Int TCS with multiway call recording

Similar Questions

• Video call recording

  Is there a way I can record my own video calls?

  Hi, Superfred, and welcome to the Skype community,

  Looking for third-party compatible with Skype software call recording.

  For example, http://www.pamela.biz/en/download/

  Kind regards

  Elaine

  __________________________________________________________________________________________________
  Your question has been answered? Please click on the link to accept as a Solutionfor everyone can quickly find what works! As a post or want to say, 'Thank You ' -? Click on the button of congratulations!
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• Unexpected behavior with the Option "record in the result.

  Hello

  I have unexpected behavior with the Option "record in the result.

  I have a few steps in the subsequence 'X', this subsequence passes a Boolean parameter. According to the value of the parameter I change the "Recorgind results" Option to report it or not. The thing is that if 'result Recorgind' set at race time I modofy by changing the value of Step.ResultRecordingOption to "Enable" and "Disable", the step is not reported until the same sous-suite 'X' is called for the second time (without changing the parameter passed).

  For example: (Preconditon: result Recorgind Option of all value sous-suite x are defined as Disable)

  1 CallSubsequenceX(Parameter: Enable)

  2 CallSubsequenceX(Parameter: Enable)

  3 CallSubsequenceX(Parameter: Disable)

  4 CallSubsequenceX(Parameter: Disable)

  Expected result:

  1. measures have been reported.

  2. measures have been reported.

  3. measures have not been reported.

  4. measures have not been reported.

  Result:

  1. measures would not same value Step.ResultRecordingOption has been changed to 'enable '. (Not Ok)

  2. measures have been reported. (Ok)

  3. measures reported same value Step.ResultRecordingOption has been changed to 'disable '. (Not Ok)

  4. measures have not been reported. (Ok)

  I use TestStand 2013 (5.1.0.226)

  Thanks in advance.

  -Josymar.

  Hi josymar_guzman,

  I just review the sequence and indeed we´re experience unexpected behavior with the Step.ResultRecordingOption callback. By a reason when you run the callback in the expression before each step section, the statement runs only until the next sequence is called, which is not what we want.

  To avoid this, you can place a statement before each step of the sequence, so you can change the State of the Option "record result" for the sequence running (and it is only the following). You can try something like this

  

  where the expression of the statement will be the recall "RunState.NextStep.ResultRecordingOption is YourCondition". With this, we guarantee that the results of the next step will be saved or not. I also remove the expression in the expression prior to each step section, because the condition is now on the statement before each step.

  I tried and it works fine. I´ll set the sequence that you share with me, with the changes. I hope this will help you and solve your problem.

• Call Blocker for BB Q10 and Q10 BlackBerry Call Recorder does not work

  It's quite funny how when you pay for the apps will not longer work.  I got call blocking for well over a year waiting for them that they told me that I must have 10.3 or higher to be able to use its features.  Well guess what I've been updated 10.3.1.1865 with OS 10.3.1.2708 and it does not block calls.  Call Recorder when free works well, but as soon as I paid for it and upgraded to the pro version it would not record the sound went dead silent.  If anyone has had these problems when trying to get intouch with developers is useless, because no one will send to you.

  Call Blocker is developed by Ajani Info Tech version 1.0.2.17 and Call Recorder is developed by appsforbb version 2.0.0.1.  Sure would like to hear from anyone with the same problems or solutions because it's frustrating.  Thank you.

  Just trying to see if there is someone who has a solution before I go to my service provider to request a refund, but thanks for the help but would still see if anyone has experienced the same problems and have got a solution.  If this isn't the case, I'll go through my supplier for a refund.

  @a wrote:

  This isn't good @rwhowes!

  If you have not already you may want to request a refund.

  From help please visit the site Web of Support of BlackBerry World. Thank you!

• A MultiSITE call may be promoted to a MultiWAY call?

  I have a customer who currently has a number of endpoints with multi-site licenses, mainly 6000MXP units.  The client is developed to the point where 6-pin calls are no longer adequate, and they'll have to buy an MCU.

  My question is:

  Can the customer still enjoy the microcontrollers built in codecs 6000MXP and use a MultiSITE call until the participants are too many and then degenerate into a MultiWAY call?

  Will there be a way to migrate the customer to a MCU usage, perhaps by the personal use of alias of bridge?

  Hi Anthony!

  Can the client still take advantage of the built-in MCUs in the 6000MXP codecs, and use a MultiSITE call until such time as the participants are too many, and then escalate to a MultiWAY call?
  

  No, it is not possible to climb a call running Multisite for a call Multiway or MCU (MXP or C series).

  On the endpoints that will be able to launch the Multiway call is an exclusive setting, therefore,

  Select Multisite, Multiway (or completely disable a Multipoint).

  In addition Multiway must be supported by the endpoint, newest Cisco/Tandberg endpoints points support

  initiate a call Multiway and many (incl. 3rd party) support to be reunited to form a Multiway call.

  It is also dependent on the version of the software.

  In short, with Multiway may as end points that do not support being redirected

  for the MCU (for example the outer parts where you don't have conrtol from software versions,...)

  do not be reunited to form the Conference.

  Is there a better way to migrate the client to the use of an MCU, perhaps through the use of personal bridge aliases?
  

  
  Is there a better way to migrate the client to the use of an MCU, perhaps through the use of personal bridge aliases?
  

  To swap the MXP multisite with a microcontroller of HD Cisco (old Codian) is a great idea in any case, it will not focus only

  the capacity of the largest conferences but also improves quality. Each participant will receive its own optimized

  video stream instead of the single video stream where the worst participant defines what you will get.

  As you said, after meeting personal room numbers (with findme or certain rules of vcs, you can also do named uris,

  as [email protected] / * /), generic virtual meeting rooms or to book through TMS/Scheduler

  may be the solution.

• Since 10.11 I have not been able to download via the usb port of my Davis Vantage weather station. No solution of tech Davis on the issue even with the new recorder of events in the resort. Is there a problem with the usb?

  Since 10.11 I have not been able to download via the usb port of my Davis Vantage weather station. No solution of tech Davis on the issue even with the new recorder of events in the resort. Is there a problem with the usb?

  What Mac? (details)

• The best Skype video call recorder?

  Hello

  I'm looking for the best Skype video call recorder. Which is the best and easy to use?

  Thank you

  Kevin

  Well, I looked for the link on the Skype forums and found that they are on the official site of Skype as well.

  Download here: http://shop.skype.com/apps/Call-recording-audio-video/SuperTintin-Video-Call-Recorder.html

  Hope this will help you.

  Thank you

• TWO wanted work stations, one with my working files and other with personal folders. It seems that spaces works with windows, no records.

  I have OS X 10.11, El Capitan, on MacBook Pro.

  I would like to have TWO offices and one with my files to work with personal folders.

  How can I do this?

  I looked at spaces, but it seems to work only with windows, no records.

  "with my work folders and the other with personal folders": you can not do this.

  The best method is to create two accounts: a personal and a professional. What accounts are for. Pass from one to the other may takes 2 seconds.

• < double > call recording does not work after 5.0.2

  Since I've updated my 2nd generation of bike g lolipop 5.0.2 I can't record my calls, the app only record my voice, but does not record the voice of the one person talk to. have you tried the other app, please can someone recommend a call recorder that works on 5.0.2 .cos I use it for my business to examine complaints and arrested.

  There is already a thread here:

  https://forums.Motorola.com/posts/bf15c93ecf

  Please continue the discussion there before creating new threads on the same issue. Search is your friend.

  Double wire closure.

• No voice with incoming calls PAP2T

  I bought a LinkSys PAP2T unlocked last year that worked beautifully until a few days ago. Outgoing calls work fine, but I now have problems with incoming calls only.

  When I receive an incoming call, the phone rings, but when I pick it up, I can't hear anything and cannot the caller on the other end. After a few seconds, my phone disconnects and begins to beep as if the line is busy. This occurs regardless of who calls me and whther they use a VoIP device or a normal PSTN phone. I have not changed the settings on the router (a Belkin) or the PAP2T, he simply stopped working a few days ago! I have a connection 10MB Virgin Media broadband service for my VoIP provider is Draytel that have verified the account and settings and says it works very well on their test kit.

  Since then, I tried just using line 2 and the deactivation of line 1 to do a hard reset of the PAP2T and updated the firmware to the latest version. Even tried to change the analog phone connected, but I still have the problem.

  Does anyone have any ideas as to why it suddenly stopped working with anything has been changed and what I might be able to do to fix? The PAP2T is still the title of the 12 month warranty period then maybe she developed a fault?

  Thank you very much in advance for all those who can help you.

  Cheers, Vic

  For me it looks like the problem of RTP streams - probably, it does not pass the firewall / nat on the path between SIP provider and you.

  This may have is that your Belkin router/NAT, or on internet ISP router/NAT enabled SIP ALG (Application Layer Gateway).

  If this is the case, you will need to disable SIP ALG on your router or ask your ISP internet to disable on the firewall router.

  Another question may have been if you (accidentally) changed the port forwarding on your Belkin router, or the PAP2T local LAN IP address changed and port forwarding is no longer works.

  So my proposals to check:

  -check and DISABLE SIP ALG on your Belkin (if the router has this feature)

  -check the port forwarding on your router Belkin... IF you set up port forwarding for 5060/61 you MUST set the same for RTP ports (16384-16483)

  -If you don't have port forwarding on your Belkin, try before installation of 5060-5061 AND 16384-16483 *.

  * (If this is too many ports, set the RTP on PAP2T portrange to 16384-16394 for example and then forward only go)

  -If neither of above of aid, change your SIP Line1 of 5060 for port we will tell 6070 and RTP ports to the beach we'll tell 17300-17310

  -Finally, you can do it on your side is to remove all the port forwarding and put the PAP2T to your router Belkin DMZ settings.

• You can save the music from itunes with the sound recorder?

  You can save the music from itunes with the sound recorder?

  The principle is simple in nature, although the tape recorder is not the best option available for you, even in free software for audio recording you can.

  If you want to use sound recorder, you will need to get a 1/8 inch male-male stero (cable basically a helmet with two male ends) and put one of your speakers into your microphone in the port (line in).

  Then open sound recorder and give it a shot. The recorded files will be in .wav format, so you need to get a freeware to convert to program them to .mp3 if necessary. Or use this:

  http://www.media-convert.com/

  You can also use Windows Media Encoder 9, which in my opinion is much better in features and maintaining the quality. It is freely accessible to the public.

  http://www.Microsoft.com/downloads/details.aspx?FamilyID=5691ba02-e496-465a-bba9-b2f1182cdf24&displaylang=en

  You use the same configuration that with tape recorder, get a stereo cable and speaker output connection > microphone.

• Voice call recording still not possible?

  Hi all...

  I've read many threads on voice call recording. I also read where Mr.Mark Sohm and Mr.Simon Hain says it is not possible... I want to ask you:

  It is still not possible even after JDE 5.0?

  Thanks in advance...

  There are services that transmits a phone call via another service, that records the call.

  It is not possible to do it locally on a BlackBerry Smartphone.

• SX 20 with Cisco Call Manager

  Dear team,

  If I add the SX20 with Cisco Call Manager, do I need to install the software cmterm - s52010tc6_2_1.cop.sgn the SX20. If this isn't the case, then what I have to do, I can see only administrator external field in my SX20, where I gave my callmanager IP address but it is not save.

  BR

  Hello

  In order to save the SX20 in CUCM you for CUCM version 8.6.2 or later, and your SX20 must be running TC5 version or a later version.

  This file you mentioned, cmterm - s52010tc6_2_1.cop.sgn, is just a upgrade file that you install on CUCM, so that CUCM can update your point of SX20 endpoint automatically. But you can upgrade your SX20 manually using the file s52010tc6_2_1.pkg.

  To get help on how to register to CUCM SX20, take a look at these guides with the name "administer endpoints TC on CUCM". The Guide according to the version of CUCM you run:

  http://www.Cisco.com/en/us/partner/products/ps11424/prod_maintenance_guides_list.html

  I hope this helps.

  Concerning

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• TC7.2.1 problems with SIP calls

  Hi, we had the following problem with sip calls:

  Here is the log:

  Version of the VCS software: Platform X7.2.3 | X8.2.1 Labor

  ______________________________________________________________________________

  Call with TC 7.2.1 does not
  ______________________________________________________________________________

  SIPMSG:
  | INVITE sip: [email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP xx.xxx.60.133:57597; branch = z9hG4bK38f566387dbd719702049ddcb2590ebe.1; rport
  Call ID: 4832bd6a60d9d3d43670c55c573667ec
  CSeq: INVITE 101
  Contact: sip: [email protected] / * /: 57597; gr = urn: uuid:6 c 856033-4cbd - 5 b 15 - bb13 - 0715c4a694cd, ob>
  From: "201101102_C40_ServiceDesk" sip: [email protected] / * />; tag = d25377890df67530
  To: sip: [email protected] / * />
  Max-Forwards: 70
  Directions: sip:xx.xxx.1.51; lr>
  Allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM
  User-Agent: TANDBERG/521 (TC7.2.1.cb31c3d)
  "Proxy-Authorization: Digest ="bb1d830b2d64949caaadfb732cd7ca77414a991699854e8971970a68fe06"nonce, realm ="tplabvcsc01.xyz.com", qop = auth, opaque = 'AwAAAMSaxlh37 + YNQULdXHDdMkXYHVQ1', user name =" ", uri ="sip:xyz.com", answer is"51f225da16a3ac710e86c1b1b5815438", algorithm = MD5, nc = 00000008 cnonce ="21c839c8313e8f7f7477786cb40c5ee6. "
  Supported: replaces, timer, 100rel, gruu, way, way out
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Length: 2842
   
  v = 0
  o = xx.xxx.60.133 IN IP4 6 2 tandberg
  s = -.
  c = IN IP4 xx.xxx.60.133
  b = AS:6000
  t = 0 0
  m = audio 16424 RTP/AVP 107 108 109 110 104 105 9 15 18 8 0 101
  b = TIAS: 128000
  a = rtpmap:107 m4as-LATM/90000
  a = fmtp:107 profile-level-id = 25; object = 23; bitrate = 128000
  a = rtpmap:108 m4as-LATM/90000
  a = fmtp:108 profile-level-id = 24; object = 23; bitrate = 64000
  a = rtpmap:109 m4as-LATM/90000
  a = fmtp:109 profile-level-id = 24; object = 23; bitrate = 56000
  a = rtpmap:110 m4as-LATM/90000
  a = fmtp:110 profile-level-id = 24; object = 23; bitrate = 48000
  a = rtpmap:104 G7221/16000
  a bitrate = 32000 fmtp:104 =
  a = rtpmap:105 G7221/16000
  a = fmtp:105 bitrate = 24000
  a = rtpmap:9 G722/8000
  a = rtpmap:15 G728/8000
  a G729/8000 rtpmap:18 =
  a = annex b fmtp:18 = yes
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  m = video 16426 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 11
  a = answer: full
  a = content: hand
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = application 28300 UDP/BFCP
  a = the installer: actpass
  a = confid:1
  a = userid:6
  a = floorid:2 mstrm:12
  a = floorctrl:c - s
  a = connection: new
  m = video 16428 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:126

  
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 12
  a = content: slides
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = 16430 RTP/AVP 100 application
  a = rtpmap:100 H224/4800
  a = sendrecv
  m = application 29789 UDP/UDT/IX
  a = ixmap:0 ping
  a = ixmap:2 xccp
  |
  ________________________________________________________________________________________________

  Work of appeal with TC 7.1.4
  ________________________________________________________________________________________________

  SIPMSG:
  | INVITE sip: [email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP xx.xxx.60.133:37491; branch = z9hG4bKe3dfed840a299516919a79e4a54cd707.1; rport
  Call ID: 480ec504782c0ad894929e882a696e74
  CSeq: INVITE 100
  Contact: sip: [email protected] / * /; gr = urn: uuid:6 c 856033-4cbd - 5 b 15 - bb13 - 0715c4a694cd, ob>
  From: "201101102_C40_ServiceDesk" sip: [email protected] / * />; tag = 3ab87b1c0d8b4d9d
  To: sip: [email protected] / * />
  Max-Forwards: 70
  Directions: sip:xx.xxx.1.51; lr>
  Allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM
  User-Agent: TANDBERG/520 (TC7.1.4.908e4a9)
  Supported: replaces, timer, 100rel, gruu, way, way out
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Length: 2842
   
  v = 0
  o = xx.xxx.60.133 IN IP4 3 2 tandberg
  s = -.
  c = IN IP4 xx.xxx.60.133
  b = AS:6000
  t = 0 0
  m = audio 16394 RTP/AVP 107 108 109 110 104 105 9 15 18 8 0 101
  b = TIAS: 128000
  a = rtpmap:107 m4as-LATM/90000
  a = fmtp:107 profile-level-id = 25; object = 23; bitrate = 128000
  a = rtpmap:108 m4as-LATM/90000
  a = fmtp:108 profile-level-id = 24; object = 23; bitrate = 64000
  a = rtpmap:109 m4as-LATM/90000
  a = fmtp:109 profile-level-id = 24; object = 23; bitrate = 56000
  a = rtpmap:110 m4as-LATM/90000
  a = fmtp:110 profile-level-id = 24; object = 23; bitrate = 48000
  a = rtpmap:104 G7221/16000
  a bitrate = 32000 fmtp:104 =
  a = rtpmap:105 G7221/16000
  a = fmtp:105 bitrate = 24000
  a = rtpmap:9 G722/8000
  a = rtpmap:15 G728/8000
  a G729/8000 rtpmap:18 =
  a = annex b fmtp:18 = yes
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  m = video 16396 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 11
  a = answer: full
  a = content: hand
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = application 29489 UDP/BFCP
  a = the installer: actpass
  a = confid:1
  a = userid:3
  a = floorid:2 mstrm:12
  a = floorctrl:c - s
  a = connection: new
  m = video 16398 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000

  
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 custom = 1280, 768, 3; custom = 1280, 720, 3; custom = 1024, 768, 1; custom = 1024, 576, 2; custom = 800, 600, 1; cif4 = 1; custom = 720, 480, 1; custom = 640, 480, 1; custom = 512, 288, 1; CIF = 1; custom = 352, 240, 1; QCIF = 1; maxbr = 20000
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 12
  a = content: slides
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = 16400 RTP/AVP 100 application
  a = rtpmap:100 H224/4800
  a = sendrecv
  m = application 31127 UDP/UDT/IX
  a = ixmap:0 ping
  a = ixmap:2 xccp

  Another problem, indicated in diagnosis on a MX200 device tests, reporting problems of NTP. I tried to choose another NTP server - but this does not solve the problem, and it works with TC7.1.4. In addition to the time and date is wrong on the screen but OK on the touchscreen device.

  In addition, an error is reported on the OSD settings.

  Do you have advice?

  Thanks for help.
  |

  So, what's the problem? You just showed us an INVITATION. What is the configuration of your system? What is the reaction to INVITE him? We need complete diagnostic logs. Is this VCS - C registred? Where VCS - C diagnostic logs

• Anyone having problems with stacked calls?

  I have been stacking create/edit lead and senior partner with external calls to campaign in a single event for years.  Monday, all our calls stacked broke.  Create/change drive call fire very well. The campaign associated with lead call fails.  I can re-run their fine since the error log, which means that the call of the associated driver is fire prior to the initiative to create a call now, OR more probably that the SFDC ID drive is not charged enough faster now for the campaign to create fire.

  All flows into that?

  Support confirmed that associate campaign calls are indeed fire before creating lead calls sometimes.  The options are desempilement of your calls or to manually run your failures.

  Fortunately, I have a third option that accompanies us.  SureShot created an app for us that allows us to circumvent of Eloqua queue outgoing sync and sync leads in 40 seconds approximately.  We decided to move the lead to create/edit all calls through the app instead.