Problems with SIP scheduling TMS > TP server

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

• Problems with SIP Trunk (an audio course)

  Hello world!

  Our client is testing a new implementation of SIP with a different ISP trunk.

  They have a SIP between a Cisco 2911 and ISP trunk to access the PSTN and a H323 trunk between CUCM worm 7.1.3.30000 - 1 good routing of calls to the Cisco2911 gateway.

  Here you have the Cisco 2911 configuration:

  VoiceGW-B #sh runn
  Building configuration...

  Current configuration: 9341 bytes
  !
  ! Last configuration change at 19:09:50 AST Thursday, January 24, 2013, by admin
  !
  version 15.0
  Service nagle
  no service button
  tcp KeepAlive-component snap-in service
  a tcp-KeepAlive-quick service
  horodateurs service debug datetime localtime show-timezone msec
  Log service timestamps datetime localtime show-timezone msec
  encryption password service
  sequence numbers service
  !
  hostname VoiceGW-B
  !
  boot-start-marker
  boot-end-marker
  !
  map of type t1 0 0
  enable secret 5 $1$ T05j$ vJkR0V2l2/Iu1mIIeVPcu1
  !
  No aaa new-model
  clock timezone AST - 4
  network-clock-participate wic 0
  network-clock-select 1 T1 0/0/0
  !
  No ipv6 cef
  IP source-route
  IP cef
  !
  !
  !
  !
  IP domain name domain.local
  Authenticated MultiLink bundle-name Panel
  !
  !
  !
  !
  primary ISDN switch type - or
  !
  !
  !
  voice-card 0
  dspfarm
  DSP services dspfarm
  !
  !
  !
  voip phone service
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  Fax protocol t38 nse force ls-0 hs-redundancy redundancy 0 relief pass through g711ulaw
  H323
  SIP
  90 min - to
  header-passage
  offer-early forced
  midcall-signalling passthru
  !
  class 333 voice codec
  g711ulaw codec preference 1
  codec preference 2 g729r8
  !
  voice class codec 2
  g711ulaw codec preference 2
  g711alaw preferably 3 codec
  !
  voice class codec 1
  g711ulaw codec preference 1
  g711alaw preferably 2 codec
  preferably 3 codec g729r8
  !
  vocal h323 class 1
  H225 timeout tcp establish 3
  !
  !
  !
  !

  !
  !

  !
  redundancy
  !
  !
  controller T1 0/0/0
  long CableLength 0dB
  time intervals PRI - Group 1-24 service mgcp
  Description SF 137-6042 primary (GWYN - A 137-6041 redundante)
  !
  !
  !
  !
  !
  interface Loopback0
  Description * USED for IPT, ROUTING, MANAGEMENT ETC... *.
  192.168.100.11 IP 255.255.255.255
  no ip redirection
  no ip proxy-arp
  H323-gateway voip interface
  H323-gateway voip bind port 192.168.100.11
  !
  interface GigabitEthernet0/0
  trunk SIP ISP description
  IP 120.100.11.135 255.255.255.128
  NAT outside IP
  IP virtual-reassembly
  automatic duplex
  automatic speed
  !
  interface GigabitEthernet0/1
  Description * has P2P to 4506 Core *.
  IP 192.168.101.6 255.255.255.252
  no ip redirection
  no ip proxy-arp
  automatic duplex
  automatic speed
  !
  interface GigabitEthernet0/2
  Description * P2P to 4506 Core B *.
  IP 192.168.101.14 255.255.255.252
  no ip redirection
  no ip proxy-arp
  automatic duplex
  automatic speed
  !
  interface Serial0/0/0:23
  Description * channel ISP_2 D *.
  no ip address
  encapsulation hdlc
  primary-dms100 ISDN switch type
  ISDN incoming-voice
  ISDN-Manager of ccm of bind-l3
  No cdp enable
  !
  !
  Router eigrp 100
  network 192.168.100.11 0.0.0.0
  network 192.168.101.6 0.0.0.0
  network 192.168.101.14 0.0.0.0
  passive-interface default
  no passive-interface GigabitEthernet0/1
  no passive-interface GigabitEthernet0/2
  EIGRP stub connected summary
  !
  IP forward-Protocol ND
  !
  IP http server
  no ip http secure server
  IP http access path flash: / GUI
  !
  IP route 120.100.0.0 255.255.0.0 120.100.11.129
  !
  record 10.2.173.5
  access-list 1 permit 192.168.5.0 0.0.0.255
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !

  control plan
  !
  !
  voice-port 0/0/0:23
  !
  Voice-port 1/0/0
  !
  Voice-port 1/0/1
  !
  Voice-port 1/0/2
  !
  Voice-port 1/0/3
  !
  Voice-port 1/0/4
  !
  Voice-port 1/0/5
  !
  Voice-port 1/0/6
  !
  Voice-port 1/0/7
  !
  CCM-Manager redundant-host 192.168.4.11
  CCM-Manager mgcp
  music of blocking CCM-Manager
  !
  MGCP
  type of service mgcp MGCP call-agent 192.168.4.12 version 0.1
  codec to voip MGCP dtmf-relay all the out-of-band mode
  MGCP rtp inaccessible timeout 1000 action notify
  voip MGCP modem ESN passthrough mode
  MGCP ip qos dscp cs3 signaling
  MGCP package rtp-package capacity
  MGCP package-capability OSH-package
  MGCP package-capability pre-package
  No package-ability mgcp package-fxr
  No mgcp timer receive-rtcp
  MGCP sdp simple
  MGCP t38 fax inhibit
  MGCP rtp payload type static g726r16
  MGCP bind control source-interface Loopback0
  MGCP bind media source interface Loopback0
  !
  profile MGCP default
  !
  !
  voice pots Dial-peer 10
  Service mgcpapp
  port 1/0/0
  !
  voice pots Dial-peer 11
  Service mgcpapp
  port 1/0/1
  !
  Dial-peer voice 12 pots
  Service mgcpapp
  port 1/0/2
  !
  voice pots Dial-peer 13
  Service mgcpapp
  port 1/0/3
  !
  voice pots Dial-peer 14
  Service mgcpapp
  port 1/0/4
  !
  voice pots Dial-peer 15
  Service mgcpapp
  port 1/0/6
  !
  voice pots Dial-peer 17
  Service mgcpapp
  port 1/0/7
  !
  Dial-peer voice 16 pots
  Service mgcpapp
  port 1/0/5
  !
  Dial-peer voice voip 3001
  your reminder alert-non-PI
  Description * Testint ISP OUTGOING for LOCAL CALLS *.
  translation-profile outgoing DN-to-E164-srst
  preference 10
  destination-model 12122067379
  session protocol sipv2
  session target ipv4:120.100.1.10
  numbers-fall of DTMF-relay rtp - nte
  Codec g711ulaw
  No vad
  !
  Dial-peer voice voip 9004
  Description * CM. PRIMER NOT piloto *.
  preference 1
  destination-model 1358
  session target ipv4:192.168.4.11
  codec voice-class 1
  DTMF-relay h245 alphanumeric
  IP qos dscp cs3 signaling
  No vad
  !
  Dial-peer voice voip 9005
  Description * secondary CM for ONLY piloto *.
  preference 2
  destination-model 1358
  session target ipv4:192.168.4.12
  codec voice-class 1
  DTMF-relay h245 alphanumeric
  IP qos dscp cs3 signaling
  No vad
  !
  Dial-peer voice voip 999
  SIP INBOUND DIALPEER description
  incoming called-number.
  DTMF-relay rtp - nte
  Codec g711ulaw
  !
  !
  NUM - exp 12126169799 1358
  2122067379 12122067379 NUM - exp
  entry door
  receive timer-RTP 1200
  !
  !
  !
  access controller
  Shutdown
  !
  !

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

  Well, we can set up incoming and outgoing calls with no problems during this test phase, but we will succeed voice entering.

  We don't have voices coming out of the voice gateway.

  We checked with the ISP and we see the RTP of ISPS to Cisco 2911Voice gateway traffic, but we did not see packets RTP voice to the ISP gateway.

  In fact, it was not all RTP packets arriving at the voice gateway on the internal network.

  Might be a routing problem?

  Internal CUCM and phones require Ip routing SIP from the ISP server access? If I understand correctly the devices internal only need to know how to get to the voice gateway Cisco2911, so it can function as a Proxy traffic and route to the SIP server?

  Thank you

  In addition to the comments of Chris,

  1. There is a routing problem: IP phones should see the route to the ISP, even if they are inside a NAT.

  2. If you want that:

  -Just IP phones reach the 2911 and IP of 2911 present the call to the ISP.

  -the Loopback0 bring the H323

  - And the int GigabitEth 0/0 for the SIP

  then

  Configure the 2911 as a CUBE in path mode

  Use the redirection ip2ip

  Configure dspfarm on the 2911

  3 also check this:

  If you have not seen all the voice gateway to ISP RTP packets

  Then

  -Check if the transport of the ISP session is TCP or UDP.

  -Set up a GUY on the 2911 to check the communication between the {2911 and ISP} and {2911 and CCM}

  Kind regards

  Antra

• Problem with directory of TMS to Jabber Video distribution

  Hello

  Recently, I have a problem with the configuration of the directories for video Jabber clients. Looked through a few topics that seem related but found them a little different.

  In short: TMS 13.2 configured Extension commissioning and off FindMe, VCS Expressway 7.2, Cisco Jabber client video v4.5.

  Steps performed:

  1. a new user has been created (inside Provisioning > users) with [email protected] / * / and Device_Address_Pattern =

  [email protected] / * /.

  2. this user got automatically in the Provisioning directory (which already had a few other contacts inside)

  3. I have connected the video Jabber client with the new account, but can not find all the users via the search field within the customer (TMS illustrates usage of licenses) put into service.

  It is easy to manually assign phone book on a device by using the function 'Set on systems. But what do I do to get his Jabber Client video inputs?

  Thanks in advance!

  Hi Alex

  In the management of directories if you click on a directory there are Access tab control that allows you to assign access to directories for users.

  Then you must make sure that you the directory server URI in the configuration of the user model commissioning as [email protected] / * / .

  Make sure then that the TMS and the VCS is synchronized, you should now be able to search for directories if everything is correct.

  / Magnus

• 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.

• I'm having a problem with sharing folders on Windows Server.

  Until this sharing folders to another user, I use the windows login password to access the folder

  How to use a different password, if I want the user to access, but I don't want to create another user account?
  I had trouble creating permissions of the user if I need to create user accounts.
  With the help of Windows Server R2
   
  Original title: sharing folder for the particular user using a different password for password sharing

  Hi jytan,

  Thanks for posting in the Microsoft Community.

  You have a problem with sharing folders on the computer. You might have tried a number of troubleshooting steps. The efforts you have put in are commendable. I appreciate it.

  I would have you post your query in the TechNet Forums because it caters to an audience of it professionals. Your question will be better addressed there.

  Check out the link-

  http://social.technet.Microsoft.com/forums/en-us/category/WindowsServer

  We know if you need help. We will be happy to help you. We, at tender Microsoft to excellence.

  Thank you.

• Problem with iface on windows 2008 server

  Hello. I have 2 identical servers with controller intel 80003es2lan gigabit ethernet based on supermicro x7dbu motherboad.

  When iam power on my servers, all virtual machines (windows 2003, xp, linux, freebsd) start normally, but in the virtual machine based on windows 2008 network interface server is not started.

  I can ping from vm to net, but cannot ping NET to vm. IAM change vm interface e1000, vmxnet2, vmxnet3 but the problem has not been resolved.

  Same problem on any edition of windows server 2008 (no sp, sp1, r2, 32-bit and 64-bit) on all my servers.

  Problem solved when iam remove network iface and rescan devices. I had to power on my servers. It's very bad because iam must reconfigure all the ifaces network.

  How to solve this problem without inserting another network card?

  Welcome to the VMware forums communities.  I moved your post to the forum of the Virtual Machine and the guest operating system. > I can ping from vm to net, but cannot ping NET to vm. IAM change vm interface e1000, vmxnet2, vmxnet3 but the problem does not resolve itself by default Windows 2008 does not respond to ICMP packets.  You must change the Firewall setting to enable this.

• Problem with connecting to the external server

  I am currently using my iMac with Mac OS X Lion 10.7.5 2011 (11G 63). I have an external server connected to the PC of the window. I tried many times and many ways and yet the error "There was a problem connecting to the server" continues to appear.

  I checked:

  -IP address
  -No firewall
  -Access rights
  -Address external & directory servers

  I used my other new imac and macbook and they have no problem connecting.

  Is this an error in version?

  Please do help.

  Read this article on the basis of knowledge of Apple, there may be something you've missed.

  OS X El Capitan: connect to shared computers and servers for files on a network

  Also this...

  OS X El Capitan: to connect to a Windows from a Mac computer

• Problem with DAQmx Schedule VI (sample clock)

  Hello to you all,.

  I'm new to this forum, please bare with me. I have some experience with LV, but I am relatively new to data acquisition projects. I use LV2009.

  I want to make sure that I use the hardware timing (instead of software distribution) in my project so I followed some of the threads here as sugested to use DAQmx Schedule VI. The problem is that no matter how I set the system I get the same error-200300 invalid calendar
  type.

  The project is simple. I encode with 1000 pulses per
  Rev and it is mounted on a shaft of a turbine water goes thru. I'm watching the frequency
  and so the rotation of the shaft which tells me that the amount of water flows through the turbine. In the end, there will be 2 channels
  by every encoder and ~ 3 encoders (turbines) total and calibrated the main meter that will give me constant impulses and all encoders will be compared to this master frequency.

  I'll use PCI6602 DAQ, but
  now, for the development, I use USB6221. Let's say that the
  frequency is between 500 Hz and 10 kHz. What I am doing wrong? Or maybe better to ask - what would be the right approach for this project?

  Thank you

  Marty

  Hi Marty,

  It seems that your question is already answered here, but Jason is correct that the 6221 neither the 6602 support a clock sampling for frequency measurements.

  As Jason mentioned, your best bet is also likely set the mode of synchronization for "implied". This means that the frequency value is sampled at the end of each period of your input signal. In addition, a solution that is clocked by the software (On-Demand) might be acceptable.

  X Series DAQ devices allow an external sample clock to use for frequency measures (described in the Manual of X series). Frequency of sample-clocked measures are useful in very specific
  circumstances, but it does not seem that you need this feature based on what you've described so far.

  (621 x) bus-powered M series can also be configured to use an external sample as the X series clock but do you not have the same features described in the manual of the X series.

  I hope this helps!

  -John

• SQLDev 4.1 ai2 - problem with connection SSO to SQL Server (new)

  I am trying to connect to SQL Server with the new version. I have the jtds 1.3 installed and get the configuration of SQL Server tab, but once more to get the infamous SSO error trying to connect:

  Status: Failure-i/o Error: failure of the SSO: library Native SSPI has not loaded. Check the system java.library.path property.

  
  

  I tried to copy ntlmauth.dll in several places:

  C:\Oracle\sqldeveloper4.1\sqldeveloper\sqldeveloper\bin

  C:\Oracle\sqldeveloper4.1\sqldeveloper\jdk\jre\bin

  C:\Oracle\sqldeveloper4.1\sqldeveloper\jdk\jre\

  C:\Oracle\sqldeveloper4.1\sqldeveloper\jdk\

  Given SQLDev running, but still get the error of SSO.

  Discovered this problem (thanks to Jeff Smith!). I had a 32-bit DLL in the path that are in conflict with the 64 bit one. Once I deleted all of the extra 32-bit versions, the SSO authentication works.

• Problem with connection to a Teamspace Server

  Mac OS X Yosemite 10.10.3

  Acrobat PRO XI

  I have problems connecting to a Shared PDF review hosted by our Teamspace server when you use Acrobat Pro XI in the MAC environment.  I see this error message in the status of the server.

  "Error: could not get new comments." Acrobat cannot access the network. Please check your computer's network connection and try again".

  I also have Acrobat X Pro under Windows in VM Ware installed on the same Mac and it connects fine.

  Can someone help me with a few setting changes on the MAC side that will allow me to connect to our server Teamspace?

  Sorry, we use an internal server. The problem occurs usually when there is an update on the server. Sorry, not sure if this is useful.

• What is the problem with Distiller? and Distiller Server?

  I used to use distiller to make my PDFs a smaller size file more optimized. A few months ago we have updated CS4, well everything that was before any version of CS. So for these last months, I had no problems at all making it reasonable size/quality PDF directly from ID... Yes, so I still want to know, is at - it not necessary at all for distilling make file sizes even smaller than the straight lines of ID?... or I just think, all these parameters are avilable in export ID.

  .. do you a large volume in which to use Distiller Server? Does this sound right?

  Adobe actually deal with enourages users to use the direct-to-PDF route without distilling since it is a richer conversion. Distiller is used via Microsoft Agent (and the PDF Maker) as another method of creating PDF files (via postscript [and sidecar with PDF Maker]) and indeed, sometimes professional users who need to distill server for a large number of operations of PDF creation.

  Distiller Server is not available for Mac.

  Jon

• Problem with LabPython - implement the Python Server

  I booted the server as follows:

  

  It was made as a separate VI, and I ran it before the opening of any other VI.

  Initializes the server, but I can't get the variables of the
  the code is calculated. Here is a very simple program, with its
  error code.

  

  I was able to run this code previously. The variable for the VI of data is identified as an integer.

  Attached is the VI above

  You forgot one thing.

  

• Problem with the DNS in windows server 2008

  When I installed the dns server role and ADDS using dcpromo, I get the error message saying that the IP address is not valid even it's a static IP 192.168. *. *

  Hi Pradeep,

  The problem you are having is more complex than what is generally answered in the Microsoft Answers forums. Appropriate in the TechNet forums. Please post your question in the TechNet forums.

  You can follow this link to ask your question:

  http://social.technet.Microsoft.com/forums/WindowsServer/en-us/home?Forum=winservergen

  For any other corresponding Windows help, do not hesitate to contact us and we will be happy to help you

• problem with pages on the glassfish Server

  Hi all

  My version Jdev: 11.1.2.4

  Glassfish server: 3.1.2.2

  When I was trying open the jsf/jsp page after the deployment of adf application in glassfish server, it opens with the page source code rather than design. and it is showing the message as

  "XML this file does not appear to be any information of style associated with it." "The document tree is shown below."

  and the page is UriPattern below

  

  Please help me on this issue. And let me know about the comparability of the server with the version jdev

  Thnaks,

  I think that your web.xml file or the URL you are using is not correct.

  You must ensure that your page is processed by the engine JSF/Facelets.

  Your URL is not the ordinary prefix for this (something like "/ faces")

  Your web.xml has parameters like:

  Faces Servlet

  faces *.

  And

  Faces Servlet

  javax.faces.webapp.FacesServlet

  1