Nokia VoIP integration - wow wow - HD Voice - amaz...

I can't understand why so called VoIP-Providers don't offer this method:
http://www.youtube.com/watch?v=ClEYf0lGRAc&hd=1

Similar Messages

  • In nokia C7 integrated GPS option not found in pos...

    In nokia C7 integrated GPS option not found in positioning method
    Attachments:
    Scr000001.jpg ‏14 KB

    @shajjeel
    AFAIK this would only occur in device originally destined for sale in Syria.
    Happy to have helped forum with a Support Ratio = 42.5

  • Nokia VoIP Client using SIPS/SRTP freeze the mobil...

    I'm using a N95 Firmware: R-159 30.0.015 (its happen with N95 8G too) and i've configurated a VoIP account with SIPS URI.
    I' can make VoIP call and speak (the quality is great!) but when we hangup (me or the called) the Nokia VoIP Client did NOT disconnect and freeze the mobile(I've to turn off the device).
    I've running the wireshark in my network and see no TLS transaction has been send on hangup from NOKIA device , so I' would like Know if someone can help me.

    I figured if choose one SIP Account to run on 3G and one on WiFi the power consumption is low.
    If both are switched to 3G as access point the power consumption is high.
    Must be some kind of bug...

  • Nokia N95-3 crashes when using Voice Mailbox (yes,...

    Hi,
    As reported by previous N95 users, N95-3 (10.2.006) crashes when you use the #1 shorcut for voicemail, and as stated previously if you dial the number, or use it inside the speed dial app, or even with voice command it works.
    After researching I found that the problem starts when I configured my SIP data parameters, if I delete the SIP profile, #1 shorcut works well.
    In my previous phone (N80ie) with a SIP profile defined, when you press #1 the phone asks if you want to call your regular voice mail or your internet phone voice mail, I think that this is that crashes my N95.
    Just and acclaration, if I connect to my VoIP account, and press #1 everything works perfect with the N95 as in N80, the problem is when you are not connected to VoIP, #1 crashes.
    Nokia please make note of this for future FW update.
    Regards,

    Solved, after the first use of internet phone, even disconected of SIP service, everything works ok.

  • VOIP + Integrated Audio Conferencing + Connect 8

    Hello -
    These forums have been a valuable resource for my individual troubleshooting in Adobe products, and I thank the contributors.
    Currently I am having an issue that I am unable to resolve on my own, nor find any postings covering simlar topics through search.
    Core Question:   Can one create a room in Adobe Connect 8 that does all of the following?:
    Allows virtual attendees to connect through VOIP
    Allows virtual attendees to connect by selecting to have the integrated conference line call them (where they must enter their phone number), where that attendee can be muted individually, their audio status is displayed by their username etc.
    Allows (and provides directin for) the attendees to dial into the conference line
    Specs: 
    Currently Adobe Connect 8 (Service Pack 1)
    PGi Conferencing Service
    Adobe product support through Clarix
    Background:
    For our implementation(s), we need to broadcast many different event types and offerings to a virtual audience in which presenters/instructors and participants can share two way verbal communication with one another.  As the host, I must command control of the audio at an individual level using an integrated conferencing service (we use PGi).  VOiP users are already controlled at the individual level.  We must provide the easiest and most flexible access to the Adobe Connect room for our virtual attendees, so that they can click a link and sign on as a guest and then choose whether or not they wish to use VOIP, dial into the conference line from their phone, or ask the Adobe Connect room to dial out to their phone.
    We have successfully delivered conferences in the format that I describe using an Adobe Connect-based product, which I presume to be built on a previous version of Adobe Connect.  This version had the little yellow 'dial out' phone button.  Let me add that I am aware of and understand what Universal Voice is and how it works, and that Universal Voice will not meet our needs.  The latest and greatest Adobe Connect 8 Help PDF had this to say on page 126 about using VOiP and Conference together:
    http://help.adobe.com/en_US/connect/8.0/using/connect_8_help.pdf
         Note:  If Universal Voice is configured, you can select both options.  Participants can join the audio conference either using their computer's   
                   microphones, or join the meeting using their phone. Start broadcasting to to enable participants to hear any telephone-based audio through the
                   computer speakers and broadcast their voices to telephone users using the microphone. When you stop broadcasting, VoIP users will be audible
                   to phone users, but phone users will not be audible to the meeting users. 
    Anybody with insight please reply, and I thank you in advance.
    Warm Regards,
    :Otto:

    Otto,
    I'm baffled by this. Connect 8 does all what you desire with Universal Voice and PGi. I do all what you desire today. Connect 8 SP1 which is being rolled out changed the menu choices to make the options really clear. They were not clear in Connect 8 and many people got confused including me!
    What is it about UV and the integrated PGI adaptor that does not work for you?  Best to take this off-line I think as we will need to discuss.
    You can reach me here:  [email protected]
    Thanks.

  • Nokia, VoIP and N86

    Does the Nokia N86 have the Internet Tel App like the N95 does?
    I'm still on my Nokia N95 because we use a VoIP system at work and i have my phone connected to it when in range, the Nokia N96 does not have it nokia thought it would be "funny" to take this feature out, i believe Nokia thought it would be really funny to take it out of the Nokia N97 too, will the Nokia N86 have this so i can make VoIP calls without having to install rubbish 3rd part applications?

    quackers82 wrote:
    I downloaded SIP_VoIP_3_x_Settings_v2_0_en.sis onto my N86, still cannot make internet calls. I have no Internet .Tel Application and "Internet Call" is not an option when dialing. Most i can get is "Registered" under SIP Settings, but no application to dial or recieve.
    Am i missing somthing Amadir?
    You are just missing the actual VoIP configuration profile which must be created with the above mentioned SIP VoIP Settings plugin. Version of the plugin mentioned above should work basically on any 3rd Edt. FP2 phone other than N96 and N78 of course which don't seem to support VoIP at all.
    After you have installed the SIP VoIP Settings plugin you will have a new "Advanced VoIP Settings" module visible on the phones "Net Settings" application. I'm not sure about the default menu structure of the N86 but typically Net Settings entry can be found either under Settings -> Connectivity or it could be located directly in the "Control Panel" menu. Look around and you will find it somewhere on the N86 menus.
    Go to the Net Settings application and select Advanced VoIP Settings -> Create New Service. Either select your your existing SIP settings or define new SIP profile and you will have a new VoIP profile available to be used.
    Starting with VoIP rel. 3.x (3rd Edt. FP2) phones there is no way to manually define VoIP settings on the phone UI without installing the VoIP Settings plugin and using the it via Net Settings.
    Also a separate "Internet Telephony" application (where VoIP registration status was controlled previously) does not exist anymore but instead VoIP activation/deactivation  has been integrated in to the "Contacts" application, after you have created the VoIP settings and profile as mentioned above.
    Your VoIP service should show up as an new "Tab" (named as your VoIP settings profile) on the Contacts and from this Tab you can activate and later de-activate (Option -> deactivate service) the VoIP registration. Once VoIP registration has been successfully established you should be able to call "Internet Call" type of calls to your contacts similarly to earlier generation VoIP enabled phones (E66, E71, N95 etc.).

  • How can I use Voip to communicat with my voice in Labview

    i want to be able to call my computer using Voip. I know this service
    cost money. I dont just want to call the computer, i want to call
    labview and all labview to have access to my voice, so i can do speech
    commands on it.

    Hello,
    Thank you for contacting National Instruments. I would like to apologize for the delay in responding to your support request. I understand that you would like to use Voip capabilities to access your computer remotely and then communicate with LabVIEW using speech recognition.
    I know of a LabVIEW example file that describes the voice recognition capabilities of LabVIEW. Also, there is a link to example programs within this database entry. To access the entry, go to the National Instruments support portal: http://niweb2.natinst.com/ae/portal.htm and search for �voice recognition LabView.� Click the first entry in the list: �Voice Recognition in LabVIEW - Example - Development Library - National Instruments.� You will be able to find helpful information here.
    I hope this helps! Please let me know if I can help you further. Have a great day!
    Kind Regards,
    Joe Des Rosier
    National Instruments

  • Nokia e7 - doesn't work the voice call

    Hello! I have nokia е7 with latest OS. I can't force to work a voice call. When pronouncing a voice tag occurs nothing. In what there can be a problem. Thanks.

    Phone work under Nokia Belle Refresh 111.040.1511.
    I have 601 contact in my phone.
    I want to make some adjustment to my post. I can call by voice only if I press link on app "Voice command", then say contact name and then, after some detection phone call to the contact person. My question was - If I could just say contact name WITHOUT touching phone to call to thecontact person. (And if answer is - NOT, could I make it with Garniture). Thank you!

  • Nokia X6 Integrated GPS

    Does anyone succeed in getting Integrated GPS working on X6 purchased from Syria??? I tried many methods without success, i cannot see the Integrated GPS or Assisted GPS in  the repositioning method

    Hi
    Now i'm lost many posts on the net claim that Nokia would not remove any hardware (including GPS) and they would only disable any feature by firmware. Hence, i changed my phone product code to a UAE and many others including EUROPE and flushed it with latest firmware 21.0.0.4 but still cannot see GPS option in Repositioning methods.
    I'm looking for a confirmative answer that NO GPS MODULE IS ISTALLED IN PHONES SOLD IN SYRIA or a working solution for my challanging problem
    I did contact Nokia in North American and once they knew my phone from Syria that was not able to help me anymore ( souded very scary too them to just confirm if GPS module is there or not!)
    Thanks

  • Nokia Messaging Integration

    Hopefully i'm not posting something that has already been discussed....
    I had to reinstall the Nokia Messaging application into my phone, and when i did, instead of doing the usual setup the way it did in Beta, the Messaging integrated itself with the native application, which icorporates all of the original nokia settings and configurations.  It looks to me like becasue of that, the push option has been effectively eliminated.  Is there any way to get back to the original settings of that new messaging program?  I definitely found it better than the original nokia messaging that came on the phone.  Any help would be appreciated.

    dear bigdog, i tried what you suggested, but it didn't work out. i went to Messaging -> Option -> Setting -> Email -> Default mailbox, there is only one account called "Email". As you can see from Fig. 1 attached, when I tried to edit "Email", nothing happens, as if "Email" doesn't exist. However, I do have a Gmail account in use. Also, as you can see from Fig. 2 attached, in Messaging (the built-in one, not the installed newest one with HTML support), there is a "Mailbox" icon, meaning no email account was configured. whenever i click on it, it will ask me to type in account info and setup a new Gmail account. once i did so, but it turned out that on my home screen, there are two email clients showing info from one Gmail account (one client is built-in, the other one client is the newest "colorful-interfaced" Nokia Messaging 1.1 with HTML support). now i am using only the newest one. thanks so much for your help!
    Message Edited by michaels on 09-Apr-2009 04:41 PM
    Sorry about my large font. I have problems with my sight.
    Please give me some KUDOS if my reply is of help for you. Thank you very much!
    Attachments:
    Figure 1.jpg ‏13 KB
    Figure 2.jpg ‏17 KB

  • Looking for telephony/VoIP integration recommendation

    We are looking for a solution to integrate our telephony and VoIP.  Most of our users will utilize the VoIP but we found in our previous tool that the telephony option was nice to have as a back-up when the inevitable microphone issues arise.  However, with Adobe Connect, we're having trouble finding a solution that is truly integrated.  For example, in one scenario we've investigated, the host can't mute the people on the phone - only on VoIP.  In another scenario, a host must join the conference call in order for it to work for participants (counter-productive since our hosts are on VoIP and we'd like to keep it that way).
    Any success stories for a truly integrated solution would be helpful.  I would be interested to know the administrative duties as well (when you set up the meeting do you need to manually select and therefore keep track of when particular numbers are in use?  If so, for companies who are potentially delivering multiple classes simultaneously and across the globe - how do you manage that?).
    Thanks in advance!

    I believe that PGi or Intercall have the ability to use the fully integrated bridge and broadcast over VoIP simultaneously.
    Phone bridges must be assigned to rooms ahead of time, but both providers should happily give you as many bridges as you want/need and they are billed and tracked separately. So you would just need to assign the individual bridges to the appropriate rooms. So long as no two rooms have the same bridge, you won't have to worry about a conflict.

  • VOIP integration

    My client uses Mitel Phones (VOIP).  Wants to inquire whether these phones are able to integrate with SAP CRM. 
    What questions do we need to ask them, specific to their phone system - I am expecting we need to know the specific model and version of the phone switch and related details.  Please advise

    Hi,
    Please elaborate your query.
    Thanks
    GS

  • CRM VOIP Integration

    I have been volunteered (never take a day off work) to research a project possibility study for the company. The basics are.
    We have clients in six or so countries around the world that call a centralised support desk in one country. My questions.
    1, We currently use Wounderdesk to log client service desk calls. However, the company is migrating to Microsoft CRM 2003 & management would like to have a Cisco IPFX (existing) or if needs be another VOIP Skype type of incoming call system that pre-populates the fields in CRM? Is this possible?
    & The Company would like to have one service desk number in each country that then routes through to the service desk, avoiding the cost of long distance phone call cost to the client base! Where would I find information on this type of project?
    Anyone point me in the right direction here?
    Thanks for any and all input, I really appreciate it!

    One option to connect your CRM with IP communications is the CRM Communications Connector which works with Callmanager, Callmanager Express, and IPCC Express. See the following link.
    http://www.cisco.com/en/US/products/ps7274/products_installation_guide_chapter09186a00805658a9.html
    The call routing functionality you're describing is called Toll Bypass and can definitely be accomplished with Callmanager or Callmanager Express.
    Hope this helps. If so, please rate the post.
    Brandon

  • Asterisk-voip integration problem

    I am trying to integrate voip in my application, using asterisk.
    When I use the below code, I am able to make a call, to the extension i want to, but the problem arises when the call is taken.
    As the call is taken, it gets transferred to the other line, and all what the person gets to hear is the ringing tone, as the call is transferred to his other line.
    For example, if the user takes the cal on line 1, then as he picks up, the call goes to line 3,, and he only hears the ringtone.
    I dnt know how to deal with this, plz help
    Code attached below :
    package asterisk;
    import java.io.IOException;
    import org.asteriskjava.manager.AuthenticationFailedException;
    import org.asteriskjava.manager.ManagerConnection;
    import org.asteriskjava.manager.ManagerConnectionFactory;
    import org.asteriskjava.manager.ManagerEventListener;
    import org.asteriskjava.manager.TimeoutException;
    import org.asteriskjava.manager.action.OriginateAction;
    import org.asteriskjava.manager.action.RedirectAction;
    import org.asteriskjava.manager.action.SetVarAction;
    import org.asteriskjava.manager.action.StatusAction;
    import org.asteriskjava.manager.event.ManagerEvent;
    import org.asteriskjava.manager.response.ManagerResponse;
    public class HelloEvents implements ManagerEventListener
    private ManagerConnection managerConnection;
    public HelloEvents() throws IOException
    ManagerConnectionFactory factory = new ManagerConnectionFactory(
    "pbx.co.uk", "username", "password");
    this.managerConnection = factory.createManagerConnection();
    public void run() throws IOException, AuthenticationFailedException,
    TimeoutException, InterruptedException
    // register for events
    managerConnection.addEventListener(this);
    // connect to Asterisk and log in
    managerConnection.login();
    // request channel state
    managerConnection.sendAction(new StatusAction());
    // wait 100 seconds for events to come in
    // Thread.sleep(10000);
    OriginateAction originateAction;
    ManagerResponse originateResponse;
    originateAction = new OriginateAction();
    originateAction.setChannel("SIP/3006");
    originateAction.setContext("default");
    originateAction.setExten("3006");
    originateAction.setPriority(new Integer(1));
    originateAction.setTimeout(new Integer(5000));
    originateResponse = managerConnection.sendAction(originateAction, 30000);
    SetVarAction setVarAction = null;
    if(originateResponse.getResponse().equals("Success"))
    System.out.println("I am coming hereeeeeeeeeeeeeee");
    System.out.println("the reason for failure is=====>>"+originateResponse.getResponse().toString());
    setVarAction.setVariable("STRING3");
    setVarAction.setValue("SIP/3006");
    originateResponse = managerConnection.sendAction(setVarAction, 30000);
    // if(originateResponse.getResponse().equals("Success"))
    // RedirectAction redirectAction = new RedirectAction();
    // redirectAction.setChannel ("SIP/3000");
    // redirectAction.setContext("mariaSIP");
    // redirectAction.setExten("444");
    // redirectAction.setPriority (new
    // Integer(1));
    // originateResponse = managerConnection.sendAction (redirectAction,
    // 30000); if (originateResponse.getResponse().equals("Success"))
    // and finally log off and disconnect
    else
    System.out.println(" i am not takin callssssssssssssssss");
    managerConnection.logoff();
    public void onManagerEvent(ManagerEvent event)
    // just print received events
    System.out.println(event);
    public static void main(String[] args) throws Exception
    HelloEvents helloEvents;
    helloEvents = new HelloEvents();
    helloEvents.run();
    }

    n yeah, though I would appreciate ur help by asking me to go to an Asterisk forum or sumwhere,,it doesnt help, coz i have posted it there n i guess 100 people have read it, but no replies.
    So that forum hardly helps,,
    as a matter of fact, even here I am yet to find any replies for an "asterisk" query i put a couple of days back.
    lolzz....
    Help guys !!!!

  • 5800 Nokia mobile XpressMusic and SIP for VoIP

    Hello;
    Anyone can advise for the needed file to be downloaded for the 5800 Nokia mobile XpressMusic to be supported for SIP so I can place voice over IP calls.
    Regards
    Bilal

    Please search this forum for detailed answers to your questions.  Summary: there is no Nokia VOIP application for the 5800.  You can download Fring and Nimbuzz and connect to your VOIP provider through them.
    Lumia 920, Lumia 800
    Nokia N8-00 (NAM, Product Code: 059C8T6), Symbian Belle, Type RM-596, 111.030.0609
    Nokia 5800 XpressMusic (NAM, Product Code: 0577454) Software v51.2.007, Type RM-428

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