Max no. of call-legs?

Hi, I've a 2811 with a ISDN PRI to the PSTN and a PVDM2-64. The router is configured for receiving and making calls and is working fine. The problem is I never seem to be able to make more than 16 calls on my PRI. Can anyone advise where i should look?

Perhaps your telco activated only 16 channels, check with them.

Similar Messages

  • Max no of calls that can be queued in ICM

    Where can I find for a given installation the max no of calls that can queued for given ICM system.

    I looked at that but I think the answer is still not well defined. It talks about max call per second.That means that it can receive 300 max calls per second in ICM/IVR system.  It does not say anything about how many concurrent calls can sit in the system and be queued.
    For example on IVR side, we clearly know that it can support X no of  max concurrent calls or whatever the figure may be based on ports.  However for queued calls in ICM, it does not seem to very obvious. Unless I am missing something here.

  • SIP reason = "This call leg has been replaced"

    Hi All,
    I am trying to troubleshoot an issue with calls being dropped, and have noticed that on some of my dropped calls I see the following in the BYE message:
    ms-diagnostics: 10026;source="Lync.domain.local";reason="This call leg has been replaced";component="MediationServer"
    ms-diagnostics-public: 10026;reason="This call leg has been replaced";component="MediationServer"
    ms-endpoint-location-data: NetworkScope;ms-media-location-type=intranet
    The call path is PSTN-->Gateway-->Lync-->Trusted application Server-->Agent
    I cant seem to find much on what the "This call leg has been replaced" message means, if anyone can help me out it would be appreciated.
    Cheers
    J

    All,
    I have now had some time to analyse the problem further. I have produced a more detailed way of reviewing the data. By using a custom SQL view (see attached) in the LCSCDR DB I can
    now connect Excel to the database and using Power Pivot quickly breakdown the data and look for some form of trend. Now I can see a the precise time, client versions and affected users. I have compared this information with the Syslog outputs on the Audio
    Codes gateway.
    It seems that all the problems I have generally occur when clients are connected to only one of my front end servers. There does not appear to be a similar number of problems on the other one.
    Can anyone else please confirm if they too have an Audio Codes gateway when suffering with this issue?
    I now have CU6 applied to all of the Lync infrastructure. This has not resolved the problem but has reduced the frequency which the errors are occurring.
    Does anyone else have any further thoughts on this?
    Thanks
    Ben

  • Every Since I Downloaded iOS5 O My IPhone My Volume Has Been Effected It's @ Max But Incoming Calls Come In @ A Minimum. When Taking A Call Thru Speaker Caller Sound Very Faint. Music Sounds Very Low Can Somebody Help Me ?

    Every Since I Downloaded iOS5 O My IPhone My Volume Has Been Effected It's @ Max But Incoming Calls Come In @ A Minimum. When Taking A Call Thru Speaker Caller Sound Very Faint. Music Sounds Very Low Can Somebody Help Me ?

    Hello ElleMalloy,
    Sorry Im Just Responding Phone Has Been Downloading And Charging.
    Well I Will Start Here,
    I Tried Everything I Even Restarted My Settings Just So I Can See if That Would Work Because Someone Sujested It But What They Neglected To Say That Their Probrolem Was With A Ipad ? Anywho To Keep It Short I Went Online Logged A Complaint With Apple Made A Appointment With A Apple Store In My Area To Take The Phone There After Trying Everything (Resetting, changing setting etc..........) Nothin Worked Upon Taking The Phone To The Store They Swapped Out My Phone For A New One Volume On New Phone Is Perfect And I Only Had To Wait 40 Minutes

  • How to set busy trigger and max number of calls for a sip phone in SRST

    How is it possible to set the maximum number of calls and busy trigger for a line on a sip phone in SRST .There is no commands like dual-line or octo-line for the max-dn like under call-manager-fallback .
    Sent from Cisco Technical Support iPad App

    So is the answer then that there is no way for a SIP phone in SRST mode to handle multiple calls? I have a site currently failed over in SRST mode, and I have noticed that the reception phone can only handle one call at a time. If I place a second call to it I get a busy signal.

  • CUCM 9.x with Gateway - Troubleshoot call leg

    Guys,
    We have a CUCM 9.x at a main site (Site-A),  with a h.323 Gateway.
    The h.323 Gateway have a Sip Dial-peer with another cisco pbx in another Country (Site-B).
    In the past, people at Site-B used to called 816XXX to reach Site-B devices, and People in site-A can dial 83XXXX to reach people at site-B.
    I know that some change occured in the last two weeks, since that it longer work for inter-site call.
    I know that the IOS of Site-A has been updated to a 15.1   ( BUT we did add the IP address truested command to trust all PBX )
    But still it's not working..
    I did a debug voip dialpeer,  and I would like to have your opinion
    When user from Site B dial : 816999
    It’s going well to our router at Bellevue, as expected
    040331: 5d10h: //-1/856FEA4E80C4/DPM/dpAssociateIncomingPeerCore:
       Calling Number=83644, Called Number=6999, Voice-Interface=0x0,
    I see there is a successful match with dial-peer 6999
    040341: 5d10h: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=6999
    This dial-peer 6999 point to Call Manager
    dial-peer voice 6999 voip
    description Outgoing to CUCM
    destination-pattern 6...
    session target ipv4:10.AA.XZ.ZZ
    dtmf-relay h245-alphanumeric
    And one have a recommendation ?

    You will need to see in depth into the h225 and h245 signalling as well ass the CCAPI process..
    You should do this
    debug voip ccapi inout
    debug h225 asn1
    debug h245 asn1
    To capture all of these logs..you need to properly setup your gateway as follows
    service sequence-numbers
    service timestamps debug datetime localtime msec
    logging buffered 10000000 debug
    no logging console
    no logging monitor
    default logging rate-limit
    default logging queue-limit
    Then..
    <Enable debugs, then test again.>
    <Enable session capture to txt file in terminal program.> (such as Putty)
    then do the ff:
    terminal length 0
    show logging
    You can attach the logs here in a text file

  • Max no of calls - Busy Trigger

    Hi,
    I don't know if this is a standard settings or not?
    When I put 6 to maximum nr and 2 to Busy trigger & when I am on the call and somebody trying to call me he receive busy and the call is remaining on
    missed calls.
    But when  I put 6 to maximum nr and 1 to Busy trigger & when I am on the call and somebody trying to call me he receive busy and the call is not remaining on missed calls
    What I want is when some body is on the call and he receive a second call, the person who trying to call to receive busy and the call to remain in missed calls.
    I have CallM 8.6

    I have configured it some time with AndPhone (http://andtek.com/communications-products-calllist.html).
    The Busy Call is forwarded to an CTI-RP which is controlled by andphone. Andphone evaluates the forwarding party and then forwards the call to a busy extension (e.g. an dial-peer on an h.323 gateway where busy signal is configured).
    I'm sure there are other 3rd party solutions.

  • UCCX - Control call legs

    Hi guys,
    We have a UCCX 8.5 cluster and we want to set up a feature like IVR Rebound: caller talks to agent, and then agent can conference the caller to another IVR (internal or external IVR) for the caller to process the authorization, during the authorization process, agent must be on hold (cannot hear/interupt the authorization process), after the authorization is done (either successful or not), agent is connected to caller again.
    Understand that it is not something nature to UCCX, I'm thinking of setting up system like this:
    1. Caller calls to CallCenter and connect to agent.
    2. Agent transfers (transfer, not conference) the call to an IVR script.
    3. IVR accepts the call and put the call on hold. Now the caller is connected to IVR, agent is free.
    4. IVR calls back to agent (as the agent will need to do something with the 2nd IVR) --> we have 3 party conference: IVR, agent, caller.
    5. IVR makes another call to the 2nd IVR (can be external IVR, which is used for authorization) --> we have 4 party conference: IVR, agent, caller, IVR#2.
    6. Agent key in some information needed by IVR#2.
    7. IVR puts agent on hold and unhold the caller so the caller can proceed the authorization himself.
    8. Authorization is done, IVR#2 disconnects, IVR needs to unhold agent so the agent and caller can talk to each other again.
    My questions are:
    Q1) In step 6 & 7: can IVR script detects a key sequence to determine when to start putting agent on hold. I'm thinking of Menu Grammar step but not sure if it is OK.
    Q2) In step 8: can we catch the disconnection event (when IVR#2 disconnects) by using exception Channel Inactive Exception? I tried but when the IVR2 disconnects, the caller and agent cannot connect to each other but can only hear ringback tone/busy tone.
    Thanks,
    hoanghiep

    CCX has no ability to conference separate contacts together so this approach will not work. I can think of two options:
    The agent transfers the caller to the authorization IVR at the conclusion of the call and checks for an approval status before processing the order. This is the cleanest approach from a CCX reporting perspective.
    The agent transfers the caller to the external IVR which accepts an agent-specific identifier and transfers the caller back to the agent after authorization. You can use CCX or an external platform for this; the experience is essentially the same.
    Please remember to rate helpful responses and identify helpful or correct answers.

  • Unity Express - Incoming calls wont get voice mail

    CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.
    However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.
    I searched for another post which suggested the following commands:
    telephony-service
    call-forward pattern .T
    voice service voip
    allow connections from h323 to sip
    I've double checked them and there's still something wrong.
    Here's my current configuration:
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    h323
    voice class codec 1
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    telephony-service
    load 7910 P00403020214
    load 7960-7940 P00305000301
    max-ephones 24
    max-dn 24
    ip source-address 192.168.20.1 port 2000
    auto assign 1 to 24
    system message Comtek
    voicemail 3000
    max-conferences 8 gain -6
    call-forward pattern .T
    moh music-on-hold.au
    time-webedit
    transfer-system full-consult
    transfer-pattern 2...
    transfer-pattern 3...
    directory last-name-first
    directory entry 2 2001 name Phone Two 7912
    directory entry 3 2000 name Phone One 7970
    ephone-dn 1 dual-line
    number 2000 secondary 441833000000
    call-forward busy 3000
    call-forward noan 3000 timeout 10
    no huntstop
    ephone 1
    no multicast-moh
    device-security-mode none
    mac-address 0017.0EF0.3642
    type 7970
    button 1:1
    So pros, any suggestions?
    Thanks

    I made a new dial-peer to handle incoming calls as follows.
    dial-peer voice 1000 voip
    description Incoming SIP
    translation-profile incoming SIPin
    voice-class codec 1
    session protocol sipv2
    incoming called-number .T
    dtmf-relay rtp-nte
    no vad
    The translation-profile puts the call through to my 2000 extension.
    This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.
    To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec
    Telephony call-legs: 1
    SIP call-legs: 1
    H323 call-legs: 0
    Call agent controlled call-legs: 0
    SCCP call-legs: 0
    Multicast call-legs: 0
    Total call-legs: 2
    1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting
    dur 00:00:00 tx:0/0 rx:0/0
    IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
    media inactive detected:n media contrl rcvd:n/a timestamp:n/a
    long duration call detected:n long duration call duration:n/a timestamp:n/a
    1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting
    dur 00:00:00 tx:0/0 rx:0/0
    Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm
    Telephony call-legs: 1
    SIP call-legs: 1
    H323 call-legs: 0
    Call agent controlled call-legs: 0
    SCCP call-legs: 0
    Multicast call-legs: 0
    Total call-legs: 2
    This is the "show call active voice brief" for an external incoming call when the call is established.
    Telephony call-legs: 1
    SIP call-legs: 1
    H323 call-legs: 0
    Call agent controlled call-legs: 0
    SCCP call-legs: 0
    Multicast call-legs: 0
    Total call-legs: 2
    1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active
    dur 00:00:02 tx:105/16800 rx:104/16640
    IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
    media inactive detected:n media contrl rcvd:n/a timestamp:n/a
    long duration call detected:n long duration call duration:n/a timestamp:n/a
    1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active
    dur 00:00:02 tx:0/0 rx:105/16800
    Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm
    Telephony call-legs: 1
    SIP call-legs: 1
    H323 call-legs: 0
    Call agent controlled call-legs: 0
    SCCP call-legs: 0
    Multicast call-legs: 0
    Total call-legs: 2
    Not too sure where to go from here.

  • Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP

    Hi Cisco Community,
    I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
    On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
    That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
    The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
    I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
    Below is an example of a call that is connected with the current setup:
    Note:
    IP: 10.18.81.2 (CUBE)
    IP: 10.18.81.11 (CUCM SUB)
    IP: 10.111.111.254 (ITSP SBC)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    Session-Expires:  1800
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 301
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
    s=SIP Call
    c=I
    PM-HO-VG-01#N IP4 10.18.81.2
    t=0 0
    m=audio 22256 RTP/AVP 18 0 8 101
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf9
    PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: application/media_control+xml,application/sdp,application/xml
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Type: application/sdp
    Content-Length: 236
    v=0
    o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.80.40
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    PM-HO-VG-01#
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    PM-HO-VG-01#sh sip
    PM-HO-VG-01#sh sip-ua call
    PM-HO-VG-01#sh sip-ua calls 
    Total SIP call legs:2, User Agent Client:1, User Agent Server:1
    SIP UAC CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 27218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC04018 0x10000100 0x0
       CC Call ID              : 64511
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.111.111.254]:5060
       Destn SIP Resp Addr:Port: [10.111.111.254]:5060
       Destination Name        : 10.111.111.254
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64511
         Stream Type              : voice+dtmf (0)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22256
         Media Dest IP Addr:Port  : [10.111.111.254]:20074
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Client(UAC) calls: 1
    SIP UAS CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 0218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC0401E 0x10000100 0x80004
       CC Call ID              : 64510
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.18.81.11]:5060
       Destn SIP Resp Addr:Port: [10.18.81.11]:5060
       Destination Name        : 10.18.81.11
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64510
         Stream Type              : voice+dtmf (1)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22350
         Media Dest IP Addr:Port  : [10.18.80.40]:21928
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Server(UAS) calls: 1
    PM-HO-VG-01#
    PM-HO-VG-01#
    PM-HO-VG-01#
    As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
    NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22256 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 360
    v=0
    o=BroadWorks 316169737 2 IN IP4 10.111.111.254
    s=-
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    a=inactive
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22350 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    Content-Type: application/sdp
    Content-Length: 306
    v=0
    o=BroadWorks 316169737 3 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 2
    PM-HO-VG-01#00 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 213
    v=0
    o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.81.10
    t=0 0
    m=audio 4000 RTP/AVP 18
    a=X-cisco-media:umoh
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=sendonly
    Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 101 BYE
    Reason: Q.850;cause=86
    P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 104 BYE
    Reason: Q.850;cause=65
    P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 Race Condition
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    Timestamp: 1417347889
    CSeq: 104 BYE
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 65
    Disconnect Cause (SIP)   : 200
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 101 BYE
    Content-Length: 0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 86
    Disconnect Cause (SIP)   : 200
    PM-HO-VG-01#

    Hi Manish,
    Again, excellent feedback. Much appreciated.
    I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
    But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
    If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
    One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
    I will be doing some intensive test again later on this week and will send the logs. 
    Here is my question to both of you:
    Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
    Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
    From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
    I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
    Thanks again for your support fellows.

  • Can't make outgoing call with Skype Connect

    I have my Asterisk PBX configured with Skype Connect using SIP with TLS and SRTP. Most of my outgoing calls go through, but sometimes I can't get call out. I was able to leave asterisk console up and collect verbose and sip debug data. Can somebody help me diagnose why my calls aren't going through?
    I've changed my external IP (I'm behind a NAT'd firewall) to 1.2.3.4 and my SIP profile's user ID to 11111111111111. and my domain name to test.com. If someone working for Skype needs that information they can email me and I'll send it privately.
    My config:
    [general]
    context=default_context
    allowguest=no
    alwaysauthreject=yes
    allowoverlap=no
    udpbindaddr=0.0.0.0
    tlsenable=yes
    tlsbinddir=0.0.0.0
    tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
    tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
    tlscipher=ALL
    tlsclientmethod=tlsv1
    tcpenable=yes
    tcpbindaddr=0.0.0.0
    transport=udp,tcp,tls
    srvlookup=yes
    dynamic_exclude_static = yes
    buggymwi=yes
    contactpermit=192.168.1.0/24
    register => tls://111111111111111:[email protected]
    [skype]
    type=friend
    context=from-skype
    dtmfmode=rfc2833
    host=sip.skype.com
    username=11111111111111
    fromuser=11111111111111
    secret=abcd12345
    disallow=all
    allow=ulaw
    allow=alaw
    nat=yes
    fromdomain=sip.skype.com
    insecure=port,invite
    transport=tls
    srtpcapable=yes
    encryption=yes
    SIP Debugging enabled
    [2012-08-23 19:22:33] NOTICE[16552]: chan_sip.c:13465 sip_reregister: -- Re-registration for [email protected]
    > doing dnsmgr_lookup for 'sip.skype.com'
    > ast_get_srv: SRV lookup for '_sips._tcp.sip.skype.com' mapped to host 1.sip.skype.com, port 5061
    REGISTER 11 headers, 0 lines
    Reliably Transmitting (NAT) to 63.209.144.201:5061:
    REGISTER sip:sip.skype.com:5061 SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport
    Max-Forwards: 70
    From: <sip:[email protected]>;tag=as6edf93cf
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 32495 REGISTER
    User-Agent: Asterisk PBX 10.5.2
    Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com:5061", nonce="5036b5770000182c78c1d1909cfd5c74e33f033c952d240d", response="81001ceacd91b16ebb956d3c55991471"
    Expires: 120
    Contact: <sip:[email protected]:5061;transport=TLS>
    Content-Length: 0
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 200 OK
    From: <sip:[email protected]>;tag=as6edf93cf
    To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
    Call-ID: [email protected]
    CSeq: 32495 REGISTER
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport=50541;received=1.2.3.4
    Expires: 45
    Contact: <sip:[email protected]:5061;transport=tls>;expires=45
    Content-Length: 0
    <------------->
    --- (9 headers 0 lines) ---
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
    [2012-08-23 19:22:33] NOTICE[17932]: chan_sip.c:21399 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
    <--- SIP read from UDP:192.168.1.16:5060 --->
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Max-Forwards: 70
    Contact: "Scott's Office" <sip:[email protected]:5060>
    Expires: 240
    User-Agent: Cisco/SPA504G-7.5.2b
    Content-Length: 234
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
    Supported: replaces
    Content-Type: application/sdp
    v=0
    o=- 88651316 88651316 IN IP4 192.168.1.16
    s=-
    c=IN IP4 192.168.1.16
    t=0 0
    m=audio 16484 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv
    <------------->
    --- (14 headers 12 lines) ---
    Sending to 192.168.1.16:5060 (NAT)
    Using INVITE request as basis request - [email protected]
    Found peer 'scott_office' for 'scott_office' from 192.168.1.16:5060
    == Using SIP RTP CoS mark 5
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format PCMU for ID 0
    Found audio description format PCMA for ID 8
    Found audio description format telephone-event for ID 101
    Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.1.16:16484
    Looking for 19739928881 in home (domain asterisk.test.com)
    list_route: hop: <sip:[email protected]:5060>
    <--- Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    <------------>
    -- Executing [19739928881@home:1] Dial("SIP/scott_office-000000b0", "SIP/skype/+19739928881") in new stack
    == Using SIP RTP CoS mark 5
    Audio is at 9302
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 63.209.144.201:5061:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 10.5.2
    Date: Thu, 23 Aug 2012 23:22:34 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 370
    v=0
    o=root 1671301052 1671301052 IN IP4 192.168.1.15
    s=Asterisk PBX 10.5.2
    c=IN IP4 192.168.1.15
    t=0 0
    m=audio 9302 RTP/SAVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
    -- Called SIP/skype/+19739928881
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 100 Trying
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 407 Proxy Authentication Required
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    set_destination: Parsing <sip:[email protected]> for address/port to send to
    set_destination: set destination to 63.209.144.201:5060
    Transmitting (NAT) to 63.209.144.201:5061:
    ACK sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 10.5.2
    Content-Length: 0
    Audio is at 9302
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 63.209.144.201:5061:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 10.5.2
    Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
    Date: Thu, 23 Aug 2012 23:22:34 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 370
    v=0
    o=root 1671301052 1671301053 IN IP4 192.168.1.15
    s=Asterisk PBX 10.5.2
    c=IN IP4 192.168.1.15
    t=0 0
    m=audio 9302 RTP/SAVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 100 Trying
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]' Method: REGISTER
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 180 Ringing
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    User-Agent: SipGW 8
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (9 headers 0 lines) ---
    list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    -- SIP/skype-000000b1 is ringing
    <--- Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    <------------>
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 408 Request Timeout
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Content-Length: 0
    <------------->
    --- (7 headers 0 lines) ---
    [2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
    set_destination: Parsing <sip:[email protected]> for address/port to send to
    set_destination: set destination to 63.209.144.201:5060
    Transmitting (NAT) to 63.209.144.201:5061:
    ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 103 ACK
    User-Agent: Asterisk PBX 10.5.2
    Content-Length: 0
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
    == Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
    <--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 503 Service Unavailable
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0
    <------------>
    <--- SIP read from UDP:192.168.1.16:5060 --->
    ACK sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 ACK
    Max-Forwards: 70
    Contact: "Scott's Office" <sip:[email protected]:5060>
    User-Agent: Cisco/SPA504G-7.5.2b
    Content-Length: 0
    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]' Method: INVITE
    Really destroying SIP dialog '[email protected]' Method: ACK

    I wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
    Scott
    [general]
    context=default_context
    allowguest=no
    alwaysauthreject=yes
    allowoverlap=no
    udpbindaddr=0.0.0.0
    tlsenable=yes
    tlsbinddir=0.0.0.0
    tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
    tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
    tlscipher=ALL
    tlsclientmethod=tlsv1
    tcpenable=yes
    tcpbindaddr=0.0.0.0
    transport=udp,tcp,tls
    srvlookup=yes
    dynamic_exclude_static = yes
    buggymwi=yes
    contactpermit=192.168.1.0/24
    register => tls://[email protected]
    [skype]
    type=friend
    context=from-skype
    dtmfmode=rfc2833
    host=sip.skype.com
    username=user
    fromuser=user
    secret=pass
    disallow=all
    allow=ulaw
    allow=alaw
    nat=yes
    fromdomain=sip.skype.com
    insecure=port,invite
    transport=tls
    srtpcapable=yes
    encryption=yes

  • Show call history voice brief output

    The output as below;
    tok-fr-rtr>sh call hist voice brief
    <ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
    dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
    IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
    delay:<last>/<min>/<max>ms <codec>
    MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
    last <buf event time>s dur:<Min>/<Max>s
    FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
    <codec> (payload size)
    ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
    <codec> (payload size)
    Telephony <int> (callID) [channel_id] tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
    MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> disc:<cause code>
    speeds(bps): local <rx>/<tx> remote <rx>/<tx>
    Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
    bw: <req>/<act> codec: <audio>/<video>
    tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
    rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
    Telephony call-legs: 105
    SIP call-legs: 0
    H323 call-legs: 105
    MGCP call-legs: 0
    Total call-legs: 210
    28CD : 232788573hs.5992 +162 +24465 pid:8712 Originate 8717898
    dur 00:04:03 tx:11998/239960 rx:12151/243020 10 (normal call clearing (16))
    IP 192.168.139.36:17424 rtt:247ms pl:237520/220ms lost:0/22/0 delay:50/50/70ms g729r8
    28CD : 232788264hs.5993 +471 +24748 pid:46201 Answer
    dur 00:04:02 tx:12139/242780 rx:11998/239960 10 (normal call clearing (16))
    Telephony 3/0:0 (6048) [3/0.2] tx:235590/235590/0ms g729r8 noise:-82dBm acom:32dBm
    1E6B : 232806587hs.5994 +417 +21065 pid:1 Answer
    dur 00:03:26 tx:10447/208940 rx:10314/206280 10 (normal call clearing (16))
    IP 192.168.139.104:19276 rtt:342ms pl:204470/10ms lost:0/0/0 delay:60/50/70ms g729r8
    1E6B : 232806588hs.5995 +416 +21030 pid:46201 Originate 4622916
    dur 00:03:26 tx:10297/205940 rx:10447/208940 10 (normal call clearing (16))
    Telephony 3/0:0 (6051) [3/0.3] tx:207560/207560/0ms g729r8 noise:-82dBm acom:41dBm
    Can someone tell me what does the value for pl:<play>/<gap>ms is refering to?
    cheers!
    csyeo

    anitachoi3,
    You can use the below command to clear a voice call. Please exercise caution to ensure that this is not an on-going valid conversation.
    look at the output of 'show call active voice brief' and grab the ID. Let's say you want to clear the call with the ID 7D4 in your example. Execute the below CLI:
    clear call voice causecode 16 id 7D4
    Execute 'show call active voice brief' to see if the call is cleared.

  • Calls are not getting thru in Cisco voice GW for a particular Number

    Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
    the output of the Q931 debug :
    Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref = 0x7E05
            Bearer Capability i = 0x8090A2
                    Standard = CCITT
                    Transfer Capability = Speech
                    Transfer Mode = Circuit
                    Transfer Rate = 64 kbit/s
            Channel ID i = 0xA98396
                    Exclusive, Channel 22
            Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
    1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
    938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
    F4DA50C06062B0C02FF373730020500
            Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
    F204C4F4E472044495354414E4345
            Calling Party Number i = 0x2183, '8168911010'
                    Plan:ISDN, Type:National
            Called Party Number i = 0x89, '18553808521'
                    Plan:Private, Type:Unknown
            Sending Complete
    Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0xF
    E05
            Channel ID i = 0xA98396
                    Exclusive, Channel 22
    Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x
    FE05
            Cause i = 0x80BF - Service/option not available, unspecified
    Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x7E0
    5
    Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref =
    0xFE05
    The Qsig and dial-peer configration :
    interface Serial0/0/0:23
    no ip address
    encapsulation hdlc
    isdn switch-type primary-qsig
    isdn overlap-receiving
    isdn incoming-voice voice
    isdn send-alerting
    no cdp enable
    dial-peer voice 1 voip
    description To CBTS GK
    destination-pattern +1T
    signaling forward rawmsg
    session protocol sipv2
    session target ipv4:10.9.5.10
    session transport tcp
    voice-class codec 1
    dtmf-relay rtp-nte
    no vad
    interface Serial0/0/0:23
    no ip address
    encapsulation hdlc
    isdn switch-type primary-qsig
    isdn overlap-receiving
    isdn incoming-voice voice
    isdn send-alerting
    no cdp enable
    dial-peer voice 1 voip
    description To CBTS GK
    destination-pattern +1T
    signaling forward rawmsg
    session protocol sipv2
    session target ipv4:10.9.5.10
    session transport tcp
    voice-class codec 1
    dtmf-relay rtp-nte
    no vad

    Hi Raj,
    My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
    According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
    dial-peer voice 1 voip
    destination-pattern 1T
    The T is a wild card for any digit any length
    Or you can be very specific.
    dial-peer voice 1 voip
    destinaton-pattern 18553808521
    The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
    Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
    But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
    The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
    Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
            Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
    1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
    938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
    F4DA50C06062B0C02FF373730020500
    decode -->
    Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
    **Note:
    **0x91/0x9f both be used by older qsig spec, including:
    **ISO 11582:1995, ETSI 300 239:1993/1995
    **newer qsig spec use 0x9f only, including:
    **ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
    **see CSCeb58118 for CCM compatibility issue
    NetworkFacilityExtension ::= {
    sourceEntity: 0
    destinationEntity: 0
    NetworkProtocolProfile not present
    APDU is a ROSE
    0
    DivertingLegInformation2Invoke ::= {
    invokeID: 1793
    operationValue: 21
    argument: DivertingLegInformation2Arg ::= {
    diversionCounter: 1
    diversionReason: 1
    originalDiversionReason: 1
    divertingNr: PrivatePartyNumber ::= {
    privateTypeOfNumber: 2
    privateNumberDigits: 50005998
    originalCalledNr: PrivatePartyNumber ::= {
    privateTypeOfNumber: 2
    privateNumberDigits: 50005998
    redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
    originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
    Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
    However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
    Here are some good documents on ISDN, IOS dial-peers and call legs:
    Understanding debug isdn q931 Disconnect Cause Codes
    http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
    Configuring Telephony Call-Redirect Features
    Two B-Channel Transfer
    http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
    Understanding Dial Peers and Call Legs on Cisco IOS Platforms
    http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
    Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
    http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
    Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
    http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
    Voice Translation Rules
    http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
    Let me know how you go.
    Thanks again for asking the tuff questions.
    Cheers
    Edson

  • Dial-Peer matches but fails to call out

    Hello,
    Am trying to get my CME configured for Callcentric.  I have both an inbound and an outbound plan.
    With my dial-peers configured for standard 11-digit and 10-digit dialing, calls go to fast busy after all digits except the last two are dialed.  Debug shows a dial-peer match initially, then states no match and the call fails.  If I change the destination pattern to match my cell phone number exactly, I can dial all the digits but the call still fails.  Anyone have a suggestion?
    Here are my dial peers:
    dial-peer voice 700 voip
    description SIP Trunk - Incoming
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target dns:callcentric.com
    incoming called-number .%
    dial-peer voice 701 voip
    description SIP Trunk - Outgoing 3-Digit Calls
    translation-profile outgoing SIP_1
    preference 1
    destination-pattern 9[2-8]11
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    dial-peer voice 702 voip
    description SIP Trunk - Outgoing 11-Digit Calls
    translation-profile outgoing SIP_1
    preference 1
    destination-pattern 91[2-9].......
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    dial-peer voice 703 voip
    description SIP Trunk - Outgoing 10-Digit Calls
    translation-profile outgoing SIP_1
    preference 1
    destination-pattern 9[2-9].......
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    no vad
    And here is the debug associated with a call:
    *Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=7018, Called Number=, Voice-Interface=0x4A4AE7B0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20009
    GMIT-VOICEROUTER01#
    *Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9
    *Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=91, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=91
    *Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Numbe
    GMIT-VOICEROUTr=, Called Number=912, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=912
    *Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120
    *Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=91207, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=91207
    *Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=912072, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=912072
    *Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:31.542: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120722, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722
    *Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=91207227, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=91207227
    *Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=912072277, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=912072277
    *Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    *Dec 26 22:26:32.606: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    *Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722776
    *Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=702
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722776
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=702
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722776
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    *Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=702
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=9120722776, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722776
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=702
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=NO_MATCH(-1) After All Match Rules Attempt
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=9120722776, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=702
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722776
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    *Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=702
    *Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9120722776
    *Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    *Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=702
    *Dec 26 22:26:33.574: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=91[2-9]......., Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:33.578: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=NO_MATCH(-1) After All Match Rules Attempt
    *Dec 26 22:26:36.374: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=7018$, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    *Dec 26 22:26:36.378: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=NO_MATCH(-1) After All Match Rules AttemptER01#

    Translation profile:
    voice translation-rule 3
    rule 1 /^7../ /2072267262/
    voice translation-rule 4
    rule 1 /^9\(1....\)/ /\1/
    rule 2 /^9207\(...\)/ /\1/
    rule 3 /^9\(011.*\)/ /\1/
    rule 4 /^9\([2-9]11\)/ /\1/
    voice translation-profile SIP_1
    translate calling 3
    translate called 4
    Here is debug ccsip messages:
    *Dec 27 14:10:16.598: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
    Max-Forwards: 69
    Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
    From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
    CSeq: 938331054 OPTIONS
    Organization: MetaSwitch
    Supported: resource-priority, 100rel
    Content-Length: 0
    Contact:
    To:
    *Dec 27 14:10:16.606: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
    From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
    To:
    GMIT-VOICEROUT166>;tag=F1B5120-18BD
    Date: Fri, 27 Dec 2013 14:10:16 GMT
    Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 938331054 OPTIONS
    Supported: 100rel,resource-priority,replaces,sdp-anat
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Accept: application/sdp
    Content-Type: application/sdp
    Content-Length: 172
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4484 7548 IN IP4 66.55.220.166
    s=SIP Call
    c=IN IP4 66.55.220.166
    t=0 0
    m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
    c=IN IP4 66.55.220.166
    ER01#
    GMIT-VOICEROUTER01#
    *Dec 27 14:10:34.834: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5080 SIP/2.0
    Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
    From: "Server Room" [email protected]>;tag=F1B9854-8A5
    To: [email protected]>
    Date: Fri, 27 Dec 2013 14:10:34 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 2066961728-1849102819-2185007278-567139419
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, B
    GMIT-VOICEROUTER01#YE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1388153434
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 297
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
    s=SIP Call
    c=IN IP4 66.55.220.166
    t=0 0
    m=audio 19258 RTP/AVP 18 101 19
    c=IN IP4 66.55.220.166
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:19 CN/8000
    a=ptime:20
    *Dec 27 14:10:34.906: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 407 Proxy Authentication Required
    v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
    f: "Server Room" [email protected]>;tag=F1B9854-8A5
    t: [email protected]>
    i: [email protected]
    CSeq: 1
    GMIT-VOICEROUT01 INVITE
    Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="3755ae79fd668c2035ebb90cdc12d030", opaque="", stale=TRUE, algorithm=MD5
    l: 0
    *Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5080 SIP/2.0
    Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
    From: "Server Room" [email protected]>;tag=F1B9854-8A5
    To: [email protected]>
    Date: Fri, 27 Dec 2013 14:10:34 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    *Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5080 SIP/2.0
    Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
    From: "Server Room" [email protected]>;tag=F1B9854-8A5
    To: [email protected]>
    Date: Fri, 27 Dec 2013 14:10:34 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 2066961728-1849102819-2185007278-567139419
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1388153434
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="17772882353",realm="callcentric.com",uri="sip:[email protected]:5080",response="cbac03a76a23b6a35ebbee966c00a577",nonce="3755ae79fd668c2035ebb90cdc12d030",opaque="",algorithm=MD5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 297
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
    s=SIP Call
    c=IN IP4 66.55.220.166
    t=0 0
    m=audio 19258 RTP/AVP 18 101 19
    c=IN IP4 66.55.220.166
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:19 CN/8000
    a=ptime:20
    *Dec 27 14:10:34.990: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 403 Incorrect Authentication
    v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
    f: "Server Room" [email protected]>;tag=F1B9854-8A5
    t: [email protected]>
    i: [email protected]
    CSeq: 102 INVITE
    l: 0
    *Dec 27 14:10:35.002: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5080 SIP/2.0
    Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
    From: "Server Room" [email protected]>;tag=F1B9854-8A5
    To: [email protected]>
    Date: Fri, 27 Dec 2013 14:10:34 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Here is debug voip ccapi inout:
    GMIT-VOICEROUTER01#debug voip ccapi inout
    voip ccapi inout debugging is on
    GMIT-VOICEROUTER01#
    *Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
       cc_api_call_setup_ind_common:
       cisco-username=
       ----- ccCallInfo IE subfields -----
       cisco-ani=7018
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=0
       dest=
       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
    *Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
       Interface=0x4A4AE7B0, Call Info(
       Calling Number=7018,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
       Called Number=(TON=Unknown, NPI=Unknown),
       Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE,
       Incoming Dial-peer=20009, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
    GMIT-VOICEROUT, Calling IE Present=TRUE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
    *Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
       In: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
    *Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
       Out: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
    *Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    *Dec 27 14:10:55.326: :cc_get_feature_vsa malloc success
    *Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    *Dec 27 14:10:55.326:  cc_get_feature_vsa count is 1
    *Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    *Dec 27 14:10:55.326: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234808,feature_id:151
    *Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
       Set Up Event Sent;
       Call Info(Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
       Called Number=(TON=Unknown, NPI=Unknown))
    *Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
       Event=0x49A103B8
    *Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/ccCallSetContext:
       Context=0x4C5A319C
    *Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
       >>>>CCAPI handed cid 12898 with tag 20009 to app "_ManagedAppProcess_Default"
    *Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccCallSetupAck:
       Call Id=12898
    *Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_api_set_transfer_info:
       Transfer Number=, Transfer Reason=0x0
    *Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccGenerateToneInfo:
       Stop Tone On Digit=TRUE, Tone=Dial Tone,
       Tone Direction=Network, Params=0x0, Call Id=12898
    *Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
       Initial Digit Timeout=-1000(ms), Inter Digit Timeout=-1000(ms)
    *Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
       Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=10000(ms))
    *Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
       (callID=0x3262, digit_event=0x1, enable=TRUE, consume=FALSE)
    *Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
       Enabled=TRUE, Call Id=12898
    *Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
       (vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
    *Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
       Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
    *Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
       Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
    *Dec 27 14:10:56.650: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=9, DigitBeginFlags=0x0,
       Rtp Timestamp=0x9D41D0, Rtp Expiration=0x0
    *Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=9, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:56.970: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=1, DigitBeginFlags=0x0,
       Rtp Timestamp=0x9DBED0, Rtp Expiration=0x0
    *Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=1, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:57.290: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
       Rtp Timestamp=0x9E3BD0, Rtp Expiration=0x0
    *Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=2, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:57.610: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=0, DigitBeginFlags=0x0,
       Rtp Timestamp=0x9EB8D0, Rtp Expiration=0x0
    *Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=0, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:57.890: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
       Rtp Timestamp=0x9F35D0, Rtp Expiration=0x0
    *Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=7, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
       Rtp Timestamp=0x9FB2D0, Rtp Expiration=0x0
    *Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=2, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
       Rtp Timestamp=0xA02FD0, Rtp Expiration=0x0
    *Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=2, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
       Rtp Timestamp=0xA0ACD0, Rtp Expiration=0x0
    *Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=7, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
       Rtp Timestamp=0xA129D0, Rtp Expiration=0x0
    *Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=7, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=6, DigitBeginFlags=0x0,
       Rtp Timestamp=0xA1A6D0, Rtp Expiration=0x0
    *Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
       Source Call Id=12898, Digit=6, Duration=100,
       Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
    *Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
       Call Entry(Handoff Depth=0)
    *Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
       (callID=0x3262, digit_event=0x0, enable=FALSE, consume=FALSE)
    *Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
       Enabled=TRUE, Call Id=12898
    *Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
       (vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
    *Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
       Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
    *Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
       Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
    *Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallProceeding:
       Progress Indication=NULL(0)
    *Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
       Destination=, Calling IE Present=TRUE, Mode=0,
       Outgoing Dial-peer=702, Params=0x4C5A0BDC, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
    *Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
       In: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
    *Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
       Out: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
    *Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
       Destination Pattern=91[2-9]......., Called Number=120722776, Digit Strip=FALSE
    *Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
       Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
       Called Number=120722776(TON=Unknown, NPI=Unknown),
       Redirect Number=, Display Info=Server Room
       Account Number=, Final Destination Flag=FALSE,
       Guid=8912F77B-6E37-11E3-8243-90AE21CDDC5B, Outgoing Dial-peer=702
    *Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
       ccCallSetupRequest:
       cisco-username=
       ----- ccCallInfo IE subfields -----
       cisco-ani=20722672628
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=0
       dest=120722776
       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
    *Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
       Interface=0x48C27BD0, Interface Type=3, Destination=, Mode=0x0,
       Call Params(Calling Number=20722672628,(Calling Name=Server Room)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
       Called Number=120722776(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
       Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=702, Call Count On=FALSE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
    *Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    *Dec 27 14:10:59.034: :cc_get_feature_vsa malloc success
    *Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    *Dec 27 14:10:59.034:  cc_get_feature_vsa count is 2
    *Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    *Dec 27 14:10:59.034: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234584,feature_id:152
    *Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
       SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
    *Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccCallSetContext:
       Context=0x4C5A0B8C
    *Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccSaveDialpeerTag:
       Outgoing Dial-peer=702
    *Dec 27 14:10:59.038: //12899/8912F77B8243/CCAPI/cc_api_call_proceeding:
       Interface=0x48C27BD0, Progress Indication=NULL(0)
    *Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
       Cause Value=57, Interface=0x48C27BD0, Call Id=12899
    *Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
       Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
    *Dec 27 14:10:59.270: //12898/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:
       release reserved xcoding resource.
    *Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallSetAAA_Accounting:
       Accounting=0, Call Id=12899
    *Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
       Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=57)
    *Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
       Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
    *Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
       Disposition=0, Interface=0x48C27BD0, Tag=0x0, Call Id=12899,
       Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
    *Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
       Call Disconnect Event Sent
    *Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    *Dec 27 14:10:59.274: :cc_free_feature_vsa freeing 4C6D58D0
    *Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    *Dec 27 14:10:59.274:  vsacount in free is 1
    *Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
       Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
    *Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
       Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
    *Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/cc_api_get_transfer_info:
       Transfer Number Is Null
    *Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
       Disposition=0, Interface=0x4A4AE7B0, Tag=0x0, Call Id=12898,
       Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
    *Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
       Call Disconnect Event Sent
    *Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    *Dec 27 14:11:02.250: :cc_free_feature_vsa freeing 4C6D59B0
    *Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    *Dec 27 14:11:02.250:  vsacount in free is 0ER01#

  • Cisco Jabber for Windows in Extend and Connect mode and making outbound calls

    Hi guys,
    I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
    However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
    After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
    Has anyone got this working or can provide some guidance?
    Thanks.

    Hi guys,
    I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
    However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
    After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
    Has anyone got this working or can provide some guidance?
    Thanks.

Maybe you are looking for

  • Setting a variable values only in the first time

    Hi all, I have a problem regadring PL-SQL coding..actually its something related to programming logics. I'm calling a method (modify) from client application and which calls several other methods. (check_update and update) method update updates the d

  • IPhone 2.0 software update: in Calendars, how to you customize colors?

    With the update (2.0) released on Friday July, 11, 2008; for existing iPhone customers, is there any way to customize the calendar colors? Even if I can just change the color to one color (because I hate red) is there any way to do this? Thanks

  • I cant edit my events

    This is new to me, I have used ical before, and all of a sudden, I can type an event on the calendar, but cannot edit it. So, I can't put it the time of the event etc. I have checked all my settings, and can't see that I have changed anything, anyone

  • I have a question about the 7390...

    My phone's software has crashed, is there a way to reflash or reinstall the firmware without the Updater program? (My Upgrader says no new version available. Other than this it's a great phone! My mobile is 3 (UK) branded incase that matters. Thanks

  • "There was an error downloading"? - addons bug

    Hello Firefox team. Thank you for reading this. I'm currently using Ubuntu 14.04 LTS, and using Firefox as the internet browser. Few days ago, I downloaded and installed an update for it. Since them, I found a bug when I try to update my addons. Ever