Problems with SIP Trunk (an audio course)

Similar Questions

• 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

• Problem with Windows Media Player Audio sync DivX playback in full screen

  Hello

  I have found lately that when playing videos on Windows Media Player divx mode full screen audio is out of sync.  If the video is played in window mode, there is no problem.  I had no such problem in the past, but now for about 1 month or so, I have this problem.

  I have fitted iq508a to vista 64-bit, with the latest drivers delivered HP installed (using the HP update utility)-does anyone else have this same problem too?

  Thanks in advance

  Jim

  Laptop HP ENVY 17-j005tx, HP ENVY 27 - k001a, HP ProLiant MicroServer Gen8 G2020T, HP MediaSmart EX495 Server tilt, HP MediaVault 2020, HP ENVY 120 AiO Printer

  I can't believe! I solved the problem with this problem of video audio sync during playback in full screen!

  While watching the Pineapple Express, I thought WTF I'll take another closer look at this, and while they were smashimg their cell phones so that they could not be found by triangulation, (don't ask - u just I have to watch - stoner movie )...

  in any case, I looked under tools options in WMP.  I unchecked the option on the performance tab "use video smoothing"-the video as reloaded and bingo! everything in harmony!

  BTW, I checked my another PC running Vista 32 bit and the option ' use video smoothing ' also exists and it is checked if this seems to be a problem with WMP works on Vista 64 bit or running desktop Touchsmart of WMP... In case I am much happier now...

• Problem with the keyboard and audio on Satellite L40

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  I just installed XP SP2 on my Satellite L40 (BTW thanks to all of you who helped me overcome the problems during this operation with your son), but I have a problem with my keyboard. I have a QWERTY keyboard and after I installed xp, he worked as a qwerty, I fixed it, now all my letters are working very well, but I have some problems with certain symbols, such as @,., _, if you understand me.

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  I also have problems with my audio, in fact I have no sound at all, in the Panel, he said that there is no audio device found.
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• I am facing problem with playback of the audio recorded.

  Original title: sound recording problem

  Use built in mic, I recorded w/vocals/guitar,... and playback starts well, but after a few seconds the sound loses the basis... or low frequencies.

  It is uniform everytime I try. What I am doing wrong?

  Rich

  [Moved from comments]

  Hello

  What version of the Windows operating system is installed on your computer?

  If I understand correctly you are facing some problem with the audio recorded. There is no way to lose the sound after a few seconds; There seems to be a problem with the registration. I suggest you to run the game audio recording Fixit then try to register once more and check if it works.

  Automatically diagnose and fix problems of Windows audio recording:

  http://support.Microsoft.com/mats/AudioRecording/

  For more information, see the following link:

  http://Windows.Microsoft.com/en-us/Windows7/record-audio-with-sound-recorder

  Hope the information is useful.

• Problem with the playback of audio books

  Hi, I wonder if anyone can help please I have a problem of reading audio books on my rocket.  The problem occurs when I download audio books from my local library; the library allow it, are aware it will be downloaded to an MP3 player and no permit is required.  Most of the books are multi frame for example from 1 to 10.  I have download all 10 parts of my rocket.  Parts 1 & 2 always play without problem. When I try to play other parts 3, 4 etc. they will not play and I get a message saying: 'you can get the license by transferring the title to the device using software audiobook.  No license is required and he plays parts 1 and 2 with success.  I tried to download only 1 & 2, they play, then delete 1 & 2 and 3 & 4 download, they will not play and I get the same message.  In parts 1 and 2 of any book, I download play, the rest of the parts will not and the message above.  In the past, I managed to download books and had no problems.  I would be grateful for any help please?  Thank you.

  Thanks to this solution of sugested.  I now have the issue resloved, namely that I don't have my camera "MPT" the value of the parameters of the USB, it has been updated the 'MASTER '.  I made the change deleted audio books that are not open, downloaded again and now all THE game pieces.  I have this post (1) to say thank you and (2) in case it helps someone else.

• Problem with DELL Inspiron 3521 audio!

  Hello

  I bought this laptop DELL Inspiron 3521 sometimes back and I am facing a problem with the audio. I had the experience that I use only. Audio keeps switching from left to right and sometimes works only on one side. Initially, I thought that it was a problem with my headphones but I tried with several headphones and still I have experienced the same problem. But the speakers of the laptop works very well. Could this be a problem with the audio port? Can someone help me on this.

  Thank you

  Umesh

  umeshjaishankar

  Could this be a problem with the audio port?

  Hello, Daniel. How to determine who must pass the native Windows audio driver Realtek audio driver. If you don't have the driver from Realtek, you can get it from the Inspiron 3521 support page.

  1. open Device Manager (type devmgmt.msc in the box of Windows search).
  2. develop the 'audio controllers, video & game' and right click on "Realtek High Definition Audio".
  3 Select this option to "Update driver software".
  4. click on "Browse my computer for driver software".
  5. click on "Let Me pick from a list of drivers on my computer".
  6. put the "Show compatible hardware" check box if not already checked.
  7. in the list of devices, click on 'High Definition Audio"(native driver).
  8. click on "next".
  9. in the update driver warning box, click 'Yes' (the driver).
  10 restart the portable computer if you are prompted. If not invited, so no need to restart.
  [To return to the Realtek driver, do it again but reverse the names in steps 1-6.]

  If the headphone does not work correctly when you use the Realtek or driver audio native then it's a hardware problem.

  If it works fine with the native driver but not the Realtek driver, then the Realtek driver is buggy. In this case let me know which operating system your laptop and I'll see if there is a Realtek driver different, that you can use.

• Configure (fxs) analog phones with Sip trunk

  Hi all

  IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.

  Configuring fxs to work with her?

  any help thanks.

  Is there a PBX at stake here, i.e. CUCM or CME?

  In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.

• Problems with SIP when saved in VCSE

  Hi guys,.

  VCSE and VCSC are both on X5.2. When a customer movi entered the VCSE and we try to dial in the MPS800 which is registered to the VCSC participants receive no video or audio. However even when call is made by using the thin fucntions H323 call. Is it possible to force the vcse do a h323 through the firewall on the highway.

  Thank you

  No, you must configure VCS - C address as the address of the SIP on MPS server.

  In the otherwise static Conference SIP and unique dial number with SIP don't enroll not on VCS - C.

  For VCS sent the appeal to members when receiving URL Conference personal other, you must configure in the shot (just repeat what I mentioned earlier).

  ====================================================================================

  [MPS]

  -SIP server: the VCS - C IP address

  -Default transport protocol: TCP (you can use TLS if you want)

  -Configure the personal URL of conference (with a specific format can easily configure rule research on VCS to MPS not to overlap with other aliases to point of termination/SIP UA).

  [CV]

  -Neighbouring area

  Type: neighbor

  H323 Mode: Off

  SIP: Mode: on

  SIP port: 5060 (assume MPS transport protocol configured as TCP)

  SIP Transport: TCP (suppose MPS transport protocol configured as TCP)

  Peer 1 address: address IP SC SPM

  Area profile: default

  -Rule search

  Mode: Alias matching

  Model type: Regex

  String pattern: [email protected] / * / (personal Conference suppose aliases is [email protected] / * /)

  Model behavior: leave

  Target: MPS-SIP-neighbor (this is the nearby area SIP created and point members)

  ====================================================================================

  With above configuration, VCS - C before calling SIP with the URL [email protected] / * / forward to members of Parliament.

  I would recommend you first do a test call in VCS - C (do not call VCS - e) to verify the call.

• BlackBerry Smartphones Hi there - I am new here but experiencing problems with my Storm 2 audio streaming

  I have terrible problems with my Storm 2 like very weak signal in the House - when my partner on the same network, but using a curve has no problems (she gets 4 bars of signal, I'm lucky if I get 2)

  also, I can't stream audio from a site I want to hear - something that she is capable of doing.

  When I try to live from a radio station, I get the following message: «!» "This selection contains no valid songs" memory buffer... is displayed in the upper right corner, but no connection is ever made.

  Any ideas what I need to do to correct the thing?

  Also I can listen to or download mp3s I put it online for access via my storm (unfortunately I use mac os x 10.4.11 and the blackberry desktop manager is only for os x 5.5 or higher - mark/space missing syncalso has serious problems with the synchronization of all media to the storm)

  could do with some really helps and have had a good look but can find very little to help me so it's a bit of a long first post, I'm afraid.

  Hi and welcome to the forums!

  A few places to visit: Blackberry 101    tips & tricks

  I would check the Storm Forum

  There are a lot of great users out there who can help get your wheelchair storm.

  It's great to have you here! Keep coming back.

• Problems with SIP scheduling TMS > TP server

  I have a problem of planning where the TMS seems to say my TP server to dial a number to the preconfigured endpoints rooms/external to H323, despite endpoints being configured only with a SIP URI and no ID of H323.

  My external termination points are added as 'rooms' to TMS.  They 'allow reservations' and ' allow incoming SIP URI configured dialing, but all the other slots to be unchecked. "  They have no ID H323, E164, or configured firewall (gatekeeper is set to "off".

  When I have distributed them in a conference, connection settings developed as 'SIP-H323"instead of just"SIP", so he tries H323 numbering first.  It is a problem because many of these external endpoints are CTS-3000 units and if composed as H323, TIP does not work they only connect with a single screen.

  If I manually dial the SIP from the server TP, or if I add an external endpoint to the Conference through TMS and specify SIP, it works very well. It is only a matter because TMS seems to want the MCU try H323 first, despite the configuration 'House' with no option availible H323.

  It just came out recently is because we use TP Server 2.2 in conjunction with the CUBE and manually add endpoints to TP server as "legacy" systems of CTS.  We're heading TP Server 3.0 + VCS - E where TRICK works automatically, but it doesn't seem to work if the call is interoperability of H323, SIP.

  Any ideas how to get MSDS to compose rooms as SIP only (or at least first try SIP)?

  Versions:

  3.0 (2.48) Server TP

  TMS 14.2.2

  VCS 7.2

  Hi Nick,

  I had the same problem with TMS 14.2.2. I have fixed only after you have configured the 'Active SIP server address' field in the configuration page / equipment of the room. You can put any IP address in this field, any.

  It seems that TMS sets TP server to call using SIP only when this field is set to the configuration / the equipment in the room, if I leave this field blank, the result will be the same problem that you are experiencing, a configuration of the connection with H323--> SIP.

  Just to repeat, these are the areas that I have configured in hardware Configuration room to have work with SIP only:

  SIP mode: on

  Active SIP server address: 10.10.10.10

  SIP URI: [email protected] / * /

  Gatekeeper discovery: Off

  Allow the reservation: check

  Allow incoming SIP URI numbering: check

  IP maximum bandwidth: 6000

  You also need to set "Maximum bandwidth IP" field. If this field is '0', you will get an error message "no possible route between participants: TPserver and participating: tests.

  I hope this helps.

  Concerning

  Paulo Souza

  Please note the answers and mark it as "answered" as appropriate.

• Problem with iPhone 7Plus, lightning audio commands, connector and inline

  Just bought a new iPhone 7 more and understand that there more listener intake. I have a pair of Bose In-Ear-Headphones who controls inline for voice/mic, volume, etc. But when connected to the adapter and plugged into the connection of the lightning and listen to music, the controls do not work. Others found that I'm certainly not the only one having this problem? The sound is very good, but the controls works do not, concerned the micro does not work either. It is my main use of headphones for meetings and teleconferences.

  Thoughts?

  Kind regards

  Bryan

  I have the same problem with my iPhone7, when you unplug and plug it in new job wil.

  After releasing music with the control, you can put it on once again, but when you wait long but when waiting in time he won't answer any more until you unplug and plug back taking.

  I contacted the support and there seems to be a software bug.

  Restoring say ore phone wait for the first update of IOS10

  Hello Michel,

• problems with windows 7 in the course of running games msn and msn explorer__

  Why do I have problems with msn games, opperating with windows 7 in addition cant use msn explore

  Hi jermaynard,
   
  This question would be more appropriate in the Windows Live Help forum. Please post your query it better support.
   
  Here is the link:
   
  Welcome to Windows Live Solution Center
  http://www.windowslivehelp.com/

  Kind regards

  Shinmila H - Microsoft Support

  Visit our Microsoft answers feedback Forum and let us know what you think.

• Problems with my RealTek HD audio device

  My RealTek audio device manager does not work, I tried to download it from downloads.com, but my computer stop himself... I thought he caught a virus: P any suggestions please... A little, I can't run without my sounds!

  Hi Tinks774,
   
  If you are suspicious of virus attack, an analysis online on your computer for this problem
   

  Check out the links given below to do the same thing:

  http://OneCare.live.com/site/en-us/Center/whatsnew.htm

  Microsoft Security Essentials

  Check if an updated audio driver is available from the website of the manufacturer.

  You can use Microsoft Update to help you determine your current sound driver and if an update is available. Then communicate with the hardware driver sound or the computer manufacturer for an updated audio driver.

  See the link below for more details:

  How to update a sound card driver in Windows Vista and Windows XP

• Problem with Windows Movie Maker - Audio is messed up

  OK, so I'm doing a film, with the lyrics of a song. I put pictures of her and paired words. It worked fine for a few songs. Then when I went to make another, I imported the song, but the time of audio changes. I mean, it's that I can't say... 0:20 seconds in the song and it should play the same part of the song, whenever I go to 0:20 seconds. On mine, it keeps switching autour. I'll go to 0:20 seconds and it will play what he normally plays at say 0:15 seconds. Or I'll set it to 0:40 seconds and he will play the music to 0:20 seconds. That's when I jump to that point in the song. If I start from the beginning and let him play through, then he plays at the normal time to 0:20 or 0:40 seconds. But he messes up and plays the wrong thing when I skip some time that is not at the beginning. Does anyone else have this problem? How should I do? Should I uninstall WMM and redownload it? Or is it a thing of setting? Help, please! Thank you!

  I had this same problem. Assuming that the music is your use of WMM in iTunes, follow this:

  1. open iTunes

  2. click on Edit--> preferences

  3. click on import settings

  4. in the drop-down arrow next to the import to help:, select WAV encoder

  5. press the OK button

  6. now click on the songs you want to use in WMM

  7. click on advanced and click on create the WAV Version.

  8. now, when you return to WMM, click on import Audio or music and go to your iTunes folder. Versions WAV songs you create must be there and when you import, you should have no trouble in :D