Overall CVP: SIP closing Service

Similar Questions

• The overall transaction (GTT) and service channel timeout (TPETIME)

  Hi all
  
  And thanks in advance.
  
  * Tuxedo 8.1 and 10 on Solaris, AIX, and Linux.
  
  Consider the following scenario:
  
  3 services are involved S1, S2, S3.
  
  TRANSCENDING = Y (for all services)
  
  S1 - visits from its TMQFORWARD service (/ Q) then call S2 with TPNOBLOCK.
  S2 - receive calls for S1 service, then call S3 with TPNOTRAN.
  S3 - times out.
  
  Example:
  / Q > TMQFORWARD-> S1-> tpcall (TPNOBLOCK)-> S2-> tpcall (TPNOTRAN)-> S3 (timeout)
  
  In this scenario, TMQFORWARD is the beginner of the overall transaction and therefore, it is responsible of committing or rolling back according to the (tpreturn) return code, it works very well.
  I.e. back code path is:->->-> TMQFORWARD S1 S2 S3
  
  But, on the case of the time-out period, the order is backwards:
  That is the timeout order is: TMQFORWARD then S1 and S2 and S3.
  
  Our question is whether TMQFORWARD is also responsible for the abandonment of the transaction for all services of string in the case of timeout?
  
  Or if each chain services must call abandon for the case of timeout?
  
  Warning in the ULOG:
  LIBTUX_CAT:1397: WARN: failure of tpreturn transaction processing
  
  Thank you very much
  Tech dims

  Hello

  Once a Tuxedo transaction times out, no more calls on behalf of this transaction will succeed. Also, if TMQFORWARD has launched the operation, it is the responsibility of the appellant tpcommit() or tpabort() and will do.

  Regarding who should call tpabort(), only the initiator of the transaction may call tpabort(). So if TMQFORWARD was set to launch the operation, none of the services he needs to call tpabort(). They will receive an TPEPROTO error if they try to abandon the transaction as Tuxedo requires the semantics of transaction enabled, which means that only the initiator can end (successfully or unsuccessfully) the transaction.

  Also, I'm not sure what you mean when you say that the order is backwards?

  Kind regards
  Todd little
  Chief Architect of Oracle Tuxedo

• Cisco closed service 'Free' JabberVideo

  Says that this 'free' service of JabberVideo's going to be close on 16 may 2015, notification will be sent to all users of jabber.com February 16. We talk about migration to Jabber comments and/or cmr cloud/hybrid. Guess we'll find out the details Monday.

  /Jens

  Please note the answers and score the questions as "answered" as appropriate.

  Limited to occur. Wouldn't make sense to keep assets woth then drop them other Jabber Video products (Jabber Video telepresence) and Jabber video for iPad in favor of CUCM Jabber or Jabber comments.

• Agent query CVP/UCCE 10.5 Whisper - Whisper quick read failed

  Hello Cisco community support,

  We are currently experiencing problems with Agent Whisper, it's something we want to implement. At present when us dial in a script we headed over to the agent without a prompt whispers playing, we followed the official guide on the Cisco site and are currently receiving the following error message in the SVC call logs from the server during a call in the UCCE script (# were replaced when the IPS as stated due to concerns IS) :

  87768869: #. ###. ##. #: 2 August 2016 09:05:11.360 + 0100: % CVP_10_5_SIP-3-SIP_ERROR_SNMP: CALLGUID = A97FB1F357BE11E6B478881DFCAC6500 LEGID = A97FB1F357BE11E6B478881DFCAC6500-14701251111571276 - [OUTGOING] failed to read quick Whisper. Check the media name and address. VXML log bridge can provide more details with "err deb voip appl" trace on. [id: 5010]

  On the bridge, we have the following relevant commands configured:

  Gateway Gateway/VXML penetration

  Host mediaserver IP #. ###. ##. ###

  mediaserver-backup host IP #. ###. ##. ###

  !

  Dial-peer voice voip 919191

  CVP SIP ringtone description

  ring service

  session protocol sipv2

  entering the number called 919191T

  voice-class sip rel1xx disable

  DTMF-relay rtp - nte

  Codec g711ulaw

  No vad

  !

  On the console of the CVP Ops we have configured the following:

  CVP OPs Console

  Ringtone 91919191

  Local name Static Route IP address/host/Server Group name: VXML. #. co.uk

  Send calls to the originator

  Whisper 9191919100 announced

  Local name Static Route IP address/host/Server Group name: VXML. #. co.uk

  Send calls to the originator

  Script of the ICM

  Queue skills was also set to "activate target Requery.

  The WAV file was spelled correctly, when we put the wav file to a device variable 9 and PM - 9 as the audio file plays fine meaning that the codec is certainly appropriately, as gateway plays the message the value of the post to run the Script. However, when we just run the Script of post PM value and Call.WhisperAnnounment the call just are routed through and without murmur is played.

  Can anyone tell what else could be the cause this error message?

  If you want something else, then please let me know.

  Kind regards

  Christopher Hawkins

  OK, so if 'My term' does not work then you might want to try the "logging console command.

  but I do not see what I wanted,

  /en_gb/app/Whisper.wav - 80 - #. ###. ##. # cisco-IOS-C3900e/15.3 404 0 2 0

  The 404 indicates that the file is not available at the location defined in media server.

  The very basic test you can do is, open the browser on the machine where you can join the media server and type below URL.

  http:///en_gb/app/whisper.wav

  and if you see the 404 error, most probably the file is missing in app/en_gb/site, or maybe you're trying to find bad file.

  one last thing, en_gb (trait underscore between fr and gb) will not be supported in a later version of the CVP.

  It should be replaced by en-gb(en hyphen gb).

  therefore ensure that the locale folder CVP could be defined as en - us, but you're trying to get en_US.

  concerning

  Colinet

• Autonomous CVP to full cvp

  Hello guys,.

  I have a question, I have a stand-alone CVP VXML Self Service and I need to integrate the CVP UCCE as SIP CVP full, which is necessary to change the configuration? Need reinstall the cvp Server?

  Can anyone suggest me how I can switch to this new scenario with the least impact?

  Thank you

  Not a not too post this question on the Forum from the developer CVP? I replied very briefly here.

  You don't have to reinstall - that depends on how you ran Setup initially. If you have installed the call Server and ORM layer when you installed CVP VXML, you have everything there.

  I made an installation year last for a customer. They initially wanted full CVP, including a financial application self-service talk to SAP. But then they wanted to wait for ICM 8.5 (2) W2008 so the project has been split into 2 phases of deployment - the application self-service as a Standalone CVP and then later the CVP contact center supplements (SIP).

  Because I knew a normal full installation came I just installed according to the normal and added the servers VXML itself (not 'stand-alone') to the device Console Ops map. The stand-alone application was developed and deployed to production. A few months later, the contact center was built with full CVP (SIP).

  The stand-alone application remains in place. You can mix them if you wish. Peer option selector the company.

  The only thing you can change if you have added servers VXML device Console Ops such as 'Stand-alone' card, you can remove them and add them again as usual servers VXML.

  I can't say for sure, you'll have everything you need. It depends on how the initial installation was made. Could have checked a couple of boxes that you leave normally set.

  Kind regards

  Geoff

• PEI - SIP - CME - SIP - error CUCM Media is not Acceptable

  Hello world

  I have a problem with a TRUNK of SIP ITSP, the question is apparently "SIP/2.0 488 not acceptable media.

  I tried several things, I Don t know how to solve this problem.

  Outgoing calls is already OK, the problem is with incoming calls: of the ITSP to the CUCM.

  I have this topology of the ITSP SIP TRUNK:

  ITSP - sip sip - CME - CUCM

  The CME configuration is:

  Dial-peer voice voip 67
  Description * SIP trunk ITSP *.
  destination-model 591 [67]...
  session protocol sipv2
  session target ipv4:172.17.0.13
  session udp transport
  voice-class sip forced early offer
  no interaction of dtmf
  Codec g711ulaw

  !

  Dial-peer voice voip 68
  Description * SIP trunk CUCM *.
  reply-to address. T
  session protocol sipv2
  session target ipv4:172.16.6.3
  voice-class sip forced early offer
  Codec g711ulaw

  !

  SIP - ua
  Disable-early-media 180
  connection-reuse

  !

  voip phone service
  list of approved IP addresses
  IPv4 0.0.0.0
  IPv4 0.0.0.0 0.0.0.0
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  no service additional h450.7
  no additional service moved temporarily sip
  no service additional sip refer
  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
  SIP
  binding control source-interface GigabitEthernet0/1.5
  bind media source-interface GigabitEthernet0/1.5
  Registrar Server
  offer-early forced

  !

  On the side of CUCM:

  End point of media (checked)
  Disable the media beginning on 180 (unchecked)
  Requires the idle exchange of SDP for call Media Change (checked)
  Early support for voice and video calls (checked)
  Send send-receive SDP appealed INVITES (checked)

  The result of "debug messages ccsip" and "debug dialpeer inout voice" is:

  001062: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  GUEST sip:[email protected]/ * /: 5060; user = phone SIP/2.0
  Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
  Call ID: [email protected]/ * /.
  From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
  To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >
  CSeq: 1 INVITE
  Max-Forwards: 69
  Contact:
  Allow: INVITE, ACK, OPTIONS, CANCEL, INFO, BYE PRACK, NOTIFY, MESSAGE, UPDATE
  P - asserted-Identity has:
  Supported: 100rel, histinfo, prerequisite
  P-early-Media: support
  Content-Length: 362
  Content-Type: application/sdp

  v = 0
  o = HuaweiSoftx3000 1102026905 1102026906 IN IP4 172.17.0.11
  s = SipCall
  c = IN IP4 172.17.0.11
  t = 0 0
  m = audio RTP/AVP 8 18 116 10386
  a = rtpmap:8 PCMA/8000
  a G729/8000 rtpmap:18 =
  a rtpmap:116 telephone-event/8000 =
  a = ptime:5
  a = sendrecv local curr:qos
  a = distance zero curr:qos
  a = sendrecv local compulsory are: qos
  a = sendrecv distance optional with: qos
  a = 3gOoBTC

  001063: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Number = 69200020, called number = 69200020, Peer Info Type = DIALPEER_INFO_SPEECH
  001064: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Associate the rule of = DP_MATCH_DEST; Called number = 69200020
  001065: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  No outbound dial-peer does not; Result = NO_MATCH(-1)
  001066: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring = 69200020, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1
  001067: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
  Result = NO_MATCH(-1)
  001068: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Number = 70965999, called number =, Voice-Interface = 0 x 0.
  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
  Peer Type Info = DIALPEER_INFO_SPEECH
  001069: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
  001070: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
  001071: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Number = 70965999, called number =, Voice-Interface = 0 x 0.
  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
  Peer Type Info = DIALPEER_INFO_SPEECH
  001072: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
  001073: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
  001074: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
  Number = 70965999, called number = 69200020, Voice-Interface = 0 x 0.
  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
  Peer Type Info = DIALPEER_INFO_SPEECH
  001075: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
  Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
  001076: 03:31:03: //-1/CD12136099D5/DPM/dpMatchSafModulePlugin:
  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
  SIP: Attempt to analyze the attribute not supported at the level of the media
  001077: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP/2.0 488 Media is not Acceptable
  Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
  From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
  To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
  Date: Thu, November 6, 2014 13:01:04 GMT
  Call ID: [email protected]/ * /.
  CSeq: 1 INVITE
  Allow-events: telephone-event
  Reason: Q.850; cause = 65
  Server: Cisco-SIPGateway/IOS-15.3.3.M2
  Content-Length: 0

  001078: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:

  SIP GatewayTelf_JD #ACK:[email protected]/ * /: 5060; user = phone SIP/2.0
  Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
  CSeq: 1 ACK
  Call ID: [email protected]/ * /.
  From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
  To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
  Max-Forwards: 69
  Content-Length: 0

  Little light at the end of the tunnel?

  Thanks in advance!

  Hello

  can you collect debug voice ccapi inout & debugging ccsip GCE message and attach the logs here please?

  your provider sends A Law G711 codec in the PROMPT message, but you have configured G711 U right in the dial-peers.

  can you try to fix G711 has the right dial-peers and check out them? Also make sure you have TPMS in the MRGL applied to CUCM SIP Trunk.

• Global closing screen problem

  Hey,.

  My app sends a global event to trigger a global screen (pushed with invokeLater).

  I have 2 problems:

  1. the overall form is closed
  2. A screen of Dialog.inform

  More details:

  1. When I close the app with my button 'Cancel' or by the method close() disappears from the screen and then appears again
  2. When I am pressing on the button 'Add' he runs the addContact() method and adds the touch, even if she never shows the Dialog.inform (calling + "is now a contact")
     It is also close to the global screen and shows once again that when I am using the Cancel button...

  Here is my code:

   

  public class AddToContactsScreen extends MainScreen implements FieldChangeListener
  {
      private final int LBL_COLOR = Color.LIGHTSKYBLUE;
      private final int BG_COLOR = Color.BLACK;
      Caller caller;
  
      VerticalFieldManager vfm;
  
      /**
       *
       */
      public AddToContactsScreen( Caller caller )
      {
          super(MainScreen.VERTICAL_SCROLL | MainScreen.VERTICAL_SCROLLBAR);
  
          setBackground(BackgroundFactory.createSolidTransparentBackground(BG_COLOR, 200));
          this.caller = new Caller(caller);
          initFields();
      }
  
      private void initFields()
      {
          VerticalFieldManager vfmCentre = new VerticalFieldManager(VerticalFieldManager.USE_ALL_WIDTH);
  
          String lblTxt = "Add " + caller + " to your contacts?"; 
  
          LabelField lbl = new LabelField( lblTxt , LabelField.USE_ALL_WIDTH | DrawStyle.HCENTER )
          {
              public void paint(Graphics g)
              {
                  g.setColor( LBL_COLOR );
                  super.paint(g);
              }
          };
          lbl.setFont( Font.getDefault().derive(Font.BOLD, 30) );
          ButtonField addBtn = new ButtonField("Add", ButtonField.FIELD_HCENTER);
          ButtonField cancelBtn = new ButtonField("Cancel", ButtonField.FIELD_HCENTER);
  
          /* Add button's listeners */
          addBtn.setCookie( new String("addBtn") );
          cancelBtn.setCookie( new String("cancelBtn") );
          addBtn.setChangeListener(this);
          cancelBtn.setChangeListener(this);
          /* -- */
  
          vfmCentre.add(lbl);
          vfmCentre.add(addBtn);
          vfmCentre.add(cancelBtn);
  
          this.add(vfmCentre);
      }
  
      protected void paint(Graphics graphics)
      {
          super.subpaint(graphics);
      }
  
      public void fieldChanged(Field field, int context)
      {
          if( field.getCookie().equals("addBtn") )
          {
              addContact();
          }
          else if( field.getCookie().equals("cancelBtn") )
          {
              close();
          }
      }
  
      private void addContact()
      {
          try
          {
              BlackBerryContactList contactList = (BlackBerryContactList)
              PIM.getInstance().openPIMList(PIM.CONTACT_LIST, PIM.WRITE_ONLY);
  
              BlackBerryContact contact = (BlackBerryContact) contactList.createContact();
  
              String[] name = new String[ contactList.stringArraySize( BlackBerryContact.NAME ) ];
              name[BlackBerryContact.NAME_GIVEN] = caller.getFirstName();
              name[BlackBerryContact.NAME_FAMILY] = caller.getLastName();
              contact.addStringArray(BlackBerryContact.NAME, BlackBerryContact.ATTR_NONE, name);
              contact.addString(BlackBerryContact.TEL, BlackBerryContact.ATTR_WORK, caller.getNumber() );
  
              contact.commit();
          }
          catch (PIMException e)
          {
              e.printStackTrace();
          }
  
          Dialog.inform( caller + " is now a contact" );
          close();
      }  
  
      public boolean onClose()
      {
          UiApplication.getUiApplication().popScreen( getScreen() );
          return true;
      }
  }
  

  Any ideas?

  Thank you

  Rotem

  Check that if you only add the earpiece of the phone once (and unregister it properly on shutdown), it is possible that the former listeners remain.

  If this does not help, you will need to check for the event must be unique, for example by ignoring an event with the same data.

• Node ReqICMLabel failing to do apply for a standalone app than ICM CVP course

  Hello. I have a stand-alone CVP application that contains a node of reqICMLabel that seems to be failing to apply for road running. I tested two scenarios; in the first, I provide a dnis in the second, I'm leaving the unspecified dnis and have built a number dialed "NA" and it associated with the routing for my CVP customer. In both cases I followed the VRU PIM and not seeing any routes sent queries.

  When the DNIS is specified, the node reqICMLabel seems not to recognize the dnis, even if it is defined in the activity log. These logs are here:

  2.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:24.387, start, newcall,.
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014, 15:02:24.387, ani, NA
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014, 15:02:24.387, postcode, NA
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:24.387, start, Exchange, NA
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:24.387,Start, dnis, ewt
  15:02:24.387, early, Karim 10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014, NA
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:24.387, start, iidigits, NA
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:24.387,Start, settings, _dnis = ewt
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:24.387, CVP Subdialog Start_01, enter,
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:31.887, CVP Subdialog Start_01, exit, completed
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:31.887, ReqICMLabel_01, enter,
  15:02:31.887, ReqICMLabel_01, 10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014, doDecision, custom- the callGuid or DNIS is null.
  15:02:31.887, ReqICMLabel_01, 10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014, error output
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:02:31.887, Copy_of_Audio_01, enter,
  Output of 15:03:15.356, Copy_of_Audio_01, 10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014,
  10.122.23.189.1406152944387.5.VIVR_RBS,07/23/2014 15:03:15.356, end, how, hang up

  In the second scenario when the DNIS is not specified reqICMLabel node fails with a null pointer exception:

  2.23.189.1406153843968.11.VIVR_RBS,07/23/2014 15:17:23.968, start, newcall,.
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014, 15:17:23.968, ani, NA
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014, 15:17:23.968, postcode, NA
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014 15:17:23.968, start, Exchange, NA
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014, 15:17:23.968, dnis, NA
  15:17:23.968, early, Karim 10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014, NA
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014 15:17:23.968, start, iidigits, NA
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014 15:17:23.968, CVP Subdialog Start_01, enter,
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014 15:17:27.765, CVP Subdialog Start_01, exit, completed
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014 15:17:27.765, ReqICMLabel_01, enter,
  15:17:27.765, ReqICMLabel_01, 10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014, error, a custom element element has encountered an exception.
  Output of 15:17:27.765, ReqICMLabel_01, 10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014,
  10.122.23.189.1406153843968.11.VIVR_RBS,07/23/2014, 15:17:27.765, error,.

  My VRU PIM is in place and the customer routing is active. I use VRU type 10 for this. Wondering if this should be a different type of VRUS? -Update, just found some documentation that does not specify any VRU network must be configured for stand-alone CVP PIMs, so I guess that's not the issue.

  Thank you!

  Brad

  your VXML server is a stand-alone?

  using CVP tcl self-service for this application?

  can you post your call logs from the server for that?

  Kind regards

  Colinet

• SIP Dialer with voice gateway using the FXO Ports

  Hello

  We have a laboratory UCCE consisting of an outgoing SIP dialer and a h323 2811 voice modem router with 2 FXO ports. The Dialer reserve an agent and sending the call to the bridge, but the call fails immediately with the bridge return a "404 not found" error to the Dialer. Call directly from a CUCM extension attempts to connect successfully through the FXO. Any thoughts?

  IOS Ver - T4 15.1 (3)

  UCCE worm - 8.5.4

  The gateway configuration:

  version 15.1
  horodateurs service debug datetime msec
  Log service timestamps datetime msec
  no password encryption service
  sequence numbers service
  !
  hostname LAB-RS-ING_VXML-GW
  !
  boot-start-marker
  boot-end-marker
  !
  !
  logging buffered 50000000
  no logging monitor

  !
  No aaa new-model
  no location network-clock-participate 1
  !
  voice-card 0
  !
  voice-card 1
  dspfarm
  !
  dot11 syslog
  IP source-route
  !
  !
  IP cef
  !
  !
  !
  IP host mediaserver 10.15.80.83
  No ipv6 cef
  Authenticated MultiLink bundle-name Panel
  !
  !
  !
  !
  !
  !
  Circuits-port FXO group
  hunting-sequential diagram
  !
  !
  !
  voip phone service
  No IP trust to authenticate
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  no service additional sip refer
  service additional media - renegotiate
  signs before any
  H323
  Modem passthrough codec g711ulaw nse
  SIP
  interface FastEthernet0/0 source control binding
  bind media source interface FastEthernet0/0
  header-passage
  !
  voice class codec 1
  g711ulaw codec preference 1
  preferably 5 codec g729r8
  !
  vocal h323 class 1
  H225 timeout tcp establish 3
  !
  !
  !
  !
  translation-article 99 of the voice
  rule 1 / ^ 4 / /334/
  !
  !
  voice translation-profile PROFILE_TO_CVP
  translate called 99
  !
  !
  Crypto pki token removal timeout default 0
  !
  !
  !
  !
  Archives
  The config log
  hidekeys
  !
  !
  !
  !
  !
  !
  !
  interface Loopback0
  255.255.255.255 IP address 172.20.1.1
  !
  interface FastEthernet0/0
  IP 255.255.255.0
  automatic duplex
  automatic speed
  !
  interface FastEthernet0/1
  IP 255.255.255.0
  Shutdown
  automatic duplex
  automatic speed
  !
  !
  Router eigrp 15
  10.0.0.0 network
  Auto-resume
  !
  IP forward-Protocol ND
  !
  no ip address of the http server
  no ip http secure server
  !
  IP route 0.0.0.0 0.0.0.0 10.x.x.x
  !
  !
  !
  !
  control plan
  !
  !
  voice-port 0/0/0
  Ports FXO 1 trunk group
  surveillance cut dualtone Mid-communication
  output attenuation - 3
  no echo - cancel enable
  No non-linear
  No vad
  broadcast-maximum: 250
  broadcast / nominal 200
  minimum of playout / high-delay
  broadcast-delay mode fixed
  call waiting times - disconnect 5
  timeouts wait-version 5
  connection ÉRA 4930
  4930 description
  !
  voice-port 1/0/0
  2 ports FXO trunk-group
  surveillance cut dualtone Mid-communication
  output attenuation - 3
  no echo - cancel enable
  No non-linear
  No vad
  broadcast-maximum: 250
  broadcast / nominal 200
  minimum of playout / high-delay
  broadcast-delay mode fixed
  call waiting times - disconnect 5
  timeouts wait-version 5
  connection ÉRA 4198
  4198 description
  !
  voice-port 0/0/2
  !
  voice-port 0/0/3
  !
  !
  !
  profile MGCP default
  !
  !
  voice POTS dial-peer 1
  incoming called-number.
  direct line to inside
  !
  Dial-peer voice 2 voip
  session protocol sipv2
  incoming called-number.
  codec voice-class 1
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  Dial-peer voice voip 4198
  destination-model 4198
  session target ipv4:10.15.80.81
  codec voice-class 1
  h323 voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  voice pots Dial-peer 301
  trunkgroup FXO Ports
  Description * MIA local calls *.
  destination-model 305 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  prefix 305
  !
  Dial-peer voice voip 4930
  destination-model 4930
  session target ipv4:10.15.80.81
  codec voice-class 1
  h323 voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  Dial-peer voice voip 7000
  Description of Agents
  destination-model 7...
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  !
  Dial-peer voice voip 9999
  Phone agent test description
  destination-model 9999
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  Dial-peer voice voip 1444
  Phone agent test description
  destination-model 1444
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  voice pots Dial-peer 303
  trunkgroup FXO Ports
  Description * LD calls *.
  destination-pattern 1 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  Forward-digit 11
  1 prefix
  !
  voice pots Dial-peer 302
  trunkgroup FXO Ports
  Description * MIA local calls *.
  destination-model 786 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  prefix 786
  !
  voice pots Dial-peer 9
  destination-model 91 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  port 0/0/0
  Forward-digit 11
  !
  !
  Dial-peer voice 1400 voip
  Description of Agents
  destination-model 14...
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  !
  !
  !
  !
  Line con 0
  line to 0
  line vty 0 4
  opening of session
  transport of entry all
  !
  No Scheduler allocate
  end

  Do you have installation trunks in call to the dialer Manager?

• Time remapping key frames cause other keyframes become independent to where they are on the timeline

  This is the BUG report, in that I just sent. If anyone has anything to add.

  ******BUG******
  Concise problem statement:
  When I add the time remap keyframes to a clip, other keyframes become unrelated to the part of the timeline, on what they are.

  Steps to reproduce the bug:
  1. add TIME REMAP keyframes in a MONTAGE WITH of OTHER keyframes (AS the POSITIONING of KEY frames)
  2.
  3.

  Results: I HAVE CHANGE THE COORDINATES OF UP/DOWN AND THE CLIP DOES NOT MOVE YET, SO I RUB ALONG THE CLAMP AND IT MOVES UPWARDS OR DOWNWARDS TO A COMPLETELY DIFFERENT OF THE CLIP PART. AS SOON AS I HAVE TIME REMAPPING IT SCREWS TO THE TOP OF THE FEATURE OF THE EFFECT CONTROLS PANEL.

  Expected results: READ MORE TOP! TIME REMAPPING HAS BEEN BROKEN SINCE CS4 AS AS FAR AS I KNOW AND NOTHING HAS BEEN DONE ABOUT IT...  SO FRUSTRATING!

  I have always worked around him or avoided using the selective time remapping effect but I've had enough of him... It's like the ability to change the color of the fonts on this forum, non-functional (the last time I tried to use it), but I prefer being able to be editing as colorful letters time remapping, I paid for the time remapping in CS4 and CS5 effect.

  Peter

  > Yes I have feature request/Bug report page my favorites now, but I have little confidence in their sincere care or intent. Having recently learned that my upgrade to Production Premium CS5 made me eligible for a 'free advantage', I tried to make the request. I followed the links, there were two, and they both took me to the same page, a page that had NO top at all links (This Page here). I opened a 'customer care case' and they told me to go down that same path and choose my 'delivery '.

  Bug reports and feature requests go directly to the engineering basis for Premiere Pro team. People who treat and read these reports are right next to the Director of engineering, product manager and so on. I am the person who treats these reports for After Effects, and I work closely with the people who do it for Premiere Pro. These reports and applications are very important for us.

  People who treat the benefits of registration, on overall web pages, customer service, and so on... Well, they are in an entirely different group. You cannot draw conclusions on how we treat (good or bad) how bug reports are these other things.

  This is not to say that we are in the first team Pro can help get these other things fixed. I have, on the one hand, make it a point to emphasize issues like these to the appropriate people. I'll pick up the phone and call people to report problems as you mentioned here as soon as I hit submit on this response.

• I can not configure Windows messenger in Windows XP

  I have installed Windows XP system. I could not configure windows messenger to communicate with my colleagues. When I checked the tools-> Options-> tab accounts, I couldn't find the account of SIP communications service with which others have configured their Messenger.
  Any help with this problem would be greatly appreciated.

  Windows Messenger 5.1 is no longer available. You can see that if your company has only the version that you can download from all your colleagues are using this version.

  Or you can download MSN Messenger 7.0
  http://www.Microsoft.com/downloads/en/details.aspx?FamilyId=cf49c56c-8b3e-4EAE-9904-9505f47bed45

  Another choice would be to Windows Live Messenger 2011:
  http://www.Microsoft.com/downloads/en/details.aspx?FamilyId=edd9bdcf-16b0-4678-9bbd-e9b429296246

  Thank you

  Marilyn

• Windows XP wireless network

  Automatic Configuration of Windows service stops after every awakening upward. It is configured to run automatically.

  Hello

  Are you sure it is the closing service?

  What you describe is usually a problem of Power Saving of network card.

  That is, disable the wireless card power saving setting.

  If WZC by itself, then it could be that whenever when you restart there is also wireless utility from the seller that stop WZC.

  Check and disable any Manager without extra wire.

  See here, regarding the WZC http://www.ezlan.net/wzc.html configuration

  Jack - Microsoft MVP, Windows networking. WWW.EZLAN.NET

• new debug message 10.2

  out of my application, I see a new message on the console:

  Err: app: wm: destroy called when the window closed services-bkgd-0ef0f094126a4d534f7b15f66576b2c1e2cac062

  seems to appear only if the application presented a menu...

  ideas?

  It can be ignored. It's just an additional record.

• No dialing between sites, no voicemail Audio

  It is that me completely baffled.  We have 5 fully equipped uc520 total sites or 540's.  All sites between them can call on the MPLS with NO problems.  The problem is when the 1 site called site 5 and the call to your voicemail after 4 rings it is no sound on the end.  You can see it go to voice mail DN 5502 but not audio.  Now, if you call 5 DN voicemail directly rather than an extension Site you audio here on the end.  I have attached the configs for the two sites.

  I have attached the configs to Site 1 and site 5.  Just an info. I have CLI had the composition complete intersite.  I hope someone with more talent can see my mistake.

  I have a feeling that it is because 'voice service voip' settings do not match

  Site 1:
  voip phone service
  list of approved IP addresses
  IPv4 0.0.0.0 0.0.0.0
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  service additional h450.12
  SIP
  Registration Server expires max 600 min 60
  no update-callerid
   
  Site 5:
  voip phone service
  list of approved IP addresses
  IPv4 0.0.0.0 0.0.0.0
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  service additional h450.12
  no additional service moved temporarily sip
  no service additional sip refer
  SIP
  Registration Server expires 3600 3600 min max
  localhost dns:nnect - fca.atlo.cbeyond .net
  Proxy outgoing dns:proxy - fca.atlo.cbeyond .net
  no update-callerid
  SIP-profiles 1000

  Orders "no extra service" and the "sip-profiles 1000" could cause problems between the two systems during the transfer to the CUE."  CUE uses SIP, so these controls affect his behavior as well as the Cbeyond SIP trunk.

• Architecture Cisco UCS

  Hello world

  I am very new to the studies of data center and I am trying to build my ralted concepts Cisco's UCS. It would probably silly questions but I want to know:

  1. What is the difference between a 'profile of Service' and the virtual machine? They are two different names for the same concept?

  2. What is the concept behind the virtual switch? It is used to connect the virtual machines or Service profiles? If this is the case, it is installed on a server on a machine virtual/MS or something else?

  2. I would be grateful if someone could share a logical diagram indicating an overall concept of integrated service profile, virtual switch, VNIC, vEth, blades, etc.

  Thank you very much.

  an SP is associated with a physical blade! a SP that corresponds to a logical server.

  logical server (MS) + physical = physical server Blade (ESXi, W2012, Linux,...)