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  • How to set up a mini call center?

    - by Ralph
    I'm trying to figure out how to set up a miniature call center for a small business. Like, for 1-4 people to take calls, but hopefully expandable to more. We want to accept nation-wide calls, and then I guess distribute the calls among the available agents. If no one is available, I guess it should either put them in a queue and play some annoying music for them, or forward to call to an agent who has least-recently taken a call who can then quickly answer and say "please hold" until they're done their call. We want to have one phone number that customers can call. I guess we then need some kind of ACD system which would take each call and forward it to an agent based on some algorithm? Then we would need to purchase a separate phone line for each agent, plus one just for the distributor? Or do we need several "extra" lines to maintain a queue (one for each customer waiting too)? This "ACD" thing, is it just a device that you would plug a phone line into (or several?), and maybe connect to a computer, aided by some software? Or is a subscription thing that I would need from my local telephone provider? Next, the business we're running, the callers will be repeat customers. It would be helpful to automatically pull up their profile based on the incoming number. The "software" our agents will be running will just be a website where they can log in (preferably from home) and then enter some information they would obtain through the call. So, the system would have to somehow interface with the website if possible. If not, we'll just have to ask each customer for an identifier (phone number, username, customer number, or something). Is this possible? I guess each computer would need a device that the call would pass through, and then if I can somehow hook into that, then I can write some software that will interface with the site. So, where do I start? What hardware do we need to buy? What subscriptions do we need? We were thinking this magicJack might help us in accepting long-distance calls for cheaper, but my understanding is that they provide you with some weird-looking number, is there a way we could "mask" it with our toll-free number? And then pass the incoming calls through the distributor system, which would then get passed to the call-accepting device which would both allow an agent to answer the call and have a software hook? (I realize this might be partially out of the scope of SU, but I wasn't sure where else to ask. It is about computer(-aided) hardware and software anyway.) P.S.: I don't need any of that "press 1 to talk to..." or "say xyz to..." junk. Just a straight-forward, connect-to-next-available-agent system.

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  • Don’t miss this very popular presentation on Punchout in iProcurement on June 26th 2012

    - by user793553
    Don’t miss this very popular presentation on Punchout in iProcurement on June 26th.  See Doc ID 1448447.1 for the Webcast details. ADVISOR WEBCAST: Punchout in iProcurement PRODUCT FAMILY: EBZs- Procurement   June 26, 2012 at 14:00 UK / 15:00 Cairo / 6:00 am Pacific / 7:00 am Mountain / 9:00 am Eastern This one-hour session is recommended for technical and functional users who are maintaining and/or implementing the Punchout from iProcurement. The session will provide an overview of the different Punchout model, setup, and the Punchout to PO xml/cxml cycle. Also, it will provide tips in troubleshooting the common issues when new supplier is added to Punchout or the existing one stops working. TOPICS WILL INCLUDE: Overview of the Punchout Models. Provide the knowledge in the Punchout to PO Process cycle. Demo - Punchout. Certificates and setup. Learn the common issues and how to address in an efficient way. (Documentation and Notes) A short, live demonstration (only if applicable) and question and answer period will be included. Oracle Advisor Webcasts are dedicated to building your awareness around our products and services. This session does not replace offerings from Oracle Global Support Services. Current Schedule can be found on Note 740966.1 Post Presentation Recordings can be found on Note 740964.1 WebEx Conference Details Topic: Advisor Webcast - Punchout in iProcuremen Date and Time: Tuesday, June 26, 2012 3:00 pm, Egypt Time (Cairo, GMT+02:00) Tuesday, June 26, 2012 2:00 pm, GMT Summer Time (London, GMT+01:00) Tuesday, June 26, 2012 9:00 am, Eastern Daylight Time (New York, GMT-04:00) Tuesday, June 26, 2012 7:00 am, Mountain Daylight Time (Denver, GMT-06:00) Event number: 597 373 155 -------------------------------------------------------  To register for this meeting  -------------------------------------------------------  1. Event address for attendees: https://oracleaw.webex.com/oracleaw/onstage/g.php?d=597373155&t=a 2. Register for the meeting.  Once the host approves your request, you will receive a confirmation email with instructions for joining the meeting. InterCall Audio Instructions A list of Toll-Free Numbers can be found below. VOICESTREAMING IS AVAILABLE teleconference ID: 70528713 UK standard International:+44 1452 562 665 US Free Call: 1866 230 1938 US Local call: 1845 608 8023 Global Toll-Free Numbers MOS doc#:  https://metalink3.oracle.com/od/faces/secure/km/DocumentDisplay.jspx?id=1148600.1 Designation Number Argentina Free Call 0800 444 1009 Australia Free Call 1800 763 650 Austria Free Call 0800 111 956 Austria Local Call 0192 865 72 Belgium Free Call 0800 724 46 Belgium Local Call 0817 000 60 Brazil Free Call 0800 761 0835 Bulgaria Free Call 0080 011 511 76 Canada Free Call 1866 984 6577 Columbia Free Call 0180 091 562 17 Croatia Free Call 0800 222 305 Cyprus Free Call 8009 6341 Czech Republic Free Call 8007 007 95 Denmark Free Call 8088 8467 Denmark Local Call 3272 7506 Finland Free Call 0800 112 398 Finland Local Call 0923 114 014 France Free Call 0805 110 463 France Local Call 0359 580 290 Germany Free Call 0800 101 4918 Germany Local Call 0692 222 161 19 Greece Free Call 0080 012 8135 Hong Kong Free Call 8009 661 55 Hungary Free Call 0680 018 839 Hungary Local Call 0180 889 97 India Free Call 0008 001 006 600 Ireland Free Call 1800 300 170 Ireland Local Call 0143 198 35 Israel Free Call 1809 431 440 Italy Free Call 8007 840 87 Italy Local Call 0236 009 700 Japan Free Call 0066 338 124 31 Latvia Free Call 8000 3680 Luxembourg Free Call 8002 7941 Malaysia Free Call 1800 814 528 Mexico Free Call 0018 666 864 905 Monaco Free Call 8009 3655 Netherlands Free Call 0800 949 4596 Netherlands Local Call 0207 168 000 New Zealand Free Call 0800 451 190 North China Free Call 1080 074 413 29 Norway Free Call 8001 8057 Norway Local Call 2151 0847 Poland Free Call 0080 012 135 73 Portugal Free Call 8007 894 20 Romania Free Call 0800 895 558 Russia Free Call 8108 002 385 2044 Slovenia Free Call 0800 804 55 South Africa Free Call 0800 982 794 South China Free Call 1080 044 111 82 South Korea Free Call 0079 814 800 7887 Spain Free Call 9009 389 85 Spain Local Call 9111 421 10 Sweden Free Call 0200 214 344 Sweden Local Call 0850 596 375 Switzerland Free Call 0800 835 040 Switzerland Local Call 0445 804 280 Thailand Free Call 0018 004 421 98 UK Free Call 0800 073 1830 UK Local Call 0844 871 9364 UK National Call 0871 700 0309 UK Standard International +44 (0) 1452 562 665 USA Free Call 1866 230 1938   Back to the top   Copyright? 2010, Oracle. All rights reserved. Contact Us | Legal Notices and Terms of Use | Privacy Statement

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  • Server extremely lags and logs bunch of 'internal dummy connection'

    - by Dmitry
    Having a web-server (don't know actually whoset it up, it's my heritage). Few hours ago it started working very (extremely!) slow, mysqld oftenly fails requests. /var/log/mysqld.log is empty (well, it says, mysqld started, and so on, but nothing regarding today) /var/log/apache2/access_log is full of such lines: ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" ::1 - - [30/Nov/2011:10:15:05 +0100] "GET / HTTP/1.0" 200 1 "-" "Apache/2.2.3 (Linux/SUSE) (internal dummy connection)" Guys, what's that? How to heal this? I read internal dummy connections happen sometimes, but sending internal requests at 1000/sec frequency isn't freaking normal!How to find out the reason of this?

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  • EU Research for ICT - Call 7 - biggest ever at € 780 million

    - by trond-arne.undheim
    Under the Digital Agenda for Europe, the Commission has committed to maintaining the pace of a 20% yearly increase of the annual ICT R&D budget at least until 2013. The EU's flagship policy programme calls for doubling of annual public spending on ICT R&D by 2020 and to leverage an equivalent increase in private spending to achieve the goals of Europe's 2020 strategy for jobs and growth. Call 7 is one of the biggest calls ever launched for information and communications technology (ICT) research proposals under the EU's research framework programmes. It will result in project funding of € 780 million in 2011. This funding will advance research on the future internet, robotics, smart and embedded systems, photonics, ICT for energy efficiency, health and well-being in an ageing society, and more. The €780 million call for proposals is part of the biggest ever annual Work Programme under the EU's 7th Framework Programme for Research. Almost €1.2 billion has been budgeted for 2011. €220 million were made available already in July 2010 for public private partnerships focusing on ICT for smart cars, green buildings, sustainable factories and the future internet. Universities, research centres, SMEs, large companies and other organisations in Europe and beyond are eligible to apply for project funding under ICT Call 7. Proposals can be submitted until 18 January 2011, after which they will be evaluated by independent panels of experts for selection on the basis of their quality. Background: Digital Agenda: European Commission announces €780 million boost for strategic ICT research. Call text: ICT Call 7 Deadline: 18/01/2011.

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  • CRM at Oracle Series: Do Not Call & Do Not Email

    - by tony.berk
    Who you gonna call? Or not call! Sorry, just kidding, this isn't a movie blog! Do Not Call is an important topic for all businesses as there are government regulations that can lead to significant fines, and of course, possible damage to your brand. Oracle leverages Siebel CRM to develop an effective solution to address the Do Not Call and Email Permissible Use requirements. The application uses the Contacts functionality to manage communication preferences, which when defined, centrally synchronizes all contact records that share the same phone number and email address. Additionally, the relevant information is masked so Oracle employees cannot accidentally reach out to the contact. Therefore, the solution ensures that we are compliant with regulations, enables us to respect individuals' communication preferences and provides an audit trail of changes to their preferences. Today's CRM at Oracle slidecast discusses the requirements, highlights benefits and provides screen shots of the solution. CRM at Oracle Series: Do Not Call & Do Not Email Click here to learn more about Siebel CRM and other Oracle CRM products. Are you enjoying the CRM at Oracle Series? We are working on more topics for this year, but if there is a particular CRM area or function which you'd like to hear how Oracle implemented it internally, leave us a comment and we'll try to get it on our list.

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  • Log call information whenever there is a call.

    - by linuxdoniv
    Hi, I have written the android application and I want the application to send the call information whenever there is an incoming call and it ends. This way I would be sending all calls to the server irrespective of size of the call log. Here is the code public class PhoneInfo extends BroadcastReceiver { private int incoming_call = 0; private Cursor c; Context context; public void onReceive(Context con, Intent intent) { c = con.getContentResolver().query( android.provider.CallLog.Calls.CONTENT_URI, null, null, null, android.provider.CallLog.Calls.DATE+ " DESC"); context = con; IncomingCallListener phoneListener=new IncomingCallListener(); TelephonyManager telephony = (TelephonyManager) con.getSystemService(Context.TELEPHONY_SERVICE); telephony.listen(phoneListener,PhoneStateListener.LISTEN_CALL_STATE); } public class IncomingCallListener extends PhoneStateListener { public void onCallStateChanged(int state,String incomingNumber){ switch(state){ case TelephonyManager.CALL_STATE_IDLE: if(incoming_call == 1){ CollectSendCallInfo(); incoming_call = 0; } break; case TelephonyManager.CALL_STATE_OFFHOOK: break; case TelephonyManager.CALL_STATE_RINGING: incoming_call = 1; break; } } } private void CollectSendCallInfo() { int numberColumn = c.getColumnIndex( android.provider.CallLog.Calls.NUMBER); int dateColumn = c.getColumnIndex( android.provider.CallLog.Calls.DATE); int typeColumn = c.getColumnIndex( android.provider.CallLog.Calls.TYPE); int durationColumn=c.getColumnIndex( android.provider.CallLog.Calls.DURATION); ArrayList<String> callList = new ArrayList<String>(); try{ boolean moveToFirst=c.moveToFirst(); } catch(Exception e) { ; // could not move to the first row. return; } int row_count = c.getCount(); int loop_index = 0; int is_latest_call_read = 0; String callerPhonenumber = c.getString(numberColumn); int callDate = c.getInt(dateColumn); int callType = c.getInt(typeColumn); int duration=c.getInt(durationColumn); while((loop_index <row_count) && (is_latest_call_read != 1)){ switch(callType){ case android.provider.CallLog.Calls.INCOMING_TYPE: is_latest_call_read = 1; break; case android.provider.CallLog.Calls.MISSED_TYPE: break; case android.provider.CallLog.Calls.OUTGOING_TYPE: break; } loop_index++; c.moveToNext(); } SendCallInfo(callerPhonenumber, Integer.toString(duration), Integer.toString(callDate)); } private void SendCallInfo(String callerPhonenumber, String callDuration, String callDate) { JSONObject j = new JSONObject(); try { j.put("Caller", callerPhonenumber); j.put("Duration", callDuration); j.put("CallDate", callDate); } catch (JSONException e) { Toast.makeText(context, "Json object failure!", Toast.LENGTH_LONG).show(); } String url = "http://xxxxxx.xxx.xx/xxxx/xxx.php"; Map<String, String> kvPairs = new HashMap<String, String>(); kvPairs.put("phonecall", j.toString()); HttpResponse re; try { re = doPost(url, kvPairs); String temp; try { temp = EntityUtils.toString(re.getEntity()); if (temp.compareTo("SUCCESS") == 0) { ; } else ; } catch (ParseException e1) { Toast.makeText(context, "Parse Exception in response!", Toast.LENGTH_LONG) .show(); e1.printStackTrace(); } catch (IOException e1) { Toast.makeText(context, "Io exception in response!", Toast.LENGTH_LONG).show(); e1.printStackTrace(); } } catch (ClientProtocolException e1) { Toast.makeText(context, "Client Protocol Exception!", Toast.LENGTH_LONG).show(); e1.printStackTrace(); } catch (IOException e1) { Toast.makeText(context, "Client Protocol Io exception!", Toast.LENGTH_LONG).show(); e1.printStackTrace(); } } and here is the manifest file <uses-permission android:name="android.permission.ACCESS_COARSE_LOCATION"></uses-permission> <uses-permission android:name="android.permission.INTERNET"></uses-permission> <uses-permission android:name="android.permission.ACCESS_FINE_LOCATION"></uses-permission> <uses-permission android:name="android.permission.ACCESS_LOCATION_EXTRA_COMMANDS"></uses-permission> <uses-permission android:name="android.permission.INSTALL_LOCATION_PROVIDER"></uses-permission> <uses-permission android:name="android.permission.SET_DEBUG_APP"></uses-permission> <uses-permission android:name="android.permission.RECEIVE_SMS"></uses-permission> <uses-permission android:name="android.permission.READ_PHONE_STATE"></uses-permission> <uses-permission android:name="android.permission.READ_SMS"></uses-permission> <application android:icon="@drawable/icon" android:label="@string/app_name"> <activity android:name=".Friend" android:label="@string/app_name"> <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> <activity android:name=".LoginInfo" android:label="@string/app_name"> <intent-filter> <action android:name="android.intent.action.DEFAULT" /> </intent-filter> </activity> <service android:exported="true" android:enabled="true" android:name=".GeoUpdateService" > </service> <receiver android:name=".SmsInfo" > <intent-filter> <action android:name= "android.provider.Telephony.SMS_RECEIVED" /> </intent-filter> </receiver> <receiver android:name=".PhoneInfo" > <intent-filter> <action android:name="android.intent.action.PHONE_STATE"></action> </intent-filter> </receiver> </application> The application just crashes when there is an incoming call.. i have been able to log the information about incoming SMS, but this call info logging is failing. Thanks for any help.

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  • Call for Papers Ends March 21

    - by jack.flynn
    Have Something to Say? Better Say So Now. The Call for Papers for Oracle OpenWorld and the Develop Stream of JavaOne+Develop ends at midnight on Sunday, March 21. So if you want to be a part of the most influential IT events of the year, don't let this chance pass you by. This year offers opportunities to speak out about some new subjects: Oracle OpenWorld adds a whole new Server and Storage Systems stream, including Sun servers, Sun storage and tape, and Oracle Solaris operating system. And the Develop audience should be larger and more energetic than ever now that it's co-located with JavaOne. If you have something important to say, this is the time to let us know. Find all the information on the Call for Papers process, timeline, and guidelines here.

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  • Asterisk - Trying to use call files to create a conference call between two dynamic numbers

    - by Hank
    I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing an incoming call - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out This example seems to be what I'd need: Channel: SIP/mytrunk/12345678 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 I can generate this file with some scripting language and then place it into the Asterisk Call File folder. The problem I'm having is: How do I call out to two numbers and join them in a conference call? The MeetMe plugin/extension seems to be what I need in terms of conference calling, I'm just unsure as to how I'd use the two together and join them. Also, is it possible to have multiple 2-person conference calls at the same time? Is setting this up as simple as setting aside X amount of 'channels' in the meetme.conf?

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  • Asterisk - Trying to use call files to create a conference call between two dynamic numbers

    - by Hank
    I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing an incoming call - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out This example seems to be what I'd need: Channel: SIP/mytrunk/12345678 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 I can generate this file with some scripting language and then place it into the Asterisk Call File folder. The problem I'm having is: How do I call out to two numbers and join them in a conference call? The MeetMe plugin/extension seems to be what I need in terms of conference calling, I'm just unsure as to how I'd use the two together and join them. Also, is it possible to have multiple 2-person conference calls at the same time? Is setting this up as simple as setting aside X amount of 'channels' in the meetme.conf?

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  • It's the Freedom You Big Dummy

    <b>Daniweb:</b> "No one has given his life for Linux but certainly there have been sacrifices. But, like their armed soldier counterparts, it isn't about the sacrifice, it's the freedom you big dummy."

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  • Pulseaudio no sound card detected. Dummy output only

    - by Zach Smith
    I'm using 12.10 Quantal with Openbox and a .xinitrc script at login instead of a display manager. Its a relatively fresh install and I noticed when I opened pavucontrol the only output was a dummy one. I check around and it appears that my soundcard is physically installed but Pulseaudio isn't detecting it. I'm really unsure what I should do but any help getting my audio back would be appreciated. Edit: further info if its at all useful: dante@dante-ubuntu:~$ uname -a && aplay -l && cat /proc/asound/version && head -n 1 /proc/asound/card*/codec#* Linux dante-ubuntu 3.5.0-17-generic #28-Ubuntu SMP Tue Oct 9 19:31:23 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux aplay: device_list:252: no soundcards found... Advanced Linux Sound Architecture Driver Version 1.0.25. == /proc/asound/card0/codec#0 <== Codec: ATI R6xx HDMI == /proc/asound/card1/codec#0 <== Codec: IDT 92HD81B1X5

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  • No sound 12.04 (Dummy output)

    - by Edgar Adrian Alvarez
    Its been weeks with no sound. I feel like Ive tried everything but somethings just dont seem right. I am a new user and so far i love Ubuntu but this sound issue is making me unsure. Im NOT muted. Ive tried multiple jacks, front and back. in alsamixer it says choose sound card and I have only thr 'hda Intel' option. In pulseaudio I only have 'dummy output'. When I have Youtube on I can see audio being detected in pavucontrol but nothing is coming out of the speakers. Im getting desperate, some one please walk me thru this. http://www.alsa-project.org/db/?f=c7377242d96ea884edebd807f4fe71f619b8d6af What more information should i provided?

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  • Dummy output after upgrade from 12.04 to 12.10, even though sound card is detected

    - by user115441
    So I just recently upgraded my system from Ubuntu 12.04 to 12.10. However, when I booted into 12.10 for the first time, no sound comes out of my speakers. I checked the sound settings and the Dummy Output was the only thing showing up. I used "hwinfo --sound" to check to see if my sound card was actually installed, and it was installed. hwinfo --sound hal.1: read hal dataprocess 2687: arguments to dbus_move_error() were incorrect, assertion "(dest) == NULL || !dbus_error_is_set ((dest))" failed in file ../../dbus/dbus-errors.c line 282. This is normally a bug in some application using the D-Bus library. libhal.c 3483 : Error unsubscribing to signals, error=The name org.freedesktop.Hal was not provided by any .service files 11: PCI 1b.0: 0403 Audio device [Created at pci.318] Unique ID: u1Nb._aiKlM91Nt0 SysFS ID: /devices/pci0000:00/0000:00:1b.0 SysFS BusID: 0000:00:1b.0 Hardware Class: sound Model: "Intel 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller" Vendor: pci 0x8086 "Intel Corporation" Device: pci 0x2668 "82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller" SubVendor: pci 0x107b "Gateway 2000" SubDevice: pci 0x4040 Revision: 0x04 Driver: "snd_hda_intel" Driver Modules: "snd_hda_intel" Memory Range: 0x50240000-0x50243fff (rw,non-prefetchable) IRQ: 44 (91 events) Module Alias: "pci:v00008086d00002668sv0000107Bsd00004040bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=new, avail=yes, need=no, active=unknown I'm not sure what to do here. The only time the sound will actually work is when I boot into my Windows partition and then reboot into Ubuntu. I mean I don't want to have to do that every time I want to use Ubuntu. I would really appreciate any help I can get on here.

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  • Microsoft office communicator 2007 r2 can not send video call to multiple user

    - by mihan007
    so i install ms communication server 2007 r2 and ms office communicator 2007 r2 for three user. so it is possible now to send video call from one person to another. you can use it so way: you choose a person from your contact list and from context menu choose 'send video call'. but when i choose several persons which I want to talk by video this option doesn't exist. (but it should be - http://www.useto.ru/images/01/image015.jpg). what is the problem and how can i tune up communication server or communicator for ability to send video call to several persons?

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  • Cannot get Correct month for a call from call log history

    - by Nishant Kumar
    I am trying to extract information from the call log of the android. I am getting the call date that is one month back from the actual time of call. I mean to say that the information extracted by my code for the date of call is one mont back than the actual call date. I have the following in the Emulator: I saved a contact. Then I made a call to the contact. Code: I have 3 ways of extracting call Date information but getting the same wrong result. My code is as follows: /* Make the query to call log content */ Cursor callLogResult = context.getContentResolver().query( CallLog.Calls.CONTENT_URI, null, null, null, null); int columnIndex = callLogResult.getColumnIndex(Calls.DATE); Long timeInResult = callLogResult.getLong(columnIndex); /* Method 1 to change the milliseconds obtained to the readable date formate */ Time time = new Time(); time.toMillis(true); time.set(timeInResult); String callDate= time.monthDay+"-"+time.month+"-"+time.year; /* Method 2 for extracting the date from tha value read from the column */ Calendar calendar = Calendar.getInstance(); calendar.setTimeInMillis(time); String Month = calendar.get(Calendar.MONTH) ; /* Method 3 for extracting date from the result obtained */ Date date = new Date(timeInResult); String mont = date.getMonth() While using the Calendar method , I also tried to set the DayLight SAving Offset but it didnot worked, calendar.setTimeZone(TimeZone.getTimeZone("Europe/Paris")); int DST_OFFSET = calendar.get( Calendar.DST_OFFSET ); // DST_OFFSET Boolean isSet = calendar.getTimeZone().useDaylightTime(); if(isSet) calendar.set(Calendar.DST_OFFSET , 0); int reCheck = calendar.get(Calendar.DST_OFFSET ); But the value is not set to 0 in recheck. I am getting the wrong month value by using this also. Please some one help me where I am wrong? or is this the error in emulator ?? Thanks, Nishant Kumar Engineering Student

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  • WCF: Per-Call and Per-Session services...need convincing that Per-Call is worthwhile

    - by mrlane
    Hello all. We are currently doing a review of our WCF service design and one thing that is bothering me is the decision between Per-Call and Per-Session services. I believe I understand the concept behind both, but I am not really seeing the advantage of Per-Call services. I understand that the motivation for using Per-Call services is that a WCF services only holds a servier object for the life of a call thereby restricting the time that an expensive resource is held by the service instance, but to me its much simpler to use the more OO like Per-Session model where your proxy object instance always corrisponds to the same server object instance and just handle any expensive resources manually. For example, say I have a CRUD Service with Add, Update, Delete, Select methods on it. This could be done as a Per-Call service with database connection (the "expensive resource") instanciated in the server object constructor. Alternately it could be a Per-Session service with a database connection instanciated and closed within each CRUD method exposed. To me it is no different resource wise and it makes the programming model simpler as the client can be assured that they always have the same server object for their proxies: any in-expensive state that there may be between calls is maintained and no extra parameters are needed on methods to identify what state data must be retrieved by the service when it is instanciating a new server object again (as in the case of Per-Call). Its just like using classes and objects, where the same resource management issues apply, but we dont create new object instances for each method call we have on an object! So what am I missing with the Per-Call model? Thanks

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  • Change Call Screen

    - by vyana
    I need to change or customize the call screen when initiating a call on Android. After searching on google I do not find any way to do it. There is no way to send DTMF tones during a call, the idea is to send a specific number to the call screen. So when a call is made is possible to see the number to dial during a call to the PBX. I tried to putting the number in the "status bar", but the notification hide after seconds and it is not practical. I appreciate any other suggestion. Thanks

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  • Dummy Guide to NetBeans Android Development

    - by Geertjan
    Start by setting up the Android SDK (fantastic Ubuntu instructions here), then install NBAndroid. Now you can create a new Android project: Having set up the Android SDK, you're able to select your Android platform in the IDE: The project structure created by the above templates is nice and easy to understand: Build the project and you have your APK file and everything else generated in the Files window: Nice features are included, such as code completion in Android XML files: Several other features are included, as described here, such as "Export Signed Android Package", as well as deployment to the Android emulator. Now that I have everything set up (took literally about 10 minutes from start to finish), I'm going to be experimenting a bit with Android development via NetBeans IDE.

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  • Making dummy applications while not involved in LIVE work [closed]

    - by Ratan Sharma
    I know this is subjective but I am looking for some real time helpful points/advice here, which will be helpful for some to get motivated. In our company so many people are on bench(not assigned with real time work) and they do not want to experiment things by their own. What would be a good motivation for them to keep their learning spirit? I personally feel that one can learn and give more effort in live client work than regular practicing things and making dummies. Am I right here or it is just my thinking only?

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  • Bringing people into an Asterisk conference call

    - by Harley
    I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO is trying to do, but it doesn't work quite properly for me. Here's what happens: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. In the last step, A should be taken to the conference room as well. Here's the relevant logs, where 230 is person A, 231 is person B, 207 is person C, and 282 is the conference room.

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  • Over 200 active requests like "OPTIONS * HTTP/1.0" 200 - "-" "Apache (internal dummy connection)"

    - by Stefan Lasiewski
    Some details: Webserver: Apache/2.2.13 (FreeBSD) mod_ssl/2.2.13 OpenSSL/0.9.8e OS: FreeBSD 7.2-RELEASE This is a FreeBSD Jail. I believe I use the Apache 'prefork' MPM (I run the default for FreeBSD). I use the default values for MaxClients (256) I have enabled mod_status, with "ExtendedStatus On". When I view /server-status , I see a handful of regular requests. I also see over 230 requests from the 'localhost', like these: 37-0 - 0/0/1 . 0.00 1510 0 0.0 0.00 0.00 127.0.0.2 www.example.gov OPTIONS * HTTP/1.0 38-0 - 0/0/1 . 0.00 1509 0 0.0 0.00 0.00 127.0.0.2 www.example.gov OPTIONS * HTTP/1.0 39-0 - 0/0/3 . 0.00 1482 0 0.0 0.00 0.00 127.0.0.2 www.example.gov OPTIONS * HTTP/1.0 40-0 - 0/0/6 . 0.00 1445 0 0.0 0.00 0.00 127.0.0.2 www.example.gov OPTIONS * HTTP/1.0 I also see about 2417 requests yesterday from the localhost, like these: Apr 14 11:16:40 192.168.16.127 httpd[431]: www.example.gov 127.0.0.2 - - [15/Apr/2010:11:16:40 -0700] "OPTIONS * HTTP/1.0" 200 - "-" "Apache (internal dummy connection)" The page at http://wiki.apache.org/httpd/InternalDummyConnection says "These requests are perfectly normal and you do not, in general, need to worry about them", but I'm not so sure. Why are there over 230 of these? Are these active connections? If I have "MaxClients 256", and over 230 of these connections, it seems that my webserver is dangerously close to running out of available connections. It also seems like Apache should only need a handful of these "internal dummy connections" We actually had two unexplained outages last night, and I am wondering if these "internal dummy connection" caused us to run out of available connections.

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  • Call to a phone number through iPhone App

    - by Md. Faisal Rahman
    Hi iPhone developers, I want to add a feature in my iPhone app, the are: call to a phone number in my app play a recorded mp3 voice to that number after call end, relaunch the previous app I know I have to use following code snipt for dialing to a number XXXXXX: [[UIApplication sharedApplication] openURL:[NSURL URLWithString:@"tel:XXXXXX"]]; My be play record not worked, as my app will terminate when call dial launch. is there any way to do this? And, after call ended, or call failed will my previous app relaunch? please answer ASAP.

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