TC7.2.1 problems with SIP calls

Similar Questions

• 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.

• satellite pro a60 problem with game called "call of duty".

  I have a satellite pro a60 with skvideo ati radeon 7000 igp. I have the problem with the game "call of duty" because it is too slow. With my old laptop with ati igp 340 skvideo is not this problem.
  ATI have solved this problem with the new driver, but for this toshiba is not new driver, why?

  Anyone know where I can find 'new post' I did not know where to click to make a new post!

• Problem with video calling

  Dear community of Skype,
  
  These days, I have a weird problem with my Skype. I want to use the video calling option but my video is all messed up. So you can see what is my problem, I have included a screenshot. In the screenshot its my webcam that mess, but when I call a friend its their video that looks like this. I have no idea how to solve this probem and hope you can help. I reinstalled Skype already and installed the latest version but the problem remains. I also tested my webcam with different chat like Facebook video chat programs, but there is no problem.

  I hope that my problem will be solved because I really like using Skype!
  
  PS I apologize for some grammar mistakes. English is not my native language

  

  What is the name/model of your version of map and graphics driver?

  If your graphics card is nVidia, then see this:

  http://community.Skype.com/T5/video-on-Windows/video-reception-not-right/m-p/1447456#M11141

  http://community.Skype.com/T5/Windows-desktop-client/webcam-problem/m-p/1515620/highlight/true#M1289...

• Problem with video calling on Android...

  Everything works fine but the video call is very dark on my android 4.1.2. Either way, my front cam is 1.3 mega pixels, but the same problem with my camera back which is 8.0 Mega Pixels... but when I open my camera its light show app. In this way I hate this... Please solve the problem of video calling...

  qa60
  Check your internet connection or check your microphone or try again with handfree...

• 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.

• Anyone having problems with stacked calls?

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

  All flows into that?

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

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

• 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

• (Closed) Problem with Advanced call

  My wife and I have the turbo and love. I had to disable the function call of HD on my wife's phone because she was unable to make or receive calls. He begins to give her problems about 5 days ago. Before that, she had no problem. Someone at - it and ideas or suggestions for this problem?

  We have other threads with the same question. Short holiday off HD calls. The Association knows there is a problem and has several tickets on this subject. It comes out before you are ready from current principal. I'd wait until what an update just before I turn it back on.

• Problem with asynchronous call and Forget.vi and MessageBoxW (user32.dll)

  I have a problem.  I want to use the same type of structure as in "asynchronous call and Forget.vi.

  

  There is a picture of my (Message Box.vi) VI.

  

  The VI expect the 'narrow reference' I select OK or cancel.  This is not the expected behavior.  If I turn off the "narrow" reference I have the expected behavior (by renaming properly the buttons).

  

  What am I doing wrong with the asynchronous call?

  Looks like close reference wants the loop of the root and your dialogue it blocks until it is finished.   I assumed that Run in the user interface thread is selected in the COLD LAKE to the MessageBoxW function, try changing to run in any thread

• Have a problem with the call on the computer running slow

  I get a lot of calls, saying: my computer is slow and they need to fix the person said

  that windows has reported errors on my PC
  Is it a scam or real
  Thank you

  Hello

  It's a SCAM

  they want or money on your part for programs without value or make an identity theft on you to get even more money and get your internet banking passwords and credit card information, you get allows you to download the software to 'fix' or 'test' for your computer

  read this:

  http://www.Microsoft.com/security/online-privacy/msName.aspx

  Microsoft has none of the unsolicited telephone calls to help you fix your computer

  In this type of scam cybercriminals are calling you and claim to be of Support technique Microsoft. They offer help with your computer problems. Once scammers have earned your trust, they try to steal and damage your computer with malicious software, including viruses and spyware.

  Although the law enforcement can trace phone numbers, often authors use pay telephones, disposable cell phones or stolen cellular phone numbers. Better avoid fooling themselves rather than try to repair the damage afterwards.

  Treat all unsolicited sceptically telephone calls. Don't provide personal information.

  If you receive an unsolicited call from someone who claims to be from Technical Support Microsoft, hang up. We do not have such calls.

  If you think you might be a victim of fraud, you can report it. For more information, see: what to do if you think you have been scammed.

• SPA 3201 - problem with incoming calls

  Hello

  I installed a new SPA 3102 connected to a mini Server asterisk; catches of phone line to connect to the existing line of the Earth and the phone.
  The unit has current firmware, and the one and only ethernet cable taken in the WAN port - these two steps make a working direction. I can dial a number on the phone, the call goes through the asterisk and goes out to the PSTN.
  The other way, however, does not work: I get syslog entries that the call is detected, but the device doesn't send what anyone on the server (as checked with wireshark)
  The dial plan is
  (S0<:1234>)
  but I also tried
  (S0<>[email protected] / * />)
  and some variations more

  This is syslog entries:

  FXO:start CNDD
  Number of the caller analysis = callingnumber
  -Caller ID:
  -Name = (null)
  -Number distance = callingnumber
  -Dialable number = (null)
  -No reason number = (null)
  FXO:CNDD = name, number is callingnumber
  FXO:stop CNDD
  Phone = FXO:CNDD name = callingnumber
  Your RTC AUD:Stop
  FXO: on the hook
  Your RTC AUD:Stop
  the next sequence is repeated several times, probably as long as the line is actually ringing
  FXO:start CNDD
  Your RTC AUD:Stop
  FXO: on the hook
  Your RTC AUD:Stop
  FXO:stop CNDD

  Where would I look next?

  I found a tip in an ongoing discussion, and in fact this has solved the problem: "PSTN ring timeout" must be longer than the time to ring + break from the ring, and "Time of response to PSTN" must be short enough

  Unfortunately, if incomplete description of configuration. There are so many dial plans to set, but it has not specified that have configured it.

  Please read before RTC call the SPA3102 to VOIP. I hope this will help you.

• Problem with javascript call

  Hi all

  I created a javascript function to show and hide a couple of divs as follows, it worked when I used radio buttons, but I had to exchange them for pictures and nothing happens

  < script type = "text/javascript" >

  function gcg_radio_check (form_type) {}

  If (form_type == 'pro') {}

  document.getElementById("devis").style.display = 'block ';

  document.getElementById("devis_part").style.display = 'none ';

  }

  If (form_type == 'part') {}

  document.getElementById("devis_part").style.display = 'block ';

  document.getElementById("devis").style.display = 'none ';

  }

  }

  < /script >

  I'm calling him but he does nothing... Here is the HTML code. Help, please!

  < a href = "javascript:gcg_radio_check (value); "name ="gcg_formtype"class = 'pro' value = 'pro' > < / a >

  < a href = "javascript:gcg_radio_check (value); "name ="gcg_formtype"class = 'part' value = 'part' > < / a >

  Thank you very much

  Martin

  OK problem solved, it was not recognizing the value, with the following HTML code, it works: -.

  Thank you to those who have watched

• in IE progress bar problem with on call for application

  I have several on the enforcement of the application process that I have called from javascript. The application process call pl/sql procedures in turn. Some of the procedures can take up to 10 seconds to complete, and while they are running, the screen just to freeze so that the user does not know that they are actually running. So, I would add some kind of a progression or Hourglass bar or an indication that things are moving.
  
  So I found the instructions of Re: hourglass display when the Page is processing . This works fine in Firefox but not in Internet Explorer.
  
  My footer text:
  
  < Style > #AjaxLoading {padding: 5px; do-size: 18px; width: 200px; text-align: center; left: 50%; top: 20%; position: absolute; border: 2px solid #666; background-color: #FFF ;}}
  < / style >
  < div id = "AjaxLoading" style = "" display: none; ">..." Treatment...
  < img src = "" #IMAGE_PREFIX #processing3.gif "id ="wait"/ >"
  < / div >
  
  My javascript:
  
  function create_invoice (pMilestoneNumber) {}
  Okay var = ("are confirm you sure you want to charge this invoice?");
  get var = new htmldb_Get (null, $v ('pFlowId'), 'APPLICATION_PROCESS is P110_CREATE_INVOICE', $v ('pFlowStepId'));
  If (! OK)
  return;
  html_ShowElement ('AjaxLoading');
  get.addParam('x01',pMilestoneNumber);
  gReturn = get.get ();
  Alert (gReturn);
  get = null;
  doSubmit();
  }
  
  In Firefox, the progress bar is displayed immediately after the user clicks ok to the confirmation request. In Internet Explorer, it does not appear until the application process of return with the return message.
  
  Any help would be greatly appreciated.

  Hello

  OK, it wasn't - it is just because there was a bug in my process it seemed that it worked. Sorry end that.

  So I re-tested, and the same problem that you experience will occur if you use Chrome. I think the best way is to use the technique suggested in the original thread that you connected. I so like this:

  function AjaxTest(){
       if(confirm("Are you sure you want to bill this Invoice?")){
            html_ShowElement('AjaxLoading');
            var get = new htmldb_Get(null,$v('pFlowId'),'APPLICATION_PROCESS=IE_TEST',$v('pFlowStepId'));
            gReturn = get.GetAsync(g_AsyncReturn);
       }
  }
  
  function g_AsyncReturn(){
       if(p.readyState == 4){
            alert(p.responseText);
       }else{
            return false;
       }
  }
  

  That seems to work for me.

  In addition, you will notice in the example page, it shows the AjaxLoading element when loan State == 1, but in my experience, it doesn't work well not with chrome, so who's right for which I left out of the call back and just show before the process begins.

  Van
  Trent

• Problems with incoming calls not being recognized

  I have a motorazr2 V9m.

  Can I use any phone and call my motorazr2 V9m.

  The call will not be displayed on my motorazr2 V9m.

  On the phone, I'm calling from the call will ring until voice mail rises.

  A voice message on the motorazr2 V9m notification will pop up sometimes immediately or later.

  The voicemail notification pop up of time varies. Sometimes it will not show up until you try to make an outgoing call.

  I can call the other phone several times and the motorazr2 V9m will not display a call.

  I'll try 5 or 6 times before the motorazr2 V9m will show and ringing an incoming call.

  I tried this in many places with the same result.

  Any suggestions?

  Thank you

  Looks like a reset did the trick.