING"
    destination-pattern 004912345678912 <my international number>
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.provider.de
    incoming called-number .T
    no vad
    !
    dial-peer voice 4 voip
    description "OUTGOING"
    destination-pattern .T
    voice-class codec 1
    session protocol sipv2
    session target dns:sip.provider.de
    dtmf-relay rtp-nte
    no vad
    !

    voice call send-alert
    voice call convert-discpi-to-prog
    voice call carrier capacity active
    voice rtp send-recv
    !
    voice service voip
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    redirect ip2ip
    fax protocol pass-through g711ulaw
    h323
    modem passthrough nse codec g711ulaw
    sip
    bind control source-interface FastEthernet0/0
    session transport tcp
    registrar server expires max 600 min 60
    redirect contact order best-match
    !
    voice class codec 1
    codec preference 1 g711ulaw
    sip-ua
    authentication username
    <username>
    password 7 <password>
    no remote-party-id
    retry invite 2
    retry register 10
    retry options 1
    timers connect 100
    registrar dns:<sip registrar> expires 3600
    sip-server dns:<sip server>
    host-registrar
    dial-peer voice 1 voip
    description SIP TRUNK TO PROVIDER
    destination-pattern <my number in internataional format> extension-length 4 extension-pattern 1001
    session protocol sipv2
    session target dns:<sip server>
    incoming called-number .T
    voice-class codec 1
    no vad
    telephony-service
    no auto-reg-ephone
    max-ephones 20
    max-dn 40
    max-conferences 8 gain -6
    transfer-system full-consult


    ---

    incoming (works)
    ---

    The Call Setup Information is:
    Call Control Block (CCB) : 0x4A63D688
    State of The Call : STATE_DEAD
    TCP Sockets Used : NO
    Calling Number : +49xxxxxxxxxxx (my mobile, 11 digits)
    Called Number : 0049xxxxxxxxxxx (my home, 11 digits)
    Source IP Address (Sig ): 192.168.5.99
    Destn SIP Req Addr:Port : <sip provider>:5060
    Destn SIP Resp Addr:Port : <sip provieder>:5060
    Destination Name : <sip provider>

    *Sep 30 22:01:34.571: //3/962E68E78003/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : g711ulaw
    Negotiated Codec Bytes : 160
    Nego. Codec payload : 0 (tx), 0 (rx)
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0 (tx), 0 (rx)
    Source IP Address (Media): 192.168.5.99
    Source IP Port (Media): 18846
    Destn IP Address (Media): <provider ip>
    Destn IP Port (Media): 40064
    Orig Destn IP Address:Port (Media): [ - ]:0

    ---
    outgoing (works not)
    ---

    The Call Setup Information is:
    Call Control Block (CCB) : 0x4A63D688
    State of The Call : STATE_DEAD
    TCP Sockets Used : YES
    Calling Number : +49xxxxxxxxxxx (my home, 11 digits)
    Called Number : +49xxxxxxxxxxx (my mobile, 11 digits)
    Source IP Address (Sig ): 192.168.5.99
    Destn SIP Req Addr:Port : <sip provider>:5060
    Destn SIP Resp Addr:Port : <sip provider>:5060
    Destination Name : <sip provider>

    Sep 30 22:13:21.864: //20/33C2F5638033/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : No Codec
    Negotiated Codec Bytes : 0
    Nego. Codec payload : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0 (tx), 0 (rx)
    Source IP Address (Media): 192.168.5.99
    Source IP Port (Media): 18180
    Destn IP Address (Media): -
    Destn IP Port (Media): 0
    Orig Destn IP Address:Port (Media): [ - ]:0


    i have tried different outgoing formats (translation rules). I don`t know why i can`t dial out. Any suggestions?
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  3. Senior Member pitviper's Avatar
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    Right Coast, US
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    #2
    Try translating the calling party number to the exact number that is associated with your sip account for the outbound call (looks like 004912345678912). I know some providers (Callcentric is one) will reject a call if the caller-ID doesn’t match your account.
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  4. Member
    Join Date
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    Certifications
    MCSA, CCNA:V, CCNP:V, some day CCIE:V
    #3
    Hi Pitviper,

    sorry for the delay. I was very busy, now i`m working straight on the problem. At the moment it looks like this. I think i missed a little think But i dont know what:

    **CONFIG**

    voice call send-alert
    voice call convert-discpi-to-prog
    voice call carrier capacity active
    voice rtp send-recv
    !
    voice service voip
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    redirect ip2ip
    fax protocol pass-through g711alaw
    h323
    modem passthrough nse codec g711alaw
    sip
    bind control source-interface FastEthernet0/0
    session transport tcp
    registrar server expires max 3600 min 3600
    redirect contact order best-match
    localhost dns:sipgate.de
    !
    voice class codec 1
    codec preference 1 g711alaw
    !
    !
    !
    !
    voice translation-rule 3
    rule 1 /^0/ /+49/
    !
    voice translation-rule 4
    rule 1 /^1001/ /004930129854765/
    !
    voice translation-rule 11
    rule 1 /^/ /0/ type unknown unknown
    rule 2 /^/ /00/ type national national
    rule 3 /^/ /000/ type international international
    !
    voice translation-rule 12
    rule 1 /^004930129854765/ /1001/
    !
    !
    voice translation-profile SIP-Incoming
    translate calling 11
    translate called 12
    !
    voice translation-profile SIP-Outgoing
    translate calling 4
    translate called 3
    !
    !
    !
    dial-peer voice 11 voip
    description **Incoming Call**
    translation-profile incoming SIP-Incoming
    session protocol sipv2
    session target dns:sipgate.de
    incoming called-number .%
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    !
    dial-peer voice 10 voip
    description **Outgoing Call**
    translation-profile outgoing SIP-Outgoing
    destination-pattern 0T
    session protocol sipv2
    session target dns:sipgate.de
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    !
    !
    sip-ua
    credentials username 004930129854765 password 7 <password> realm sipgate.de
    authentication username 004930129854765@sipgate.de password 7 <password>
    no remote-party-id
    retry invite 2
    retry register 10
    retry options 1
    timers connect 100
    registrar dns:sipgate.de expires 3600
    sip-server dns:sipgate.de
    host-registrar
    !
    !
    !
    telephony-service
    no auto-reg-ephone
    max-ephones 20
    max-dn 40
    ip source-address 192.168.5.99 port 2000
    load 7960-7940 P00308010200
    max-conferences 8 gain -6
    web admin system name cisco secret 5 <password>
    dn-webedit
    time-webedit
    transfer-system full-consult
    create cnf-files version-stamp Jan 01 2002 00:00:00
    !
    !
    ephone-dn 1 dual-line
    number 1001 secondary 004930129854765 no-reg primary
    !
    !


    **DEBUG INCOMING**

    *Nov 2 23:05:09.759: //8/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:
    Call Entry Is Not Found
    *Nov 2 23:05:09.763: //-1/F2512023800F/CCAPI/cc_api_display_ie_subfields:
    cc_api_call_setup_ind_common:
    cisco-username=+4917661022799
    ----- ccCallInfo IE subfields -----
    cisco-ani=+4917975322866
    cisco-anitype=0
    cisco-aniplan=0
    cisco-anipi=0
    cisco-anisi=1
    dest=004930129854765
    cisco-desttype=0
    cisco-destplan=0
    cisco-rdie=FFFFFFFF
    cisco-rdn=
    cisco-lastrdn=
    cisco-rdntype=0
    cisco-rdnplan=0
    cisco-rdnpi=-1
    cisco-rdnsi=-1
    cisco-redirectreason=-1 fwd_final_type =0
    final_redirectNumber =
    hunt_group_timeout =0

    *Nov 2 23:05:09.763: //-1/F2512023800F/CCAPI/cc_api_call_setup_ind_common:
    Interface=0x49E2F7A4, Call Info(
    Calling Number=+4917975322866,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=004930129854765(TON=Unknown, NPI=Unknown),
    Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
    Incoming Dial-peer=11, Progress Indication=NULL(0), Calling IE Present=TRUE,
    Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=8




    Guid=F2512023-04DD-11E1-800F-807480744F27, Outgoing Dial-peer=20001
    *Nov 2 23:05:09.775: //8/F2512023800F/CCAPI/cc_api_display_ie_subfields:
    ccCallSetupRequest:
    cisco-username=+4917975322866
    ----- ccCallInfo IE subfields -----
    cisco-ani=+4917975322866
    cisco-anitype=0
    cisco-aniplan=0
    cisco-anipi=0
    cisco-anisi=1
    dest=1001
    cisco-desttype=0
    cisco-destplan=0
    cisco-rdie=FFFFFFFF
    cisco-rdn=
    cisco-lastrdn=
    cisco-rdntype=0
    cisco-rdnplan=0
    cisco-rdnpi=-1
    cisco-rdnsi=-1
    cisco-redirectreason=-1 fwd_final_type =0
    final_redirectNumber =
    hunt_group_timeout =0

    *Nov 2 23:05:09.779: //8/F2512023800F/CCAPI/ccIFCallSetupRequestPrivate:
    Interface=0x4BAD9944, Interface Type=6, Destination=, Mode=0x0,
    Call Params(Calling Number=+4917975322866,(Calling Name=+4917975322866 )(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=1001(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
    Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20001, Call Count On=FALSE,
    Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)


    The Call Setup Information is:
    Call Control Block (CCB) : 0x4AB78130
    State of The Call : STATE_DEAD
    TCP Sockets Used : NO
    Calling Number : +4917975322866
    Called Number : 004930129854765
    Source IP Address (Sig ): 192.168.5.99
    Destn SIP Req Addr:Port : 217.10.79.9:5060
    Destn SIP Resp Addr:Port : 217.10.79.9:5060
    Destination Name : 217.10.79.9

    *Nov 2 23:05:14.575: //8/F2512023800F/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : g711alaw
    Negotiated Codec Bytes : 160
    Nego. Codec payload : 8 (tx), 8 (rx)
    Negotiated Dtmf-relay : 6
    Dtmf-relay Payload : 101 (tx), 101 (rx)
    Source IP Address (Media): 192.168.5.99
    Source IP Port (Media): 17066
    Destn IP Address (Media): 217.10.79.9
    Destn IP Port (Media): 40144
    Orig Destn IP Address:Port (Media): [ - ]:0

    *Nov 2 23:05:14.575: //8/F2512023800F/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC) : 16
    Disconnect Cause (SIP) : 487



    **DEBUG OUTGOING**

    Guid=A9C0A698-04DB-11E1-801B-F8B7FAD2A1EA, Outgoing Dial-peer=10
    Nov 2 22:48:51.650: //12/A9C0A698801B/CCAPI/cc_api_display_ie_subfields:
    ccCallSetupRequest:
    cisco-username=
    ----- ccCallInfo IE subfields -----
    cisco-ani=004930129854765
    cisco-anitype=0
    cisco-aniplan=0
    cisco-anipi=0
    cisco-anisi=0
    dest=+4917975322866
    cisco-desttype=0
    cisco-destplan=0
    cisco-rdie=FFFFFFFF
    cisco-rdn=
    cisco-lastrdn=
    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

    Nov 2 22:48:51.650: //12/A9C0A698801B/CCAPI/ccIFCallSetupRequestPrivate:
    Interface=0x4A07C064, Interface Type=3, Destination=, Mode=0x0,
    Call Params(Calling Number=004930129854765,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
    Called Number=+4917975322866(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
    Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=10, Call Count On=FALSE,
    Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)


    The Call Setup Information is:
    Call Control Block (CCB) : 0x4ADCE9F0
    State of The Call : STATE_DEAD
    TCP Sockets Used : YES
    Calling Number : 004930129854765
    Called Number : +4917975322866
    Source IP Address (Sig ): 192.168.5.99
    Destn SIP Req Addr:Port : 217.10.79.9:5060
    Destn SIP Resp Addr:Port : 217.10.79.9:5060
    Destination Name : sipgate.de

    Nov 2 22:48:56.826: //13/A9C0A698801B/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : No Codec
    Negotiated Codec Bytes : 0
    Nego. Codec payload : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0 (tx), 0 (rx)
    Source IP Address (Media): 192.168.5.99
    Source IP Port (Media): 18588
    Destn IP Address (Media): -
    Destn IP Port (Media): 0
    Orig Destn IP Address:Port (Media): [ - ]:0

    Nov 2 22:48:56.826: //13/A9C0A698801B/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC) : 38
    Disconnect Cause (SIP) : 503

    Hope you see a misconfig. :-/
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  5. Member
    Join Date
    Sep 2010
    Posts
    40

    Certifications
    MCSA, CCNA:V, CCNP:V, some day CCIE:V
    #4
    I have tried to debug this with ccsip messages. For incoming i get the foloowing bu t for outgoing no debug appears. Why? Any idea?

    CME#
    *Nov 4 12:33:48.639: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:004930129854765@192.168.5.99:5060 SIP/2.0
    Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2;n=4>
    Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2;rqu=WkNGWS19QD9U TEBaMHpKNS8xBANoZRB2Gw4SEmslU10/>
    Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2>
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
    Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch =z9hG4bK6b7d.dbf19946.1
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0
    Via: SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch =z9hG4bK608d35d8;rport=5060
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    To: <sip:004930129854765@217.10.79.9>
    Contact: <sip:+41796035695@213.20.246.108>
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    CSeq: 2 INVITE
    Date: Fri, 04 Nov 2011 12:31:03 GMT
    P-Asserted-Identity: +4xxxxxxxxx <sip:+4xxxxxxxxxx@carpo-2.sip.mgc.voip.telefonica.de;user=phone>
    Allow: INVITE,ACK,CANCEL,BYE,REGISTER,OPTIONS,INFO
    Max-Forwards: 49
    Content-Type: application/sdp
    Content-Length: 223
    X-Remote-IP: 217.10.79.9:5060

    v=0
    o=root 19413 19413 IN IP4 213.20.246.108
    s=-
    c=IN IP4 213.20.246.108
    t=0 0
    m=audio 40124 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=sendrecv

    *Nov 4 12:33:48.667: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch =z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch =z9hG4bK608d35d8;rport=5060
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    To: <sip:004930129854765@217.10.79.9>
    Date: Fri, 04 Nov 2011 12:33:48 GMT
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    CSeq: 2 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0


    *Nov 4 12:33:48.711: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch =z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch =z9hG4bK608d35d8;rport=5060
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
    Date: Fri, 04 Nov 2011 12:33:48 GMT
    Call-ID: 1474b05c1ef48d165419e8
    CME#072b6826d3@213.20.246.108
    CSeq: 2 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:004930129854765@192.168.5.99:5060>
    Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2;n=4>,<sip:217.10 .79.9;lr;ftag=156cabf2;rqu=WkNGWS19QD9UTEBaMHpKNS8 xBANoZRB2Gw4SEmslU10/>,<sip:217.10.79.9;lr;ftag=156cabf2>
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0


    CME#
    *Nov 4 12:33:52.327: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    CANCEL sip:004930129854765@192.168.5.99:5060 SIP/2.0
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    To: <sip:004930129854765@217.10.79.9>
    CSeq: 2 CANCEL
    Max-Forwards: 70
    User-Agent: MCCS Dispatcher 1.7.0
    Content-Length: 0


    *Nov 4 12:33:52.347: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    To: <sip:004930129854765@217.10.79.9>
    Date: Fri, 04 Nov 2011 12:33:52 GMT
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    CSeq: 2 CANCEL
    Content-Length: 0


    *Nov 4 12:33:52.351: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 487 Request Cancelled
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch =z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch =z9hG4bK608d35d8;rport=5060
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
    Date: Fri, 04 Nov 2011 12:33:52 GMT
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    CSeq: 2 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=16
    Content-Length: 0


    *Nov 4 12:33:52.435: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:004930129854765@192.168.5.99:5060 SIP/2.0
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
    CSeq: 2 ACK
    Max-Forwards: 70
    User-Agent: MCCS Dispatcher 1.7.0
    Content-Length: 0


    *Nov 4 12:33:52.443: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 487 Request Cancelled
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch =z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch =z9hG4bK608d35d8;rport=5060
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
    Date: Fri, 04 Nov 2011 12:33:52 GMT
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    CSeq: 2 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=16
    Content-Length: 0


    CME#
    *Nov 4 12:33:52.503: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:004930129854765@192.168.5.99:5060 SIP/2.0
    Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
    From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
    Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
    To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
    CSeq: 2 ACK
    Max-Forwards: 70
    User-Agent: MCCS Dispatcher 1.7.0
    Content-Length: 0


    *Nov 4 12:33:52.543: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ...
    *Nov 4 12:33:52.543: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ...
    CME#
    *Nov 4 12:34:02.835: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ...
    *Nov 4 12:34:02.835: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ...
    CME#
    *Nov 4 12:34:13.015: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ...
    *Nov 4 12:34:13.015: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ...
    CME#
    Last edited by woisch; 11-04-2011 at 03:35 PM.
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  6. Member
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    #5
    Ok. it worked.
    I have added "session transport udp" to the outgoing (and incoming) dial-peer. That was it.
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  7. Senior Member pitviper's Avatar
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    #6
    Hmm, that’s odd – never had to use that command before. I wonder if it’s ITSP specific.
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  8. Went to the dark side.... Moderator networker050184's Avatar
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    #7
    Quote Originally Posted by woisch View Post
    Hi Pitviper,
    sip
    bind control source-interface FastEthernet0/0
    session transport tcp
    registrar server expires max 3600 min 3600
    redirect contact order best-match
    localhost dns:sipgate.de
    !
    There's your issue. SIP defaults to UDP so you shouldn't need to set to UDP on your dial-peers. Looks like that gloabal SIP setting was fudging you up though.
    An expert is a man who has made all the mistakes which can be made.
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  9. Senior Member pitviper's Avatar
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    #8
    That would explain it! Good eye - I didn't even notice it in the config.
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  10. Member
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    #9
    Thanks for your replys.
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