CUCM 8.6.2 Call Recording....

Hi
I wants to records all inbound and outbound call to perticular IP Phone. Is it possible by monitoring the switch port of that phone ?
Please let me know if there are any thired party softwares which can perform this task.
Regards
Shashi

Shashi,
     When you configured your PC's port as a monitor port it would no longer accept traffic and you would loose connectivity. Monitor ports are used to replicate traffic from other ports. I would suggest if you really need to keep connectivity add a second NIC to your PC.
     I would also ammend the command your using to monitor traffic in 'Both' directions.
          monitor session 1 source interface fa0/2 both
     You could SPAN the whole voice VLAN, it really depends on what your trying to achieve, if you can give us more background maybe we can help further.
     Have a look at this document for a guide to SPAN.
          http://www.cisco.com/en/US/products/hw/switches/ps708/products_tech_note09186a008015c612.shtml
Hope this helps,
Craig
(PLEASE RATE HELPFUL POSTS)

Similar Messages

  • Audio & Video Call Recording Solution for CUCM

    Hi,
    I am looking for a audio and video call recording solution for CUCM. The requirement is the recording system shall record all audio & video calls including incoming, outgoing, and station-to-station calls. One cirtical requirement is: The users on the CUCM system shall be able to record video calls for video equipped stations with a push of a button on their Cisco IP Desk set or their Cisco soft phone.
    Does anyone know the product or third party vendor who supports audio and video recording for CUCM?
    -Thanks
    Vaijanath

    Currently Cisco doesn't support video recording with a push of a button on their Cisco IP Desk set or their Cisco soft phone. Also, I have checked with the third party vendors and they only support the video recording using Passive Call recording (port mirroring).
    http://www.verba.com/cisco-call-recording-solutions

  • CUCM 6.1.5 and CUCM 9.1 Call Recording CODEC Issue

    I have a call recording issue where if the CODEC agreed between an MGCP gateway and the IP phone is G711 call recording fails as it looks for an MTP (there are subs running XCODE in the MRGs for the devices in this scenario).  The traces seem to suggest that there is mismatch between the agreed DTMF.
    When the call between the phone and the gateway is G729 then call recording works fine and doesn't request MT resource.
    SIP traces show that the recorder (Verint Impact 360) responds with a SIP OK which contains options for PCMA, PCMU, and G729.
    Call manager traces show that as soon as the INVITE is sent to the call recorder from the CUCM the request for MTP resource begins (i.e. before any media options are presented and even before the 100 TRYING is sent back)
    The region setting between the phone and the gateways, in the failure scenario, is set to G711 and link loss type is low loss.  On CUCM 9.1 Ive set a CODEC preference list which I have tried g711 as the preference and then also G729.
    The region setting between the phone and the gateway regions towards the region the call recording SIP trunk is in allows max. G711.
    The question is, why when the call is set up using G729 there is no requirement for an MTP on the call recorded leg whereas when the call is setup with G711 then the CUCM looks for MTP?

    To answer your question we need to look at the traces  , please attach both the traces here.
    Thanks
    Manish

  • CUCM 9.1 Call Recording for analogue devices

    Hi,
    We are designing a CUCM 9.1 solution where Active Call Recording for analogue devices feature is required.
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/9_0_1/ccmfeat/CUCM_BK_CEF0C471_00_cucm-features-and-services-guide_chapter_0100110.html
    According the feautre list support it seems that is supported by the old ata 186 but not by the ata 187.
    Then, how could we achieve this on a new deployment? Has anyone had the chance to deal with this?
    Many thanks in advance. Thank you very much,
    Francesc

    This is a weak spot in the native Cisco solution portfolio. I'll try to give one option but hopefully others will chime in with something more clever.
    The MediaSense product can record trunk-side from Cisco CUBE which allows for device-agnostic recording; however, that only works if you're not concerned about internal calls to/from the analog phones (i.e. just to/from the PSTN). Getting the last part seems to require old fashioned SPAN recording to a capture server of a third party. This can get dicey though because most of the third-party recording vendors manually decipher the SCCP packets to figure out the calling/called party information. Be sure to ask them whether SCCP- or MGCP-controlled gateways are supported.
    Please remember to rate helpful responses and identify helpful or correct answers.

  • On demand call recording on QM server with CUCM

    Hi ,
    I have a customer who wants to enable on-demand call recording on QM server .
    Is there a plugin that needs to be installed on Cisco side to make it happen .
    QM Server 9.0.1.57
    Calabrio Call Recording: 8.8.1.57
    Calabrio Quality Management: 8.8.1.57
    Call Manager 9.1.1
    UCCX 9.0.2
    or any infomation regarding that would be helpful
    Thank you

    This is something I have not done before, but I recall seeing the feature for IP Phones.  It was an IP Phone Service which showed a menu for starting and stopping recordings for workers without a PC.
    On the QM 9.0 Data sheet, the feature is mentioned in the following bullet point:
    - The ability to control recording for manual start,  pause, and add metadata through Cisco Agent Desktop or Calabrio recording control browser or IP phone applications
    Source: http://www.cisco.com/en/US/prod/collateral/voicesw/custcosw/ps5693/ps8293/data_sheet_c78-710576.html#wp9000180
    I'm looking for the documentation to configure this feature.  Unfortunately, Calabrio has their own documentation and support site @ portal.calabrio.com which only Cisco partners can access (something I am not).  The answer may reside there.  I'll post back if I find the IP Phone Service URL to use.
    Anthony Holloway
    Please use the star ratings to help drive great content to the top of searches.

  • Call Recording/Monitoring in UCCX7

    Hello, we are getting ready to upgrade to CUCM 7.1 and UCCX 7.0 and I have a question.  Does call recording and monitoring work the same in this version as it does in version 4.1.  We are in a Citrix environment and currently use spanning to get the voice traffic.
    From what I've read, it appears to work the same in 7, but I just wanted to double check.
    Thanks!

    Hi,
    Yes, you can use Rspan Monitoring in the UCCX 7.0 it will work and configured in the same way as it is in UCCX 4.0
    Regards,
    Tere.
    If you find this post helpful, please rate!

  • Obtaining call records from CDR

    I am running CCM 6.1
    We need to pull our call records for a certain ext. THe problem i am having is in CDR is I go to CDR/search/User extensions and search.
    It will only show 100 records and trunkcate the other. Now I need to pull 3 months worth and it would be to time consuming to do each day and copy it to a report.
    I also attempted to goto device reports/gateway/detail and will puill ever 3 days the entire logs. This will put it in the PDF and allow me to search for that ext but its will not copy out of the PDF properly and I need to only show the ext I want.
    Does anyone know a way I can get a detailed call report for only 1 ext that will not truncate the results? or another way I can access this.
    In our older version of call manager (4.1) i was able to load sql and do a search on the DB and just copy the results out.

    Since this old post is being revived, I should mention that in recent versions (too lazy to research exact) of CUCM, the syntax has changed slighty, and the above given command no longer works.
    The new syntax is:
    run sql car select * from tbl_billing_data where originalcalledpartynumber='911'
    EDIT: In our very own forum member Will's blog, he states it was a change from 7x to 8x:
    http://www.ucguerrilla.com/2012/03/cucm-sql-queries-installment-5.html
    Anthony Holloway
    Please use the star ratings to help drive great content to the top of searches.

  • Cisco Unified WFO - Call Recording and Quality Management with Extension Mobility agents

    Hi All,
    We're considering Cisco Unified WFO - Call Recording and Quality Management for a customer running UCCX 8.0, agents on multiple WAN sites, all agents using extension mobility.
    The documentation I've been able to find describes three different recording methods:
    Using Desktop Recording service (Endpoint) to record from an agent’s desktop.
    Server Recording - Uses SPAN (not so good for remote sites)
    Network Recording - Uses CUCM recording service / SIP trunk / phone's built in bridge.
    Network recording or Desktop recording should be suitable for the customer but it seems that Extension mobility is not supported.  Extension Mobility is not mentioned in the 8.5 installation guide, it is mentioned as ‘not supported’ in the 8.0 guide as follows:
    'Server Recording and Network Recording have the following limitations:
    • Extension mobility is not supported.'
    Neither version's documentation specifically mention extension mobility in relation to the desktop recording method, though I realise this is a similar approach to the 'server recording' method.
    So the question I have is:  Is extension mobility supported in any way on version 8.0, or version 8.5 for recording?  And if so which recording method(s) are supported?
    Thanks,
    Jonathan

    Hi,
    I had more luck asking questions over at the Calabrio forum - they make the software and Cisco re-brand a version of it - there is some good info on their portal (http://portal.calabrio.com), you have to register but it's fairly painless.  The answer I got was:
    "QM Desktop recording has always supported extention mobility as it determines the recorded user by the desktop user's login. Extention mobility was not supported for Server and Network recording until the Calabrio QM 8.6.2 release in April 2011 and will be added to Cisco QM starting with QM 8.5.2 in June 2011"
    Regards,
    Jonathan

  • How to assign Call Recording Announcement in Call Manager?

    Hi,
    Customer wants to give an announcement instead of call recording tone. They currently using built-in-bridge recording solution. 
    These scenarios has to apply on built-bridge-recording enabled devices.
    1-7945 > cm > router > pri -----telco-----outside-phone
    At this point outsidephone has to hear call recording announcement before the call connected.
    2-Internal calls has not to be hear any announcement.
    3-outsidephone----telco---pri>router > cm > 7945
    at this scenario outside phone should have hear a call recording announcement before the call proceed.
    Is there any Idea ? 
    Regards,
    Baris.

    Hi Baris,
    If you want this for specific number (hunt group/team) and you have CUCM 9.x or later then you can use 'Native call queuing' feature. And if you are using other than this CM then you should have Unity (CUE/CUC) or UCCX.
    Suresh

  • Migrate Avaya ACD, Convergys IVR and Verint Call Recording/QM and WFM to UCCX

    Current setup today is with an Avaya ACD, Convergys IVR and Verint Call Recording/QM and WFM
    The team is considering the idea of replacing the AVAYA ACD with a Cisco UCCX system – but retain the Convergys IVR and Verint CR/QM/WFM solution.
    Today the Convergys IVR front ends all calls into the team and allows the caller to access all kinds of information from the billing systems etc, via database dips.  Once the Convergys IVR has that information, we’d like to deliver that information to a UCCX agent should the caller opt out of the IVR menu’s and wish to speak to an agent.  So I am wondering how this would be accomplished with Cisco UCCX.
    Today, with the AVAYA ACD, the Convergys IVR writes the gathered information to a database on the AVAYA system so when the agent receives the call, that information is screen-popped to the AVAYA agent. 
    What is your take on this or better yet recommendations
    Thanks

    Grace,
    Right now, I think the biggest question you will need answered is how you will create resources/agents without a CUCM cluster.
    I run a cluster where I have CUCM 8.6 and UCCX 8.5 and I configure all of the resource/agent IPCC extensions on CUCM and then assign the resources/agents the skills or resource group required for my given script/CSQ/application.  If you can't create resources/agents without CUCM, I would stick with your current solution.
    G'luck,
    Brendan

  • Call recorder for iphone 5s

    i am having a iphone 5s which i am looking for a call recording app for voice calls..

    There are many call recording apps (for example, we are developers of such apps ). However, as KiltedTim says, such apps are not tied into the core dialing feature of iPhone but are implemented through one of the two ways:
    1 - VoIP based recording: the app will use your internet connection to connect your phone to the app's server, which in turn will connect you to the recipient phone. It's convenient to use but you must use the app's dialer to make the phone call so only outgoing calls are supported.
    2 - Conference call (3-way call) recorders: the idea is making a conference call with 3 'participants' - you, the person you want to record, and another phone number that actually makes the recording. This phone number is provided and managed by the app. This method supports recording of both incoming and outgoing calls but it's a bit uncomfortable to make the merging of the call with all the participants.
    Regarding the legal issue, each country/state has its own set of rules for phone recording. In some countries a single participant consent is enough, some countries require both participants, some have different rules for recordings that are shared vs private recordings. Our apps include a feature to play 'beep' sound every minute, to inform the other party that the call is recorded. You can use it or disable it as required by the law in your country/state.
    My company have two apps -  'Call Recorder - IntCall' is based on VoIP and CallRec Lite - Record Phone Calls is based on conference calls. You can find a comparison of both apps here - although it's a comparison of our apps most of the sections are relevant for all apps that are based on those two methods so you can learn a bit more about the pros and cons of each method.
    I may receive some form of compensation, financial or otherwise, from my recommendation or link.
    <Edited by Host>

  • Since you don't allow emails any more – I suspect because of the numerous complaints with your service and the way you treat people that you don't want documented, I am calling and I want this call recorded for future reference. I have been a long time fa

    Since you don’t allow emails any more – I suspect because of the numerous complaints with your service and the way you treat people that you don’t want documented, I am calling and I want this call recorded for future reference.
    I have been a long time faithful customer of vzw and although the past year I have been late on payments many times and really couldn’t afford your exorbitant prices for services lots of other companies offer sometimes three times cheaper than what you charge, I have hung in there trying my best to meet my obligations.
    This month has been no exception. You don’t know the background; the whole story of people’s lives. I know you could care less because all you care about is the profit-the money that comes in.
    I was told when I agreed to pay my bill on the third per the recorded message that I had 14 days to pay…you cut me off anyway. The phones are not the tissue; your suspending my service means I cannot work. I may lose my job…how do you justify that? In any case? The least you could do would be to keep 4986 on and cut the phones off. But no. You refuse to compromise and meet the basic needs of your customer. What does that say about your company? I tried to call back on three separate occasions to tell you I couldn’t pay because of unexpected expenses but couldn’t get out of the automated system…sadly couldn’t get to a real person which also speaks volumes to me.
    All this tells me this is a company I don’t wish to be affiliated with any more. As soon as I can, I will discontinue service with you…I know you could care less. I will honor the remaining portion of the contract but that’s it. You don’t deserve my business. I am a good, hardworking person who, at the sacrifice of myself and my needs, always pays her bills…albeit late at times. I realize others tell you stories and lies to justify themselves. That’s not me. If you knew what I had been through the last 7 yrs you would marvel that I am  still on my feet…don’t judge too quickly. You could be wrong…and in my eyes you are by doing this to me.
    God will see us through this extremely scary time of that I have no doubt. No thanks to your company and lack of understanding and mercy. I am doing the best I can. Sadly you are not.
    See I have choices. MANY choices of providers for services you offer. I don’t have to be treated like this. I don’t have to succumb to your coldness and callousness. I intend to choose better (and cheaper). If your company doesn’t get the “people factor” back you will be sorry.

    Problem here is you admit you cannot afford the service.
    And you want to blame Verizon for losing a job because you have no cell phone.
    If your job depends on that phone I would pay it on time every time if you need a job to pay your bill.
    No other service is going to treat you any different. And if you cannot afford Verizon's monthly invoice how are you going to afford new devices, activation fees, possible security deposits on any other cellular carrier? You can't.
    Also if you made an arraignment to pay and then decide you cannot do so, why should Verizon extend you service or credit, or why is it you want to use the service and data and not pay for it as agreed.
    Get a prepay phone. Its evident the cost is too high for you to afford on post pay.
    Good Luck

  • Looking for a Telephone call recorder app

    I have tried a few  telephone call recorder apps on my droid X and they just don't seem to work. They are great when the Doctor calls with test results when you are driving.

    There are lots of different recording applications on the Android Market Place.  
    You can go to the Android Market Place on the device,  then go to search to find these applications.
    Here are two of the higher rated call recording applications:
    Phone Recorder
    Android Recorder 
    Android Market Place Applications are not  provided by Verizon Wireless and direct support for applications is not offered.  

  • Why there is no free call recording like android?

    i live in Egypt. Egypt support blacklist and call recording. When i open the appstore i can't find any free call recording, while on android i found many of them.

    Are there any paid call recording apps in the Egypt iTunes app store?
    If there are paid apps for this and no free apps, an app being paid or free is up to the app developer.

  • CUCM 8.6 Dropped call transfers involving SIP phones

    Hi All,
    I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone.  This scenario works:
    Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
    These scenarios do not work:
    SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
    SIP phone calls Cisco phone, which transfers the original call to another SIP phone
    I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem.  I am just unable to understand what the trace means:
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
    10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
    I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
    G711U
    G711A
    G722
    ILBC
    GSM
    and our SIP software is  also set to accept the first codec offered by the remote side.  It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
    I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492).  The trace snippet shown above is from this log.
    I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it.  I can try to provide more info about our CUCM configuration if needed.
    Thanks in advance!

    Leslie, so here is what I found from the traces....
    To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
    SCCP
    When the transfer key is pressed
    1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
    NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
    2.CUCM tells the held party to listen MOH from MOH server
    3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
    4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
    5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
    6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
    7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
    For SIP signalling. when the first transfer key is pressed
    1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
    2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
    NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
    3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
    4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
    5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
    6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
    7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
    Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
    lets look at succesful SCCP-----SCCP-----SIP without MTP
    Point 4 above
    ++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
    (0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
    Point 5 above
    ++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
    (0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    Point 6 Above
    +++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
    (0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
    (0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
    Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
    +++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885626,NET]
    INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 0.0.0.0
    m=audio 24560 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
    a=ptime:20
    a=inactive-----------------------------------------------------Inactive
    Still part of Point 6 for SIP signalling
    ++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
    //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
    [885628,NET]
    SIP/2.0 200 OK
    v=0
    o=- 18077 11099 IN IP4 10.10.10.104
    s=yasdjip
    c=IN IP4 10.10.10.104
    t=0 0
    a=ptime:20
    a=recvonly-------------------------------------a=recvonly
    Finally Point 7 above..
    +++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885630,NET]
    INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
    +++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
    /SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
    [885634,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
    Contact:
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    Call-ID: [email protected]
    v=0
    o=- 18077 11099 IN IP4 10.10.10.104
    s=yasdjip
    c=IN IP4 10.10.10.104
    t=0 0
    m=audio 16574 RTP/AVP 9 101
    a=rtpmap:101 TELEPHONE-EVENT/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    +Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
    (0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    (0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
    remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    +++++++++++=Next Phone sends its ACK+++++++++++++++
    (0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
    +++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885635,NET]
    ACK sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
    Date: Tue, 19 Feb 2013 21:44:45 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 237
    v=0
    o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.137
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 20352 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Now at this point all is well...and the call is connected....
    Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
    From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
    [881160,NET]
    ACK sip:[email protected]:53361;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Date: Tue, 19 Feb 2013 17:38:50 GMT
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    Max-Forwards: 70
    CSeq: 102 ACK
    o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
    t=0 0
    m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendonly---------------------------------------------------------sendonly
    +++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
    [881161,NET]
    INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Date: Tue, 19 Feb 2013 17:39:04 GMT
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 164
    v=0
    o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
    t=0 0
    m=audio 4000 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=inactive---------------------------------------------------------------------media inactive
    At this point, we should get a response back from the sip phone...
    and here is what we got..
    ++Trying which is expected++++
    //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
    [881162,NET]
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    CSeq: 103 INVITE
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Content-Length: 0
    ++++++++Then we get a BYE+++++++++++++++
    /SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
    [881163,NET]
    BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
    Contact:
    Max-Forwards: 70
    From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
    Supported: replaces, path
    User-Agent: Acrobits Softphone Business/2.4.8
    To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    CSeq: 3 BYE
    Content-Length: 0
    So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
    The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
    Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
    for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
    SIP------Media------MTP------------Media-------SCCP Phone
    When the new destination is dialled and transfer is commited,
    SIP-------------media----MTP--------media---------MTP
    The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
    ++++++++Ivite to 492 ++++++++++++++
    INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
    From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
    To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
    Date: Tue, 19 Feb 2013 21:24:59 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 214
    v=0
    o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
    t=0 0
    m=audio 25038 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    +++++++Invite to 491 +++++++++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
    [885429,NET]
    INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
    From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
    To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
    Date: Tue, 19 Feb 2013 21:24:59 GMT
    Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 237
    v=0
    o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195----------------------------------------MTP
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 25030 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Wao! That was a long one isnt it...It was fun too.
    So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

Maybe you are looking for