CFA never releases caller from Hold

A subscriber enters an off network number for their Call Forward All (such as their home or cell phone). A caller dials the main number, then enters the user's extension.
Unity 4.2(1) replies with "please wait..." and the caller is placed on hold, listening to the futuristic music Cisco provides.
The user's phone rings where it has been forwarded to (their cell phone).
When the user answers, they have dead air. If they wait long enough, they (the subscriber) will hear their own voice mail message.
The calling party stays on hold until they perform a Kevorkian disconnect (disconnect themself).
We are not using supervised calling. I verified the Unity 5.0.4a 301 timer is larger than the ring timer.
I watched the calls in Unity, and the field called "transfer" never has an entry into it.
"Call #","Time","Origin","Reason","Trunk ID","Port ID","Dialed Number","Calling Number","Forwarding Station"
"1","15:19:42","Internal","Direct","0","15","2900","817176202542"," "
"1","15:20:29","Duration = 47 sec"," "," "," "," "," "," "
The caller stayed on hold the entire time after it said "Please wait..."
Any ideas?

I also have this:
001328227| 2006/11/09 16:25:45.206| 001| SdlSig | SsCallInfoRes | getting_secondary_call_info | Transferring(1,100,57,568) | TransferManager(1,100,58,1) | (1,100,59,1).1643273-(vmp_mech01-VI6:10.50.1.32)| [R:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP: 0 DBP: 0]Type=16777220 ssKey=568 sideASsNode=1 sideASs=18903780 sideADsla=(0,0,536949258,0) AHold=0 sideAPSS=6d84ae8f-4126-84c1-65bc-e164e4effd6d sideACMDevType=4 sideAEncodingType=0 sideAIsPreferAltScript=0 sideBNode=1 sideBSs=18903783 sideBDsla=(0,18903783,570503690,5060) BHold=0 sideBPSS=14eb28c6-a3ed-8c24-7cd5-a9bcfd315e7f sideBCMDevType=8 sideBEncodingType=0 sideBIsPreferAltScript=0 cgPart=pt_mech_voicemail cgPat=2906 cgTags= cgValues= pretransCgp:tn=0npi=0nd=2906pi=0si1 cgpn:tn=0npi=1nd=7177961936pi=1si3 unModCgpn:tn=0npi=1nd=7177961936pi=0si3 untransformedCgpn:tn=0npi=1nd=2906pi=1si0 cgNameInfo=locale: 1 Name: VoiceMail UnicodeName: VoiceMail pi: 1 cgLocName=locale: 1 Name: UnicodeName: pi: 0 cgpnMailbox= cgpnVMPN= cgpnVMPCss= cgCause=0 cgDevName=vmp_mech01-VI6 cgDevCepn=3ce733d4-20bf-0e06-b2ec-8d9a0b0e3216 cgDevLocale=1 dialedNum:tn=0npi=0nd=82838829pi=0si1 cdPart=pt_mech_longdist cdPat=8.@ cdTags= cdValues= pretransCdpn:tn=0npi=0nd=82838829pi=0si1 cdpn:tn=0npi=0nd=2838829pi=1si1 unModCdpn:tn=2npi=1nd=82838829pi=0si1 cdNameInfo=locale: 1 Name: UnicodeName: pi: 1 cdLocName=locale: 1 Name: UnicodeName: pi: 0 cdpnVM= cdpnVMPN=2900 cdpnVMPCss=pt_mech_911:pt_mech_101:pt_mech_900:pt_mech_976:pt_mech_operator:pt_mech_voicemail:pt_mech_office cdCause=0 cdDevName=trk_sip2 cdDevCepn= cdDevLocale=1 oPart=pt_mech_office oPat=2403 oTags= oValues= oDialedNum:tn=3npi=1nd=2403pi=0si1 oCdpn=tn=0npi=0nd=2403pi=0si1 oCdpnVM= oCdpnVMPN=2900 oCdpnVMPCss=pt_mech_911:pt_mech_101:pt_mech_900:pt_mech_976:pt_mech_operator:pt_mech_voicemail:pt_mech_office oCdpnRFR=15 oCdpnCause=0 LRPartition=pt_mech_office LRPattern=2403 LRWithTags= LRWithValues= LRNum=tn=0npi=0nd=2403pi=1si1 rnName=locale: 1 Name: UnicodeName: pi: 0 LRNumVMbox= LRNumberVMPN=2900 LRVMPCss=pt_mech_911:pt_mech_101:pt_mech_900:pt_mech_976:pt_mech_operator:pt_mech_voicemail:pt_mech_office LRRFR=15 LRCause=0 callState=5 clientCodeRequired=0 authorizationCodeRequired=0 authorizationLevelRequired=0 GCI=(1,2119530) AQsigStatus=0 BQsigStatus=0 sideAUserState=2 sideBUserState=2 sideAUnattendedPort=0 sideBUnattendedPort=0sideAisOffnetDev=0sideBisOffnetDev=1 ssOverlapDigits= callType=0 sideAPathRepSupport=76 sideBPathRepSupport=0 ssControllingCcPid=(1,157,3759)

Similar Messages

  • Releasing messages from HOLD

    Hi all,
    # ./imsimta version
    Sun Java(tm) System Messaging Server 7.3-11.01 64bit (built Sep 1 2009)
    * I tried to release a msg which was set on HOLD using a single cmd (./imsimta qm).
    * After setting a user on hold in a test setup I send a msg to that user and observations were as below.
    # ./imsimta qm
    qm.maint> dir -held
    Wed, 23 July 2010 15:29:41 +0530 (IST)
    Data gathered from the queue directory tree
    Channel: hold Size Queued since
    1 ZZh051t0wSiy2.HELD 2 23 July 2010 15:03:04
    Total size: 2
    Grand total size: 2
    * I tried the cmd below to release the msg from HOLD and get an error msg like below
    # ./imsimta qm release -channel=hold -all
    %QM-W-NOSELECTIONS, no selections have been made with a "directory" command
    * But when try in this way the msg was succesfully delivered.
    # ./imsimta qm
    qm.maint> dir -held
    qm.maint> release 1
    %QM-I-RELEASED, released the message file /JES/opt/sun/comms/messaging64/data/queue/hold/001/ZZh051t0wSiy2.HELD
    qm.maint> dir -held
    %QM-I-NOFILES, no mail files found
    # ./imsimta cnbuild
    # ./imsimta restart
    * Now msg is delivered to the user inbox.
    * How can I manage to do the same with a single imsimta qm cmd successfully?
    * I need to include this single cmd in to a script to automate a process of mail migration using imsbackup/restore.
    * Any advice is greatly appreciated....
    Thank you,

    nusagnar wrote:
    * How can I manage to do the same with a single imsimta qm cmd successfully?There are two known issues with the version of Messaging Server you are running and releasing held messages:
    bug #6767783 - "job controller should not wait for cache synchronization to process a released .HELD message"
    bug #6879692 - "Releasing messages from hold channel when the user status is still set to hold causes them to bounce"
    The first bug is fixed in MS7 patch 12 (you are running patch 11) and the second is to be addressed in the upcoming MS7 patch 18 (a.k.a. MS7 update 4).
    I recommend raising a Sun support request to get a copy of MS7 patch 18 when it is available before continuing your testing.
    Regards,
    Shane.

  • Relese the orders from hold

    Hi All
    I have to release orders from hold(particular hold). i think we have the API OE_HOLDS_PUB.RELEASE_HOLDS but how to call this API, what are the parameters we need? if any sample code please post it. i am working on 11.5.10.2 version EBS , O/S windows
    Thanks

    See http://sanjaimisra.blogspot.com/2008/08/releasing-order-holds.html
    Hope this helps,
    Sandeep Gandhi

  • Insert into hold lock and never release.

    I put following insert in a SP:
    Insert into mytab(name, access_date,username, ipaddress )
         Select  'my name', getdate(), 'NA', ipaddr from master..sysprocesses where spid = @@spid   
    and then run pb app to call this SP. This cause a ex lock hold on table mytab and never release even auto commit =true in pb until I stop the app.
    Not sure why. It is because of @@spid or master? how to release lock for this insert?

    While you *can* issue a 'set chained' command you typically have to do this as the first command after closing our the previous transaction (see my example where I issue 'commit tran' before issuing 'set chained off').
    I have no idea why you're trying to enable chained mode at the beginning of your proc ... ?????
    As for wreaking havoc with the application ... this comes down to a basic understanding of transaction management.  Consider the following example:
    1 - application: set chained on, insert/update/delete, call your proc ...
    2 - your proc: commit tran, set chained off, insert/update/delete, set chained on ...
    3 - application: insert/update/delete ...
    4 - application: hits an error and issues a rollback, expecting to rollback this entire example (ie, back to line 1), but ...
    Your proc broke the transaction so ... lines 1/2 have been permanently committed while lines 3/4 have been rolled back ... *probably* not a good thing, eh?!?!?!
    Another example:
    1 - application: set chained on, insert/update/delete, call your proc ...
    2 - your proc: commit tran, set chained off, insert/update/delete, exit without re-enabling chained mode
    3 - application: continues processing with assumption it's still running in chained transaction mode, not realizing that it's actually operating with no transactional wrappers/controls ... again, *probably* not a good thing, eh?!?!?!
    If you're doing all the coding (client side, middleware, backend/stored procs) and you have an intimate knowledge of how your transactions are being managed ... then you can do anything you want.
    Unfortunately, in the real world it typically doesn't work this way.  You have multiple folks writing/maintaining different pieces of the code line (either now or down the road after you've left), so everyone has to know how everyone else is managing their transactions ... chained vs not-chained, nested transactions, DDL (not) in transactions, etc.
    My kludgy example (previous post), assuming you pay attention to the details, will probably address *your* current issue but will likely screw up the transactional model being used by the calling process.
    While this may seem like a minor issue to you (as the stored proc developer) I can tell you that this is a major, recurring issue I've seen over the years at too many clients.  I've seen production support groups that have folks assigned (full time) specifically to tracking down and fixing the 'bad' data issues that arise from these types of broken transactions.
    As for your current issue you need to spend some time understanding the transactional model being used by the calling process, then either:
    a) code your proc to fit within (and without breaking) said transactional model; you likely do *NOT* want your stored proc making any changes to the transactional model used by the calling process (eg, do not rollback/commit to @@trancount=0;  do not change the value of @@tranchained)
    or
    b) work with the person(s) maintaining the calling process to see if you can come up with a transactional model that will support everyone's needs

  • 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.

  • My IPhone 6 with AT&T is dropping 5-10 calls a day. Calling from same location as my 5S which never had dropped calls. Is this a hardware or software issue?

    My IPhone 6 128GB with current software using AT&T is dropping 5-10 calls a day. I am calling from an office with good signal strength and in the same location as i did with my 5S which almost never had dropped calls. Is this a hardware or software issue? Is this a common issue with the new IPhones?

    Hi jackneedshelp, 
    Thanks for visiting Apple Support Communities. 
    If calls are not going through on your iPhone 6, I recommend going through the troubleshooting method found at this link:
    iPhone: Troubleshooting issues making or receiving calls - Apple Support
    These steps can help identify and resolve the symptom that might be causing the connection issue.  
    All the best,
    Jeremy

  • UCCE: Forceful Release an Agent Call from ICM Script, Can I?

    Hi, let me explain the requirement first. Customer wants to make their IVR free of cost but they want to start billing only when the call is landed to skill group/agent. So far I can think to make it possible by triggering their billing server by ODBC gateway through Application Gateway process. But also the customer wants to release that particular call when that pre-paid caller is out of charge. They might trigger one of my application or can modify any particular database field and put the calling# into there and my task would be release that call.
    I have thought an idea to develop a TCL script run into the voice gateways and release the call from there by searching the particular call with calling#, but I do not know TCL scripting or any idea how to develope TCL , can't I release that call from ICM script? Do I have any control on calls from ICM when the call is landed and connected to agent?
    Any help would be hightly appreciated.

    That's a nasty piece of work. Just imagine how jacked off you would be if you are the customer, you have enough in your bank account to get to an agent who is then starting to help you, and in the middle of your conversation you are simply cut off!
    I don't think it's possible - although CVP would be your best shot because of the switch leg.
    But not only that, I don't think it is desirable. If you check the customer's balance before going to an agent, that should be sufficient. Anything else is just terrible customer service.
    Regards,
    Geoff

  • CCA-UC560-call from UC560 phones to PBX phones the call is never established.

    I have a UC560 which is in a location (remote site) and I have a PBX in other location (headquarters)
    The extension of UC560 phones are 3xx y the PBX extensions are 2xx.
    I have a Gateway which is connected to UC560 via trunk SIP and this gateway is connected
    to PBX via H.323
    The UC560 was configured with CCA.
    When I call from PBX phones to UC560 phones the PBX phones ring but when the call is
    answered the call is finished.
    When I call from UC560 phones to PBX phones the call is never established. The PBX phones
    don´t ring.
    Can you help me with the configuration?

    Hi Paolo and Alexander,
    First thanks for your help and I apologize for the delay in response because I could not answer before.
    I made some changes and now the situation is as follows:
    -When I call from PBX phones to UC560 phones the UC560 phones ring and sometimes when the call is answered there are RTP packets (the call is fine) but sometimes when the call is answered there is no voice (there are not RTP packets).
    -When I call from UC560 phones to PBX phones the PBX phones ring but in this case there is never voice (there are not RTP packets)
    I captured traffic with the Wireshark and I saw the following:
    - When the call is fine from PBX Phones to UC560 Phones in the protocol H.245 is opened a logical channel between the PBX and the Gateway H.323-SIP after Terminal Capability Set (Master Slave Determination).
    But when the call is not fine because there are not RTP packets I see in the protocol H.245 that no logical channel is opened after between the PBX and the Gateway H.323-SIP after Terminal Capability Set (Master Slave Determination).
    Is there some parameter about protocol H.245 on Cisco gateway H.323-SIP which I can change?
    If you want I can send the captured traffic.
    Regards

  • API to release suppliers and their sites from hold

    Hyee Forumates,
    Can you please let me know if there is any API to release a supplier and his corresponding sites from HOLD. Any help is appreciated thanks in advance.
    Regards,
    Veronica.M

    If it can be done manually, perhaps it may be able to do it by api. I'm not sure considering data inconsistency issues if associated information exists. What's the purpose? If duplicating a merge may be an option, if not try disabling them.

  • How can i transfer a call from SIP 9971 to PBX system on CME router

    hello everybody,
       I have a critical problem about interaction of transfering feature between CME router and pbx panasonic system in some status. let me explain more detail about this issue..i have a SIP 9971(CP-9971) registered on CME at the one site and a voice gateway that is connect with PBX system through a E1 pri trunk connection at the other site. totally the integration between CME and PBX is ok and there is no problem in two direction, i mean i can call pbx system from cp-9971 and vise versa but when i call from a phone  which is registered on PBX site to SIP 9971 which is registered on cisco CME call is connected,then when i try to transfer that call to another phone at PBX site, the session is open between two panasonic phones but no audio transmited in two direction. in addition every thing works fine about SCCP phones(transfer feature works fine). here is my configuration file. i hope someone could help me because i've searched a lot but no result help help help plz....
    cme router 3845 configuration
    VOIP-3845#show running-config
    Building configuration...
    Current configuration : 12657 bytes
    ! Last configuration change at 11:44:01 UTC Mon Oct 31 2011 by admin
    ! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
    ! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname VOIP-3845
    boot-start-marker
    boot-end-marker
    no aaa new-model
    clock calendar-valid
    dot11 syslog
    ip source-route
    ip cef
    no ipv6 cef
    multilink bundle-name authenticated
    voice-card 0
    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
    sip
      bind control source-interface Loopback10
      bind media source-interface Loopback10
      registrar server
    voice register global
    mode cme
    source-address 192.168.2.1 port 5060
    max-dn 720
    max-pool 262
    load 9971 sip9971.9-1-1SR1.loads
    authenticate register
    authenticate realm cisco.com
    tftp-path flash:
    file text
    create profile sync 0063544528862458
    camera
    video
    voice register dn  1
    number 500
    voice register dn  2
    number 600
    voice register dn  3
    number 700
    name test
    voice register template  1
    softkeys idle  Newcall Redial Cfwdall
    softkeys connected  Confrn Endcall Hold Trnsfer
    voice register pool  1
    id mac B8BE.BF23.5242
    type 9971
    number 1 dn 1
    template 1
    username test password test
    camera
    video
    blf-speed-dial 4 600 label "test"
    voice register pool  2
    id mac B8BE.BF9C.5476
    type 9971
    number 1 dn 2
    template 1
    username bank password bank
    camera
    video
    voice register pool  3
    id mac B8BE.BF9C.51D4
    type 9971
    number 1 dn 3
    template 1
    username test1 password test1
    camera
    video
    voice register pool  4
    id mac B8BE.BF9C.4FA2
    number 1 dn 1
    camera
    video
    crypto pki token default removal timeout 0
    crypto pki trustpoint TP-self-signed-1576175886
    enrollment selfsigned
    subject-name cn=IOS-Self-Signed-Certificate-1576175886
    revocation-check none
    rsakeypair TP-self-signed-1576175886
    crypto pki certificate chain TP-self-signed-1576175886
    certificate self-signed 01
      30820241 308201AA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
      31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
      69666963 6174652D 31353736 31373538 3836301E 170D3131 31303038 30393034
      34365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
      4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 35373631
      37353838 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
      8100D6EC 47BCDC3C 82F43FF3 23522678 2616868D 9910DCD2 E36016B3 D7B40DA7
      53A6E339 4978D451 21F051BE B21F8AD5 86B952DC 1ECCE371 3E094B54 26A41E14
      A3055C06 AE860756 425E5C50 E62B3287 631B1E87 9BAC2E39 2810E120 DA3BF823
      947EA591 81CA5489 1B868239 E835EC7C 0AA7651A 22D6E47F 545EBEF3 A172C9A3
      5A0D0203 010001A3 69306730 0F060355 1D130101 FF040530 030101FF 30140603
      551D1104 0D300B82 09564F49 502D3338 3435301F 0603551D 23041830 1680146C
      934AD072 99DDC600 ECD6F389 8F71E0C2 18EC2E30 1D060355 1D0E0416 04146C93
      4AD07299 DDC600EC D6F3898F 71E0C218 EC2E300D 06092A86 4886F70D 01010405
      00038181 000E82F6 5FBB847C 49226955 6F7DECE7 0B093513 D57C35D5 4CD22FA7
      8144A080 B0D56C8D 86AF8156 0152443A A3FBE59F B1AEFFBC BEB43E09 35757BAD
      4C06FC4A 0F3695E0 B00FBD30 4E8F36CE 7748F39C F9602650 7A1D2D48 DBC31237
      AE3D63CE 593D31F5 62E4916F D20E30E8 30DC55C0 120FBD26 D2768DBC A67DDC34
      5BDB66B1 E3
            quit
    license udi pid CISCO3845-MB sn FOC14421Q1Y
    archive
    log config
      hidekeys
    username admin privilege 15 secret 5 $1$Zf7j$P93opukmmEBIioVpjmHB3.
    redundancy
    interface Loopback10
    ip address 192.168.2.1 255.255.255.0
    interface Tunnel1
    ip address 172.25.10.1 255.255.255.0
    no ip redirects
    ip nhrp map multicast dynamic
    ip nhrp network-id 10
    tunnel source GigabitEthernet0/1.1
    tunnel mode gre multipoint
    tunnel key 100
    interface Tunnel2
    ip address 172.25.11.1 255.255.255.0
    no ip redirects
    ip nhrp map multicast dynamic
    ip nhrp network-id 20
    tunnel source GigabitEthernet0/1.2
    tunnel mode gre multipoint
    interface Tunnel14
    ip address 192.168.13.129 255.255.255.252
    tunnel source GigabitEthernet0/1.1
    tunnel destination 10.2.68.25
    interface Tunnel18
    ip address 192.168.13.137 255.255.255.252
    tunnel source GigabitEthernet0/1.1
    tunnel destination 10.9.160.236
    interface GigabitEthernet0/0
    no ip address
    shutdown
    duplex auto
    speed auto
    media-type rj45
    interface GigabitEthernet0/1
    no ip address
    duplex auto
    speed auto
    media-type rj45
    interface GigabitEthernet0/1.1
    encapsulation dot1Q 10
    ip address 10.9.160.25 255.255.255.0
    interface GigabitEthernet0/1.2
    encapsulation dot1Q 50
    ip address 10.10.9.25 255.255.255.0
    router eigrp 202
    network 172.25.11.0 0.0.0.255
    network 192.168.2.0 0.0.0.15
    redistribute static route-map MYMAP1
    router eigrp 201
    network 172.25.10.0 0.0.0.255
    network 192.168.2.0 0.0.0.15
    redistribute static route-map MYMAP1
    ip forward-protocol nd
    ip http server
    ip http secure-server
    ip http path flash:/gui
    ip route 10.2.68.0 255.255.255.0 10.9.160.1
    ip route 10.10.0.0 255.255.0.0 10.10.9.1
    ip route 10.64.164.30 255.255.255.255 10.9.160.1
    ip route 192.168.14.0 255.255.255.0 192.168.13.130
    ip route 192.168.17.0 255.255.255.0 Tunnel18
    ip access-list standard REDIS1
    permit 192.168.14.0
    permit 192.168.17.0
    route-map MYMAP1 permit 10
    match ip address REDIS1
    snmp-server community test RO
    tftp-server flash:term11.default.loads
    tftp-server flash:dkern9971.100609R2-9-0-3.sebn
    tftp-server flash:kern9971.9-0-3.sebn
    tftp-server flash:rootfs9971.9-0-3.sebn
    tftp-server flash:sboot9971.111909R1-9-0-3.sebn
    tftp-server flash:sip9971.9-0-3.loads
    tftp-server flash:skern9971.022809R2-9-0-3.sebn
    tftp-server flash:sccp11.9-0-2sr1s
    tftp-server flash:SCCP11.9-1-1SR1S.loads
    tftp-server flash:apps11.9-1-1TH1-16.sbn
    tftp-server flash:cnu11.9-1-1TH1-16.sbn
    tftp-server flash:cvm11sccp.9-1-1TH1-16.sbn
    tftp-server flash:dsp11.9-1-1TH1-16.sbn
    tftp-server flash:jar11sccp.9-1-1TH1-16.sbn
    tftp-server flash:term06.default.loads
    tftp-server flash:sip9971.9-1-1SR1.loads
    tftp-server system:cme/sipphone
    tftp-server flash:Desktops/320x212x12/NantucketFlowers.png
    tftp-server flash:Desktops/320x212x12/TN-CampusNight.png
    tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png
    tftp-server flash:Desktops/320x212x12/TN-Fountain.png
    tftp-server flash:Desktops/320x212x12/TN-MorroRock.png
    tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png
    tftp-server flash:Desktops/320x212x12/Fountain.png
    tftp-server flash:Desktops/320x212x12/CiscoLogo.png
    tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png
    tftp-server flash:Desktops/320x212x12/List.xml
    tftp-server flash:Desktops/320x216x16/List.xml
    tftp-server flash:Desktops/320x212x16/List.xml
    tftp-server flash:gui/admin_user.html
    tftp-server flash:gui/admin_user.js
    tftp-server flash:gui/CiscoLogo.gif
    tftp-server flash:gui/Delete.gif
    tftp-server flash:gui/dom.js
    tftp-server flash:gui/downarrow.gif
    tftp-server flash:gui/ephone_admin.html
    tftp-server flash:gui/logohome.gif
    tftp-server flash:gui/normal_user.html
    tftp-server flash:gui/normal_user.js
    tftp-server flash:gui/Plus.gif
    tftp-server flash:gui/sxiconad.gif
    tftp-server flash:gui/Tab.gif
    tftp-server flash:gui/telephony_service.html
    tftp-server flash:gui/uparrow.gif
    tftp-server flash:gui/xml-test.html
    tftp-server flash:gui/xml.template
    tftp-server flash:ringtones/Analog1.raw
    tftp-server flash:ringtones/Analog2.raw
    tftp-server flash:ringtones/AreYouThere.raw
    tftp-server flash:ringtones/AreYouThereF.raw
    tftp-server flash:ringtones/Bass.raw
    tftp-server flash:ringtones/CallBack.raw
    tftp-server flash:ringtones/Chime.raw
    tftp-server flash:ringtones/Classic1.raw
    tftp-server flash:ringtones/Classic2.raw
    tftp-server flash:ringtones/ClockShop.raw
    tftp-server flash:ringtones/DistinctiveRingList.xml
    tftp-server flash:ringtones/Drums1.raw
    tftp-server flash:ringtones/Drums2.raw
    tftp-server flash:ringtones/FilmScore.raw
    tftp-server flash:ringtones/HarpSynth.raw
    tftp-server flash:ringtones/Jamaica.raw
    tftp-server flash:ringtones/KotoEffect.raw
    tftp-server flash:ringtones/MusicBox.raw
    tftp-server flash:ringtones/Piano1.raw
    tftp-server flash:ringtones/Piano2.raw
    tftp-server flash:ringtones/Pop.raw
    tftp-server flash:ringtones/Pulse1.raw
    tftp-server flash:ringtones/Ring1.raw
    tftp-server flash:ringtones/Ring2.raw
    tftp-server flash:ringtones/Ring3.raw
    tftp-server flash:ringtones/Ring4.raw
    tftp-server flash:ringtones/Ring5.raw
    tftp-server flash:ringtones/Ring6.raw
    tftp-server flash:ringtones/Ring7.raw
    tftp-server flash:ringtones/RingList.xml
    tftp-server flash:ringtones/Sax1.raw
    tftp-server flash:ringtones/Sax2.raw
    tftp-server flash:ringtones/Vibe.raw
    tftp-server flash:APPS-1.2.1.SBN
    tftp-server flash:SYS-1.2.1.SBN
    tftp-server flash:GUI-1.2.1.SBN
    tftp-server flash:CP7921G-1.2.1.LOADS
    tftp-server flash:TNUX-1.2.1.SBN
    tftp-server flash:TNUXR-1.2.1.SBN
    tftp-server flash:WLAN-1.2.1.SBN
    tftp-server flash:apps37sccp.1-2-1-0.bin
    tftp-server flash:APPSH-1.3.1.SBN
    tftp-server flash:GUIH-1.3.1.SBN
    tftp-server flash:CP7925G-1.3.1.LOADS
    tftp-server flash:SYSH-1.3.1.SBN
    tftp-server flash:TNUXH-1.3.1.SBN
    tftp-server flash:WLANH-1.3.1.SBN
    tftp-server flash:SCCP11.9-2-1S.loads
    tftp-server flash:Desktops/320x212x12/CampusNight.png
    tftp-server flash:Desktops/320x212x12/CiscoFountain.png
    tftp-server flash:Desktops/320x212x12/MorroRock.png
    tftp-server flash:skern9971.022809R2-9-2-1.sebn
    tftp-server flash:sip9971.9-2-1.loads
    tftp-server flash:sboot9971.031610R1-9-2-1.sebn
    tftp-server flash:rootfs9971.9-2-1.sebn
    tftp-server flash:dkern9971.100609R2-9-2-1.sebn
    tftp-server flash:kern9971.9-2-1.sebn
    tftp-server flash:United_States/g4-tones.xml
    tftp-server flash:English_United_States/gd-sip.jar
    tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
    tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
    tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
    tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
    tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
    control-plane
    mgcp profile default
    dial-peer voice 1 voip
    description connection-trough-PBX
    destination-pattern 0....
    session target ipv4:192.168.13.130
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 100 voip
    description K
    destination-pattern 9T
    session target ipv4:192.168.13.130
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 5 voip
    shutdown
    destination-pattern *3709
    session protocol sipv2
    session target ipv4:192.168.13.130
    session transport tcp
    dtmf-relay h245-alphanumeric
    codec g711ulaw
    no vad
    dial-peer voice 2 pots
    incoming called-number .
    dial-peer voice 10 voip
    gatekeeper
    shutdown
    telephony-service
    em logout 0:0 0:0 0:0
    max-ephones 262
    max-dn 400
    ip source-address 192.168.2.1 port 2000
    load 7911 SCCP11.9-2-1S
    max-conferences 12 gain -6
    web admin system name admin secret 5 $1$IKnn$tyKyuBcGqXFl6nhxCSu.z0
    dn-webedit
    time-webedit
    transfer-system full-consult
    transfer-pattern .T
    create cnf-files version-stamp 7960 Oct 29 2011 12:39:25
    ephone-template  1
    softkeys connected  Confrn Endcall Trnsfer Hold
    keep-conference endcall
    ephone-dn  1  dual-line
    number 200
    label test
    name test
    ephone-dn  2  dual-line
    number 300
    label Sepahbod
    name Sepahbod
    ephone-dn  4  dual-line
    number 666
    ephone-dn  5  dual-line
    number 660
    ephone-dn  6  dual-line
    number 670
    ephone-dn  7  dual-line
    number 770
    ephone-dn  8  dual-line
    number 770
    ephone-dn  9  dual-line
    number 999
    ephone  1
    device-security-mode none
    mac-address 18EF.639F.BCB0
    keep-conference endcall
    button  1:1
    ephone  2
    device-security-mode none
    mac-address 0025.8418.B017
    ephone-template 1
    keep-conference endcall
    button  1:2
    ephone  3
    device-security-mode none
    mac-address F04D.A243.3154
    keep-conference endcall
    button  1:4
    ephone  4
    device-security-mode none
    mac-address 6CF0.496A.69E9
    button  1:4
    ephone  5
    device-security-mode none
    mac-address 0015.E987.345F
    keep-conference endcall
    button  1:5
    ephone  6
    device-security-mode none
    mac-address 0024.1DEA.614A
    keep-conference endcall
    button  1:6
    ephone  9
    device-security-mode none
    mac-address 001D.7D4D.4DCB
    button  1:9
    line con 0
    line aux 0
    line vty 0 4
    login local
    transport input telnet
    scheduler allocate 20000 1000
    end
    and Voice Gateway connected two PBX system configuration
    Current configuration : 3486 bytes
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Voice-GW
    boot-start-marker
    boot-end-marker
    card type e1 0 2
    no aaa new-model
    network-clock-participate wic 2
    dot11 syslog
    ip source-route
    ip cef
    no ipv6 cef
    multilink bundle-name authenticated
    isdn switch-type primary-net5
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    h323
    voice-card 0
    crypto pki token default removal timeout 0
    license udi pid CISCO2811 sn FHK1352F0E9
    username admin privilege 15 secret 5 $1$O6AN$1kvvqiLdIl3/ZTHoyYRy0/
    redundancy
    controller E1 0/2/0
    framing NO-CRC4
    pri-group timeslots 1-31
    controller E1 0/2/1
    interface Tunnel14
    ip address 192.168.13.130 255.255.255.252
    tunnel source FastEthernet0/1
    tunnel destination 10.9.160.25
    interface Tunnel17
    ip address 192.168.13.134 255.255.255.252
    tunnel source FastEthernet0/1
    tunnel destination 10.9.160.25
    interface FastEthernet0/0
    ip address 192.168.14.252 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    ip address 10.2.68.25 255.255.255.0
    duplex auto
    speed auto
    interface Serial0/2/0:15
    no ip address
    encapsulation hdlc
    isdn switch-type primary-net5
    isdn overlap-receiving
    isdn incoming-voice voice
    no cdp enable
    router eigrp 201
    network 172.25.10.0 0.0.0.255
    network 192.168.14.0
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 10.9.160.0 255.255.255.0 10.2.68.1
    ip route 10.128.0.69 255.255.255.255 Tunnel14
    ip route 192.168.2.1 255.255.255.255 192.168.13.129
    ip route 192.168.17.0 255.255.255.0 Tunnel14
    tftp-server flash:SCCP11.9-2-1S.loads
    tftp-server flash:jar11sccp.9-2-1TH1-13.sbn
    tftp-server flash:dsp11.9-2-1TH1-13.sbn
    tftp-server flash:cvm11sccp.9-2-1TH1-13.sbn
    tftp-server flash:cnu11.9-2-1TH1-13.sbn
    tftp-server flash:apps11.9-2-1TH1-13.sbn
    control-plane
    voice-port 0/0/0
    caller-id enable
    voice-port 0/0/1
    voice-port 0/0/2
    supervisory disconnect dualtone mid-call
    dial-type pulse
    disc_pi_off
    output attenuation 1
    echo-cancel coverage 32
    timeouts call-disconnect 5
    timeouts wait-release 1
    timing hookflash-out 50
    timing sup-disconnect 50
    connection plar 600
    caller-id enable
    voice-port 0/0/3
    caller-id enable
    voice-port 0/2/0:15
    mgcp profile default
    dial-peer voice 1 pots
    description connection-to-PBX
    destination-pattern 0....
    direct-inward-dial
    port 0/2/0:15
    forward-digits 4
    dial-peer voice 10 voip
    destination-pattern ...
    session target ipv4:192.168.13.129
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 20 pots
    description FXO-K
    destination-pattern 9T
    progress_ind alert enable 8
    progress_ind progress enable 8
    progress_ind connect enable 8
    direct-inward-dial
    port 0/0/2
    prefix 9
    dial-peer voice 30 pots
    description FXO-K2
    destination-pattern 9T
    direct-inward-dial
    port 0/0/1
    prefix 9
    telephony-service
    max-ephones 20
    max-dn 100
    ip source-address 192.168.14.252 port 2000
    cnf-file location flash:
    load 7911 term11.default.loads
    max-conferences 8 gain -6
    transfer-system full-consult
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1
    number 770
    line con 0
    line aux 0
    line 1/0 1/15
    line vty 0 4
    login local
    transport input telnet
    scheduler allocate 20000 1000
    end

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

  • Never release the temp space

    It seems my oracle database never release temp space.
    The usage of temp space kept at 99.8% ,but there was no alarm in alert.log.
    It is not acceptable to shutdown the machine (7*24).
    It is said that oracle do not use temp space if the memory for sorting can hold processing statement.
    But there is no large work load on the machine and whenever
    select * from v$sort_usage
    no record
    what s wrong?
    how can i release temp space without shutting down the server?
    Sun fire v880 4 cpu at 1050Mhz Memory 8G
    Solaris 8 0202
    Oracle 92

    You do not need to worry about it, unless you are getting "Unable to allocate ... in table TEMP" messages.
    I believe that the space will be released if you bounce the database.
    The 99% you are seeing is like a high water mark, which says that at some point 99% of your TEMP space was used. However no rows in v$sort_usage means that no space is currently being used.
    So, relax!

  • Clicking on PSTN calls only when placing call on hold/taking off hold

    Hey all,
    We have an ongoing issue that we thought was isolated to a call flow going to our UCCX, but after extensive troubleshooting, have found that it is occuring when a PSTN caller is being placed on hold by or being taken off of hold (most noticeable when taking off of hold) either via an IP Phone or another system (Unity Connection when releasing to switch is transferring outside of Unity Connection, UCCX and it's hold/unhold steps). If we mirror this call flow, only calling from an IP Phone to Another and being placed on hold, or including Unity Connection or UCCX in the call flow, this doesn't occur. We've been working with TAC and our ITSP for the past couple weeks but haven't really made much traction. Wireshark traces on the CUBE, so far, only have the audible noise on the ITSP facing interface of the CUBE, which is what I would expect given that it's not heard internally.
    Call flow:
    PSTN Caller -> ITSP via SIP -> CUBE -> SIP -> CUCM -> IP Phone
    CUCM Version 8.5.1.15900-4
    CUBE Version 15.1(3)T2
    MOH and CUBE have a G711 region relationship. Most IP Phones and systems have a G729 relationship, but as a test I put a phone in my G711 test region and the issue still occurs.
    The traces show the call being setup with G729 (preference on my dial-peer), but the stream is all G711 (audio from my IP Phone and MOH). This was verified both in the trace and via the CUBE when doing a "show sip calls <my cell phone number>" that both legs were G711 and the issue was still occuring.
    We regularly restart the IPVMS service as our MOH changes from time to time, but again this is only happening on PSTN calls, so my focus at this time has been the CUBE.
    Has anyone run into this before? Perhaps it's a bug and I need to update my CUBE's firmware?

    Hi,
    Did you enable “Play music on hold”?
    If yes, you can try to disable it and then test again.
    Please also check if there are any error message on FE Server when the issue happen.
    Please also update to the latest version for Lync Server 2013 and have a test.
    Best Regards,
    Eason Huang
    Eason Huang
    TechNet Community Support

  • Avaya calls to CUCM 8.6: Call Park/Hold Fail

    This may be similar to the issue I posted here:  https://supportforums.cisco.com/message/3496110#3496110
    4 digit dialing from an Avaya system to a CUCM 8.6.2 server works fine, the Avaya is set as an h.323 gateway in CUCM.
    Once the call is active if the Cisco phone places the Avaya caller on hold they hear no hold music and the call stays active until the Cisco phone tries to retrieve it.  Then it's a fast busy on both end.
    The same thing happens with calls placed on park.
    Any suggestions?

    Hi
    Are you using G711 or G729 for these calls? If using G729 the CUCM software MTP cannot be used as it only supports g711, you would have to use a gateway-based MTP.
    Try setting the calls to G711 to see if it works as a test.
    Aaron

  • Credit release transfer from Delivery order to sale order

    Dear Master's,
    Our setting for credit release from delivery order but now change requirement from our organization, they like this check from sale order and i create setting it in test server but there is a problem with me, every thing is ok, system also block sale order for credit release but the value that is, net value. our requirement is different, we need gross value as we already working in previous with delivery order. anybody can solve my problem, i try to find it and feel there is no problem in setting but problem still in system.
    Regards
    Khan

    I already "A" which is credit amount adjust in Gross value after tax and all other discount values but i am confuse why system release value from before tax and trader offers
    Muhammad Arafkhan 
    Please use correct English. it is very hard to understand your requirement the above is one of the  small examples of your post. Am also not perfect.However, will try to use correct English to my level.
    Please never use any SMS language and try to avoid the maximum  grammatical errors.
    Coming into your requirement as per my understanding, previously you were using the credit control at Delivery level, now the same setting is changed to order level is that correct?
    Now kindly explain your requirement once again what is the issue after this change in OVA8? I did not understand completely.
    thanks,
    Srinu.

  • How can I block 'no caller id' calls from contacting me?

    Over the last couple of days someone has got a hold of my number, and has been calling me with a private number. I want to know if there is a way I can block all calls on my phone that come from a private 'no caller ID' number, and only allow ones that have a caller ID through. I have tried contacting my service provider Telstra which they said the most they can do is put me on the do not call register (which doesn't help my situation at all, as the person isn't a telemarketer). Does anyone have any ideas about how to fix my problem?
    Thanks.
    (I have an iPhone 5 iOS 7.1.1 and my service provider is Telstra Australia)

    There is no Third Party App you can load onto an iPhone that will be allowed to take over your iPhone's Phone without jailbreaking your device (which cannot be discussed here nor would I ever recommend doing it).
    One feature you can use is the Do Not Disturb feature.  You can set it to allow calls from your Favorites then add all of your people to your favorites, but that is not quite what you are asking for.

Maybe you are looking for