Search Results

Search found 358 results on 15 pages for 'dial up'.

Page 6/15 | < Previous Page | 2 3 4 5 6 7 8 9 10 11 12 13  | Next Page >

• How to control routes added by RasDial

  - by Robert Dodier

  I am using the RasDial function on a Windows box (Windows Server 2008) to dial a device from which the server then reads data. It seems that some new routes are added to the network routing table when the dial-up connection is made. That interferes with other network interfaces on the server. In particular, RasDial adds a default route which routes traffic to the device, which makes the server unreachable until the connection is dropped. Is there a way to control which routes are added by RasDial? I have been studying Microsoft's document for RasDial and associated items (RASDIALPARAMS, RASDIALEXTENSIONS) without finding anything about routing. There is an option for "Use default gateway on remote network" when configuring a VPN, but I don't see how to apply that in this case. Thanks for any light you can shed on this problem.

  Read the article

• Unofficial Prep guide for TS: Microsoft Lync Server 2010, Configuring (70-664)

  - by Enrique Lima

  Managing Users and Client Access (20 percent)   Objective Materials Configure user accounts http://technet.microsoft.com/en-us/library/gg182543.aspx Deploy and maintain clients http://technet.microsoft.com/en-us/library/gg412773.aspx Configure conferencing policies http://technet.microsoft.com/en-us/library/gg182561.aspx Configure IM policies http://technet.microsoft.com/en-us/library/gg182558.aspx Deploy and maintain Lync Server 2010 devices http://technet.microsoft.com/en-us/library/gg412773.aspx Resolve client access issues http://technet.microsoft.com/en-us/library/gg398307.aspx   Configuring a Lync Server 2010 Topology (21 percent)   Objective Materials Prepare to deploy a topology http://technet.microsoft.com/en-us/library/gg398630.aspx Configure Lync Server 2010 by using Topology Builder http://technet.microsoft.com/en-us/library/gg398420.aspx Configure role-based access control in Lync Server 2010 http://technet.microsoft.com/en-us/library/gg412794.aspx http://technet.microsoft.com/en-us/library/gg425917.aspx Configure a location information server http://technet.microsoft.com/en-us/library/gg398390.aspx Configure server pools for load balancing http://technet.microsoft.com/en-us/library/gg398827.aspx   Configuring Enterprise Voice (19 percent)   Objective Materials Configure voice policies http://technet.microsoft.com/en-us/library/gg398450.aspx Configure dial plans http://technet.microsoft.com/en-us/library/gg398922.aspx Manage routing http://technet.microsoft.com/en-us/library/gg425890.aspx http://technet.microsoft.com/en-us/library/gg182596.aspx Configure Microsoft Exchange Unified Messaging integration http://technet.microsoft.com/en-us/library/gg398768.aspx Configure dial-in conferencing http://technet.microsoft.com/en-us/library/gg398600.aspx Configure call admission control http://technet.microsoft.com/en-us/library/gg520942.aspx Configure Response Group Services (RGS) http://technet.microsoft.com/en-us/library/gg398584.aspx Configure Call Park and Unassigned Number http://technet.microsoft.com/en-us/library/gg399014.aspx http://technet.microsoft.com/en-us/library/gg425944.aspx Manage a Mediation Server pool and PSTN Gateway http://technet.microsoft.com/en-us/library/gg412780.aspx   Configuring Lync Server 2010 for External Access (19 percent)   Objective Materials Configure Edge Services http://technet.microsoft.com/en-us/library/gg398918.aspx Configure a firewall http://technet.microsoft.com/en-us/library/gg425882.aspx Configure a reverse proxy http://technet.microsoft.com/en-us/library/gg425779.aspx   Monitoring and Maintaining Lync Server 2010 (21 percent)   Objective Materials Back up and restore Lync Server 2010 http://technet.microsoft.com/en-us/library/gg412771.aspx Configure monitoring and archiving http://technet.microsoft.com/en-us/library/gg398199.aspx http://technet.microsoft.com/en-us/library/gg398507.aspx http://technet.microsoft.com/en-us/library/gg520950.aspx http://technet.microsoft.com/en-us/library/gg520990.aspx Implement troubleshooting tools http://technet.microsoft.com/en-us/library/gg425800.aspx Use PowerShell to test Lync Server 2010 http://technet.microsoft.com/en-us/library/gg398474.aspx

  Read the article

• Join us for our WebLogic Communtiy webcast on November 2nd 2012! OOW update WebLogic & ExaLogic

  - by JuergenKress

  NOVEMBER 2nd, 2012 AT 11:00 -11:45 AM CET (BERLIN TIME) Do you want to learn the latest about WebLogic and ExaLogic? Would you like to know what did Oracle announce at Oracle OpenWorld 2012? Join us for our WebLogic Communtiy webcast on November 2nd 2012! Don't miss this unique opportunity and learn the latest announcement and product updates on WebLogic and ExaLogic portfolio. Agenda Update announced about Cloud Application Foundation ExaLogic Update Key examples from successful customer Updated Roadmap Q&A Register to Attend! To Join the Webcast (Employees and Partners) AND / OR DIAL IN If you would like to dial in to the audio portion of the Webconference. Call ID: 8000524 & Call Passcode: 333111 Austria : +43(0)19286512 Belgium: +32(0)24010528 Denmark: +4532729222 Finland: +358(0)923193923 France: +33(0)176728936 Germany: +49(0)69222216106 Ireland: +353(0)12475650 Italy: +39(0)236008198 Netherlands: +31(0)207143543 Norway: +4721033443 Spain: +34914143755 Sweden: +46(0)856619465 Switzerland: +41(0)445804003 United Kingdom: +44(0)2081181001 Please click here to find more Local Numbers. Presenters: Maciej Gruszka Senior Principal Product Manager Tel: +4 (0) 8601 156 464 E-Mail: [email protected] LinkedIn: http://pl.linkedin.com/pub/maciej-gruszka/2/169/89 Jürgen Kress WebLogic Partner Adoption EMEA Tel. +49 89 1430 1479 E-Mail: [email protected] LinkedIn: http://de.linkedin.com/in/kress Twitter: http://twitter.com/wlscommunity Delivery Format This FREE online LIVE eSeminar will be delivered over the Web. Registrations received less than 24hours prior to start time may not receive confirmation to attend. Duration: 1 hour Looking forward to your participation! WebLogic Partner Community For regular information become a member in the WebLogic Partner Community please visit: http://www.oracle.com/partners/goto/wls-emea ( OPN account required). If you need support with your account please contact the Oracle Partner Business Center. BlogTwitterLinkedInMixForumWiki Technorati Tags: WebLogic Community Webcast,webcst,OOW updates WebLogic ExaLogic,WebLogic ppt,presenation,sales,WebLogic Community,Oracle,OPN,Jürgen Kress

  Read the article

• Blank New Tab Quick-Fix for Google Chrome

  - by Asian Angel

  If you have other browsers that you use set to “about:blank” for new tabs then you probably feel rather frustrated with Google Chrome’s default New Tab Page. The Blank New Tab extension is the perfect solution to that problem. Before Unless you have a “speed dial/special page” extension installed you are stuck with the default new tab page in Chrome every single time you open a new tab. What if you do not like the default new tab page or “speed dial/special page” setups? After If you are someone who prefers to have a blank page as a new tab then you will love this extension. Once you have it installed you can click to your heart’s content on the “New Tab Button” and see nothing but blank goodness. Sometimes less is more… Note: There are no options to bother with. Conclusion If you prefer a blank page when opening a new tab then the Blank New Tab extension is just what you have been waiting for. Links Download the Blank New Tab extension (Google Chrome Extensions) Similar Articles Productive Geek Tips Subscribe to RSS Feeds in Chrome with a Single ClickAccess Wolfram Alpha Search in Google ChromeFind Similar Websites in Google ChromeHow to Make Google Chrome Your Default BrowserView Maps and Get Directions in Google Chrome TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Need Help with Your Home Network? Awesome Lyrics Finder for Winamp & Windows Media Player Download Videos from Hulu Pixels invade Manhattan Convert PDF files to ePub to read on your iPad Hide Your Confidential Files Inside Images

  Read the article

• JCP 2.9 and Transparency Call for Spec Leads 9 November

  - by heathervc

  JCP Spec Leads are invited to participate in an online meeting/call this Friday, 9 November, to hear a talk about the the 2.9 version of the Java Community Process (effective date of 13 November) and discuss the changes with representatives of the Program Management Office.  This call will be recorded and published with materials for those not able to attend.  Details of the call are included below.JCP 2.9 is presented in two documents:The JCP 2.9 document:http://www.jcp.org/en/procedures/jcp2and the EC Standing Rules document:http://www.jcp.org/en/procedures/ec_standing_rulesIn addition, we will be reviewing ways to collect community feedback on the transparency requirements for JCP 2.7 and above JSRs (JCP 2.8, JCP 2.9), detailed as part of the Spec Lead Guide.Call details:Topic: JCP 2.9 and Transparency Date: Friday, November 9, 2012 Time: 9:00 am, Pacific Standard Time (San Francisco, GMT-08:00) Meeting Number: 800 623 574 Meeting Password: 5282 ------------------------------------------------------- To start or join the online meeting ------------------------------------------------------- Go to https://jcp.webex.com/jcp/j.php?ED=188925347&UID=491098062&PW=NMDZiYTQzZmE1&RT=MiM0 ------------------------------------------------------- Audio conference information ------------------------------------------------------- Toll-Free Dial-In Number:     866 682-4770 International (Toll) Dial-In Number:     408 774-4073 Conference code 9454597 Security code 1020 Outside the US: global access numbers   https://www.intercallonline.com/portlets/scheduling/viewNumbers/listNumbersByCode.do?confCode=6279803

  Read the article

• PPP connection with RAS dialer in C++

  - by user312054

  I have a windows mobile application that is using Windows CE 5.0. I have been informed by the people supplying the hardware for the unit that I need to create a socket, which I have done successfully, and then dial out to the internet with a PPP connection with a RAS dialer connection. Our old code uses an APN to dial out so I need to create the above connection with an APN. I am having trouble finding examples of this. Can someone point me to some examples of this situation?

  Read the article

• Can you dynamically resize a java.util.concurrent.ThreadPoolExecutor while it still has tasks waitin

  - by Edward Shtern

  I'm working with a java.util.concurrent.ThreadPoolExecutor to process a number of items in parallel. Although the threading itself works fine, at times we've run into other resource constraints due to actions happening in the threads, which made us want to dial down the number of Threads in the pool. I'd like to know if there's a way to dial down the number of the threads while the threads area actually working. I know that you can call setMaximumPoolSize() and/or setCorePoolSize(), but these only resize the pool once threads become idle, but they don't become idle until there are no tasks waiting in the queue.

  Read the article

• Customise a control in dynamics crm

  - by webturner

  I've written code that can make a phone dial a number from a function call, that's done and dusted. What I would like to achieve is adding a Dial button to each phone number field on the forms in Dynamics CRM. Eventually this could also create a new phone record fill in the basic details and show it to the user to enter notes and an outcome for the phone call, and perhaps some other workflow bits to schedule the next call. Can I put a custom control on a standard form in place of the standard control. I'm assuming it would have to be an IFrame to an asp.net page, that pulls in the record id, and the field name, looks up the number to show in a text box, and passes the number to the DialNumber function. Hey presto... I assume its not going to be that easy... Has anyone tried this before, what's the process, what are the gotchas?

  Read the article

• Methodology to understanding JQuery plugin & API's developed by third parties

  - by Taoist

  I have a question about third party created JQuery plug ins and API's and the methodology for understanding them. Recently I downloaded the JQuery Masonry/Infinite scroll plug in and I couldn't figure out how to configure it based on the instructions. So I downloaded a fully developed demo, then manually deleted everything that wouldn't break the functionality. The code that was left allowed me to understand the plug in much greater detail than the documentation. I'm now having a similar issue with a plug in called JQuery knob. http://anthonyterrien.com/knob/ If you look at the JQuery Knob readme file it says this is working code: <input type="text" value="75" class="dial"> $(function() { $('.dial') .trigger( 'configure', { "min":10, "max":40, "fgColor":"#FF0000", "skin":"tron", "cursor":true } ); }); But as far as I can tell it isn't at all. The read me also says the Plug in uses Canvas. I am wondering if I am suppose to wrap this code in a canvas context or if this functionality is already part of the plug in. I know this kind of "question" might not fit in here but I'm a bit confused on the assumptions around reading these kinds of documentation and thought I would post the query regardless. Curious to see if this is due to my "newbi" programming experience or if this is something seasoned coders also fight with. Thank you. Edit In response to Tyanna's reply. I modified the code and it still doesn't work. I posted it below. I made sure that I checked the Google Console to insure the basics were taken care of, such as not getting a read-error on the library. <!DOCTYPE html> <meta charset="UTF-8"> <title>knob</title> <link rel="stylesheet" href="http://ajax.googleapis.com/ajax/libs/jqueryui/1.7.2/themes/hot-sneaks/jquery-ui.css" type="text/css" /> <script type="text/javascript" src="https://ajax.googleapis.com/ajax/libs/jquery/1.7.2/jquery.js" charset="utf-8"></script> <script src="https://ajax.googleapis.com/ajax/libs/jqueryui/1.8.21/jquery-ui.min.js"></script> <script src="js/jquery.knob.js"></script> <div id="button1">test </div> <script> $(function() { $("#button1").click(function () { $('.dial').trigger( 'configure', { "min":10, "max":40, "fgColor":"#FF0000", "skin":"tron", "cursor":true } ); }); }); </script>

  Read the article

• VB.NET custom user control graphics rotation

  - by AtharvaI

  Hi, this is my first question on here. I'm trying to build a dial control as a custom user control in VB.NET. I'm using VS2008. so far I have managed to rotate image using graphics.rotatetransform . however, this rotate everything. Now I have a Bitmap for the dial which should stay stable and another Bitmap for the needle which I need to rotate. so far i've tried this: Dim gL As Graphics = Graphics.FromImage(bmpLongNeedle) gL.TranslateTransform(bmpLongNeedle.Width / 2, bmpLongNeedle.Height * 0.74) gL.RotateTransform(angleLongNeedle) gL.TranslateTransform(-bmpLongNeedle.Width / 2, -bmpLongNeedle.Height * 0.74) gL.DrawImage(bmpLongNeedle, 0, 0) As I understand it, the image of the needle should be rotated at angle "angleLongNeedle" although i'm placing the rotated image at 0,0. However, the result is that the Needle doesn't get drawn on the control. any pointers as to where I might be going wrong or something else I should be doing? Thanks in advance

  Read the article

• Opera 11.10 sort en version bêta : meilleur support du CSS3 et prise en charge du standard WOFF et du format ouvert WebP

  Opera 11.10 sort en version bêta Meilleur support du CSS3 et prise en charge du standard WOFF et du format ouvert WebP Opera Software vient de dévoiler la bêta de la version 11.10 de son navigateur, encore plus rapide, épurée et performante. Côté visuel, cette version (nom de code Barracuda) garde le même design général simplifié de la version 11. Seule la fonctionnalité Speed Dial (galerie des miniatures) a été relookée pour faciliter et accélérer l'accès aux sites favoris. Le nombre d'adresses que l'utilisateur peut désormais placer da...

  Read the article

• How to associate hardware volume control to USB speakers?

  - by Wolfger

  My Xubuntu system recognizes my speakers, and I can change the mixer settings to make them the active sound device, but no matter what I do, the hardware dial will only affect the onboard sound device. generic-usb 0003:046D:0A19.0003: input,hidraw1: USB HID v1.00 Device [ Logitech Logitech Z205 ] on usb-0000:00:1d.1-1/input2 updated to reflect the fact that this was Xubuntu desktop (from Kubuntu install originally) where I had this problem. I was able to easily do this from Ubuntu Natty installation.

  Read the article

• Methodology to understanding JQuery plugin & API's developed by third parties

  - by Taoist

  I have a question about third party created JQuery plug ins and API's and the methodology for understanding them. Recently I downloaded the JQuery Masonry/Infinite scroll plug in and I couldn't figure out how to configure it based on the instructions. So I downloaded a fully developed demo, then manually deleted everything that wouldn't break the functionality. The code that was left allowed me to understand the plug in much greater detail than the documentation. I'm now having a similar issue with a plug in called JQuery knob. http://anthonyterrien.com/knob/ If you look at the JQuery Knob readme file it says this is working code: $(function() { $('.dial') .trigger( 'configure', { "min":10, "max":40, "fgColor":"#FF0000", "skin":"tron", "cursor":true } ); }); But as far as I can tell it isn't at all. The read me also says the Plug in uses Canvas. I am wondering if I am suppose to wrap this code in a canvas context or if this functionality is already part of the plug in. I know this kind of "question" might not fit in here but I'm a bit confused on the assumptions around reading these kinds of documentation and thought I would post the query regardless. Curious to see if this is due to my "newbi" programming experience or if this is something seasoned coders also fight with. Thank you. Edit In response to Tyanna's reply. I modified the code and it still doesn't work. I posted it below. I made sure that I checked the Google Console to insure the basics were taken care of, such as not getting a read-error on the library. <!DOCTYPE html> <meta charset="UTF-8"> <title>knob</title> <link rel="stylesheet" href="http://ajax.googleapis.com/ajax/libs/jqueryui/1.7.2/themes/hot-sneaks/jquery-ui.css" type="text/css" /> <script type="text/javascript" src="https://ajax.googleapis.com/ajax/libs/jquery/1.7.2/jquery.js" charset="utf-8"></script> <script src="https://ajax.googleapis.com/ajax/libs/jqueryui/1.8.21/jquery-ui.min.js"></script> <script src="js/jquery.knob.js"></script> <div id="button1">test </div> <script> $(function() { $("#button1").click(function () { $('.dial').trigger( 'configure', { "min":10, "max":40, "fgColor":"#FF0000", "skin":"tron", "cursor":true } ); }); }); </script>

  Read the article

• Opera 11.50 disponible en version finale, avec une interface utilisateur encore plus épurée

  Opera 11.50 disponible en version finale Avec une interface utilisateur encore plus épurée Mise à jour du 28/06/11, par Hinault Romaric L'éditeur norvégien Opera Software vient de publier la mouture finale de la version 11.50 de son navigateur (disponible sur toutes les plates-formes). Opera 11.50 intègre une nouvelle interface utilisateur complètement remaniée baptisée « Featherweight » (poids plume), encore plus épurée. On remarquera de nouvelles icones et couleurs, et un affichage des fonctionnalités sur un nombre plus restreint de ligne. Le navigateur propose une nouvelle fonction d'accès rapide « Speed Dial ...

  Read the article

• Is Your Website iPhone and Blackberry Friendly?

  You want to make sure that you have the essentials. You want to make sure there is text for the phone and email. If the phone number is in a graphic you have the write it down and then dial it.

  Read the article

• How can I simulate a website loading slowly? [closed]

  - by Nomistake

  Possible Duplicate: How can I simulate a slow connection for page load? I found some old treaths about this topic but is there a new/working way to slow down the loading of a website (local websever) to a predefined speed, as if it were, for example, on a dial-up connection? I didn't find a good working one... (On Windows). It would be nice if it's a Firefox add-on/plugin. Any suggestions?

  Read the article

• asterisk extensions.conf & sip.conf

  - by Josh

  I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

  Read the article

• Transfering call asterisk to different context

  - by Necronet

  I have a Small and basic PBX, and with two contexts wich basicly are sales and supervisor both have different roles and privileges. I notice that it is possible to transfer call from the same context but it have been imposible to transfer anything to another context. Any insight, i am kinda a rookie on asterisk but currently there is no one else in charge... Thanks Edit This is the extension.conf [supervisor] include => from-internal exten => _40XX,1,Answer exten => _40XX,n,Set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d.%H.%M.%S)}) exten => _40XX,n,Set(CALLEDNUMBER=${EXTEN}) exten => _40XX,n,MixMonitor(/tmp/Para_${CALLEDNUMBER}-${calltime}-De_${CALLERID(num)}.wav) exten => _40XX,n,Dial(SIP/${EXTEN},40,TtRr) exten => _40XX,n,Hangup [sales] include => out-trunksip exten => _41XX,1,Answer exten => _41XX,n,Set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d.%H.%M.%S)}) exten => _41XX,n,Set(CALLEDNUMBER=${EXTEN}) exten => _41XX,n,MixMonitor(/tmp/Para_${CALLEDNUMBER}-${calltime}-De_${CALLERID(num)}.wav) exten => _41XX,n,Dial(SIP/${EXTEN},40,TtRr) exten => _41XX,n,Hangup and the sip.conf looks like this: [supervisor] username=sales secret=ASUPERSECRETPASSWORD type=peer ..... context=supervisor mailbox=supervisor [sales] username=sales secret=ASUPERSECRETPASSWORD type=peer ..... context=sales mailbox=sales What do you suggest in order to get the supervisor with the same privileges that he already has and the sales been able to transfer calls to him

  Read the article

• asterisk extensions.conf & sip.conf

  - by Josh

  I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

  Read the article

• Route through site-to-site VPN not working

  - by Jonathan

  I'm trying to set up a site-to-site VPN using RRAS on two 2K8r2 servers since yesterday. The connection is working at this point, but I can't get it to send traffic from one site to the other one. Set up: the set up is the same on both sites: the server is connected to a router that's connected to a modem. The routers act like a DHCP-server and assign IP addresses from the range subnet.21-subnet-.100. Both servers use a static IP address, subnet.11, and are set up as DMZ. Configuration: the servers are configured using the wizard to set up a site-to-site connection. This works with a demand-dial interface and a PPTP VPN connection. As mentioned, the VPN connection work properly. Problem: I can't get the servers to send the traffic for the other site, to be sent through the VPN connection. I added a static route on both server (home, office 1) and I can see the result in the IP routing table (home, office 1). I did this because the route didn't show up automatically. My guess is that this last step isn't right, for example because the routing table states "non demand-dial", which seems not correct. Home: Subnet: 10.0.1.0/24 Router: 10.0.1.1 Server: 10.0.1.11 (DMZ) DHCP: 10.0.1.21-10.0.1.100 RRAS DHCP: 10.0.1.101-10.0.1.150 Office 1: Subnet: 10.0.2.0/24 Router: 10.0.2.1 Server: 10.0.2.11 (DMZ) DHCP: 10.0.2.21-10.0.2.100 RRAS DHCP: 10.0.2.101-10.0.2.150 I hope someone has an idea to get this route working!

  Read the article

• Windows VPN for remote site connection drawbacks

  - by Damo

  I'm looking for some thoughts on a particular way of setting up a estate of machines. We have a requirement to install machines into unmanned, remote locations. These machines will auto login and perform tasks controlled from a central server. In order to manage patching, AV, updates etc I want these machines to be joined to a dedicated domain for this estate. Some of the locations will only have 3G connectivity (via other hardware), others will be located on customer premises in internal networks. The central server (of ours) and the Domain Controller will be on a public WAN. I see two ways of facilitating this. Install a router at each location and have a site to site VPN between the remove device and the data centre where the servers are location Have the remote machine dial up and authenticate via a Windows VPN connection to the DC via RAS Option one is more costly to setup and has a higher operational cost. It also offers better diagnostics if the remote PC goes down. Option two works well but is solely dependent on the VPN connection been made before any communication can be made to the remote machine. In a simple test, I can got a Windows 7 machine to dial a VPN prior to authentication to a domain, then automatically login to the machine using domain credentials. If the VPN connection drops, it redials. I can also create a timed task to auto connect every hour in case of other issues. I'd like to know, why (if at all) is operating a remote network of devices which are located in various out of band locations in this way a bad idea? Consider 300-400 remote machines all at different sites. I'd rather have 400 VPN connections to a 2008 server than 400 routers, however I'd like to know other opinions on this.

  Read the article

• A router that supports connecting with 2 different wifi networks

  - by Allan Deamon

  I Have the following setup in one place: We have a small local ISP through wireless. I have a external parabolic antenna, connected to a external usb wifi radio, connected through USB to a desktop old PC. The pc connects do the ISP wiki network, then do a Dial Up (PPPoE) connection through the this wifi setup. This will expand with others mobiles devices to be used. When I need, I take my home wireless router and connect though Ethernet in the PC, which is shares the internet. The problem is that the PC must be always ON and working. I would like to buy a wireless router which could be an AP to the mobile devices, notebooks, etc, as also could connect to the ISP Wifi/PPPoE network. So, this device must: Have one radio with detachable antenna to connect to the external antenna. It must connect as client to a network and then dial up the PPP Have another radio serving as AP (infrastructure) to the local place This can't be very expensive. I found a candidate: ( http://www.tp-link.com/en/products/details/?categoryid=1682&model=TL-WR2543ND ) It have 3 deatachable antennas, working with dual band. Officially, his firmware doesn't support it. My supposition: If internally there is 3 or 2 distinct wlan ports (like wlan0, wlan1), and there is support, i could use a OpenWRT, DD-WRT or Tomato to make this works. It also have 1 USB port, which I cold use to connect my actual USB Wifi card on it instead to the old PC. Another alternative, is a router that can do this out of box, with the original firmware. But I don't think this is a easy thing to find.

  Read the article

• Site-to-Site PPTP VPN connection between two Windows Server 2008 R2 servers

  - by steve_eyre

  We have two Windows Server 2008 R2 machines, one in our main office and one in a new office which we have just moved offsite. The main office has previously been handling client-to-server PPTP VPN connections. Now that we have moved our second server out of office, we want to set up a demand-dial or persistent VPN connection from the second server to the primary. Using a custom setting RRAS profile, we have successfully managed to set up a site-to-site VPN connection so that from the second server itself, it can access any of the devices in the main office and communicate back. However, any connected machines in the second office cannot use this connection, even when using the second server as gateway. The demand-dial interface is setup from the Second Server dialing into Main Server and a static route set up on RRAS for 192.168.0.0 with subnet mask 255.255.0.0 pointing down this network interface. The main office has the network of 192.168.0.0/16 (subnet mask 255.255.0.0). The second office has the network of 172.16.100.0/24 (subnet mask 255.255.255.0). What steps do we need to take to ensure traffic from the second office PCs going towards 192.168.x.x addresses use the VPN route? Many Thanks in advance for any help the community can offer. Debug Information Here is the route print output from the second server: =========================================================================== Interface List 23...........................Main Office 22...........................RAS (Dial In) Interface 16...e0 db 55 12 fa 02 ......Local Area Connection - Virtual Network 1...........................Software Loopback Interface 1 12...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter 14...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #2 24...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #3 =========================================================================== IPv4 Route Table =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 172.16.100.250 172.16.100.222 261 127.0.0.0 255.0.0.0 On-link 127.0.0.1 306 127.0.0.1 255.255.255.255 On-link 127.0.0.1 306 127.255.255.255 255.255.255.255 On-link 127.0.0.1 306 <MAIN OFFICE IP> 255.255.255.255 172.16.100.250 172.16.100.222 6 172.16.100.0 255.255.255.0 On-link 172.16.100.222 261 172.16.100.113 255.255.255.255 On-link 172.16.100.113 306 172.16.100.222 255.255.255.255 On-link 172.16.100.222 261 172.16.100.223 255.255.255.255 On-link 172.16.100.222 261 172.16.100.224 255.255.255.255 On-link 172.16.100.222 261 172.16.100.225 255.255.255.255 On-link 172.16.100.222 261 172.16.100.226 255.255.255.255 On-link 172.16.100.222 261 172.16.100.227 255.255.255.255 On-link 172.16.100.222 261 172.16.100.228 255.255.255.255 On-link 172.16.100.222 261 172.16.100.229 255.255.255.255 On-link 172.16.100.222 261 172.16.100.230 255.255.255.255 On-link 172.16.100.222 261 172.16.100.255 255.255.255.255 On-link 172.16.100.222 261 192.168.0.0 255.255.0.0 192.168.101.87 192.168.101.17 266 192.168.101.17 255.255.255.255 On-link 192.168.101.17 266 224.0.0.0 240.0.0.0 On-link 127.0.0.1 306 224.0.0.0 240.0.0.0 On-link 172.16.100.222 261 224.0.0.0 240.0.0.0 On-link 172.16.100.113 306 224.0.0.0 240.0.0.0 On-link 192.168.101.17 266 255.255.255.255 255.255.255.255 On-link 127.0.0.1 306 255.255.255.255 255.255.255.255 On-link 172.16.100.222 261 255.255.255.255 255.255.255.255 On-link 172.16.100.113 306 255.255.255.255 255.255.255.255 On-link 192.168.101.17 266 =========================================================================== Persistent Routes: Network Address Netmask Gateway Address Metric 0.0.0.0 0.0.0.0 192.168.0.200 Default 0.0.0.0 0.0.0.0 172.16.100.250 Default =========================================================================== IPv6 Route Table =========================================================================== Active Routes: If Metric Network Destination Gateway 1 306 ::1/128 On-link 16 261 fe80::/64 On-link 16 261 fe80::edf4:85c6:3c15:dcbe/128 On-link 1 306 ff00::/8 On-link 16 261 ff00::/8 On-link 22 306 ff00::/8 On-link =========================================================================== Persistent Routes: None And here is the route print from one of the second office PCs: =========================================================================== Interface List 11...10 78 d2 32 53 27 ......Atheros AR8151 PCI-E Gigabit Ethernet Controller 1...........................Software Loopback Interface 1 12...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter 13...00 00 00 00 00 00 00 e0 Teredo Tunneling Pseudo-Interface =========================================================================== IPv4 Route Table =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 172.16.100.250 172.16.100.103 10 127.0.0.0 255.0.0.0 On-link 127.0.0.1 306 127.0.0.1 255.255.255.255 On-link 127.0.0.1 306 127.255.255.255 255.255.255.255 On-link 127.0.0.1 306 172.16.100.0 255.255.255.0 On-link 172.16.100.103 266 172.16.100.103 255.255.255.255 On-link 172.16.100.103 266 172.16.100.255 255.255.255.255 On-link 172.16.100.103 266 224.0.0.0 240.0.0.0 On-link 127.0.0.1 306 224.0.0.0 240.0.0.0 On-link 172.16.100.103 266 255.255.255.255 255.255.255.255 On-link 127.0.0.1 306 255.255.255.255 255.255.255.255 On-link 172.16.100.103 266 =========================================================================== Persistent Routes: None IPv6 Route Table =========================================================================== Active Routes: If Metric Network Destination Gateway 1 306 ::1/128 On-link 11 266 fe80::/64 On-link 11 266 fe80::e973:de17:a045:aa78/128 On-link 1 306 ff00::/8 On-link 11 266 ff00::/8 On-link =========================================================================== Persistent Routes: None

  Read the article

• PSTN Trunk TDM400P Install on Asterisk / Trixbox

  - by Jona

  Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following output trixbox1.localdomain ~]# setup-pstn -------------------------------------------------------------- Detecting PSTN cards and USB PSTN Devices -------------------------------------------------------------- Hardware present! STOPPING ASTERISK Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] wcte11xp: [ OK ] wctdm24xxp: [ OK ] opvxa1200: [ OK ] wcfxo: [ OK ] wctdm: [ OK ] wcb4xxp: [ OK ] wctc4xxp: [ OK ] xpp_usb: [ OK ] Running dahdi_cfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MOH Interpret Blocked State pseudo default en default In Service 1 from-pstn en default In Service dahdi_scan returns: dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 10 basechan=1 totchans=4 irq=209 type=analog port=1,FXO port=2,none port=3,none port=4,none And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook A cat of /etc/asterisk/dahdi.conf shows: [trixbox1.localdomain ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Tue May 25 17:45:13 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 callerid= group= context=default I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". I have one outbound route which uses the dial pattern 9|. and the Trunk Zap/1 and one Zap Trunk which uses Zap Identifier (trunk name): 1 and has no Dial Rules. The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. When running tail -f /var/log/asterisk/full and placing a call I get the following output: [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP TOS bits 184 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP CoS mark 5 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP TOS bits 136 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP CoS mark 6 [May 26 11:10:52] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/801-b7ce8c28", "1?continue") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-user-callerid,s,20) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/801-b7ce8c28", "1?check") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-record-enable,s,4) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/801-b7ce8c28", "recordingcheck,20100526-111052,1274868652.1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [May 26 11:10:52] VERBOSE[2858] logger.c: recordingcheck,20100526-111052,1274868652.1: Outbound recording not enabled [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28>AGI Script recordingcheck completed, returning 0 [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/801-b7ce8c28", "1?nomax") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s,9) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/801-b7ce8c28", "1?normcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,6) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/801-b7ce8c28", "1?trunkcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,12) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/801-b7ce8c28", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/801-b7ce8c28", "DAHDI/1/01483890915,300,") in new stack [May 26 11:10:52] WARNING[2858] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [May 26 11:10:52] VERBOSE[2858] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/801-b7ce8c28", "s-CHANUNAVAIL,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/801-b7ce8c28", "1?noreport") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/801-b7ce8c28", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:1] Playback("SIP/801-b7ce8c28", "all-circuits-busy-now,noanswer") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'all-circuits-busy-now.ulaw' (language 'en') [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:2] Playback("SIP/801-b7ce8c28", "pls-try-call-later,noanswer") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'pls-try-call-later.ulaw' (language 'en') [May 26 11:10:54] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/801-b7ce8c28' in macro 'outisbusy' [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (from-internal, 901483890915, 5) exited non-zero on 'SIP/801-b7ce8c28' [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [h@from-internal:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/801-b7ce8c28", "1?skiprg") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,6) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/801-b7ce8c28", "1?skipblkvm") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,9) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/801-b7ce8c28", "1?theend") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,11) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-b7ce8c28' in macro 'hangupcall' [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-b7ce8c28' I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

  Read the article

• Oracle Customer Reference Forum – Apex IT – Oracle Sales Cloud

  - by Richard Lefebvre

  Normal 0 false false false EN-US X-NONE X-NONE MicrosoftInternetExplorer4 Apex IT, an Oracle Platinum Partner, wins Nucleus Research's ROI Award with a 724% return. Learn how you can improve your ROI with Oracle Sales and Marketing Cloud. We are pleased to invite you to a discussion with Apex IT on industry trends, why sales automation is important, the decision making process for choosing Oracle Sales Cloud, and benefits achieved since going live. Apex IT works with clients large and small, assisting them at all stages in the process: organizing ideas and developing strategies, selecting the most appropriate package, implementing it for best results, and keeping systems optimized with long-term support. Please plan to register at least three hours prior to the event taking place in order to participate and get the dial-in information associated in due time. Speakers: Bryan Hinz, Vice President of Business Development, Apex IT (Speaker) Chris Haven, Senior Director Product Management, Oracle (Moderator) Organization Profile: Since 1997, Apex IT has helped public sector, corporate and higher education clients use technology to streamline their processes and increase productivity and profitability. Based on products and best practices from Oracle our experts provide a full range of enterprise solutions including CX/CRM and related applications that support marketing, sales, and service; HR and HR Helpdesk; and Business Intelligence. Our project approach is results-driven and our attitude is people-focused. Industry: Professional Services Products/Services: Oracle Sales Cloud Organization Website: http://apexit.com/ Event Description: In this informal reference call, you will have the opportunity to hear Apex IT discuss industry trends, why sales automation is important, the decision making process for choosing Oracle Sales Cloud, and benefits achieved since going live. The call will open with a brief overview, followed by discussion, and an open question and answer session. Please allow one hour for the call. Why Oracle: Apex IT needed a mobile-enabled sales force automation tool that could promote account collaboration and integrate with Microsoft Outlook. Oracle Sales Cloud met these needs and Apex IT’s requirements for: Improved collaborative selling Improved quality of customer engagement and information Improved business development Improved pipeline management Please plan to register at least three hours prior to the event taking place in order to participate and get the dial-in information associated in due time. After you register your information will be forwarded through an Approval Process. Once your registration request has been validated against the invitation database, you will receive an email confirmation with your registration details as long as there is availability. Please be advised that Apex IT will revise the registrants list and may dismiss registrations as they see fit. Note: To access more information at the corporate site you would need an Oracle.com account. If you do not already have an account, getting one is easy and free. Click on the link and you will be prompted to create an account. After you have created your account, you will be automatically returned to the full page description of this event. Register Now! /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin-top:0cm; mso-para-margin-right:0cm; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0cm; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;}

  Read the article

< Previous Page | 2 3 4 5 6 7 8 9 10 11 12 13  | Next Page >