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  • Best Guest OS for running freeswitch under Proxmox

    - by Frank Waller
    We have a lot of asterisk Systems running on dedicated machines and would like to use freeswitch to replace a number of them. One of the advantages of freeswitch is supposed to be that it is doing much better in a virtualized environment than asterisk is. However I can find very little information about people using it in Proxmox containers. I would like to know if anyone has seen any ready to run proxmox images that include freeswitch so we can test a number of things without having to deeply go into creating our own. Or at least a clue to which system images/distro images we can use to quickly get it installed not having to deal with too many dependency or different Linux version issues. Just to be clear: It should be for a more or less current Proxmox and current freeswitch. I am not looking forward to use the KVM mode but would consider it, if it is otherwise ready to run out of the box. I would rather use a real OpenVZ based container. Thanks for anyone helping!

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  • asterisk/freeswitch in nat/no-nat setup

    - by pQd
    hi, my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party voip providers [ since it's in different countries - those are different providers, but that's irrelevant ]. i was thinking about terminating sip calls on something like asterisk/freeswitch server and having all sip-devices log on just once to such server[s] - mostly to provide things like voicemail, groupcalls, redirections etc. it seems perfectly doable but there is one problem - i cannot find examples how to prepare for nat/no nat. for calls routed to from/to 3rd party voip operator - i'll need handling for nat/stun etc, but for handling of internal calls - i do not want any nat, all traffic should go via vpns to different branches. can you provide me some hints how to configure it? any tutorials? thanks!

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  • Call issue with Freeswitch

    - by gbraad
    I am testing the following with Freeswitch and different devices (nokia n900, nokia e60, ekiga) and have similar results between them. On the Freeswitch server (1.0.4 in multi-tenant mode) I have several user profiles for a domain, e.g. 1000, 1001 for host.com The user are authenticated correctly and calls can be placede. When I place a call from a device registered as [email protected] to [email protected] it will show up at the other end (1002) as [email protected] I would expect this call to show up as [email protected]. The IP address is the one of from the Freeswitch server. Because of this, the calls are no correctly recognized by the address book on certain devices. Can the he domain FQDN of the callers domain/acount be used, instead of the IP address of the server in the SIP uri?

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  • Can I set up an "involuntary" conference call with Freeswitch?

    - by Atilla Filiz
    I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to call all of them and make announcements. A way to work around seems creating a conference and making all UAs to join via some RPC mechanism but I don't want to go that direction unless I have to. The slave UAs are linphone and I haven't decided on the master agent yet.

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  • free switch : what is tls_port ?

    - by kiruthika
    Hi all, I am beginner to free switch.I have gone through the configuration file vars.xml in free switch. In this I have seen the following configurations. <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5070"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5071"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/> In the above I am having the doubt with tls_port. What is the use of tls_port .I have searched about this in net and I have read that tls protocol is used for secure data transfer in network. So please explain me about the communication in freeswitch. Thanks in advance.

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  • Using an extension to block a caller

    - by Trewq
    I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I typically hang up. I think want to dial some number (like *23 or something) and I'd like the last number that was received to be put in a database. Any future call from that number will be directed to VM or a busy tone. I'd appreciate some pointers on how I'd go about doing this.. I prefer an open source solution.

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  • Is it possible to have my desktop loaded before login?

    - by Dims
    I am connecting to the system with Putty (this is Windows SSH client) then running some service in interactive personal mode. For example this is a script to run freeswitch: dims@nebulla:~$ cat freeswitch.sh #!/bin/sh cd ~/bin/freeswitch/bin gnome-terminal -e ./freeswitch & I.e. it is installed in user directory. Also DISPLAY is set to :0 The problem is that I can't use this script until login once. Script responds with dims@nebulla:~$ ./freeswitch.sh dims@nebulla:~$ No protocol specified Failed to parse arguments: Cannot open display: After login, I can do "Switch user" and see login screen but script will be able to run since desktop exists. My question is: is it possible to "preload" my desktop, so that initial situation was as if I loogged in and the went to switch user?

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  • Getting RINGING response on SIP UAC without sending it from the other UAC

    - by TacB0sS
    Hi, I hope this would be my last question about this SIP subject, I have managed to overcome the last issue I had by asking a friend to help me from a remote computer, I'm able to connect between the computers, but here is the thing, according to all the examples I saw, the Callee should invoke the Ringing response, but in my application case I didn't implement it yet, but I still receive on the Caller UAC a Ringing response, this is the SIP messages that are on the caller end: Outgoing Request 5: INVITE sip:[email protected] SIP/2.0 Contact: "Client 310" <sip:[email protected]> From: "Client 310" <sip:[email protected]> Max-Forwards: 32 CSeq: 2 INVITE Call-ID: [email protected] Allow: INVITE,CANCEL,ACK,BYE,OPTIONS Content-Type: application/sdp Proxy-Authorization: Digest username="310",nonce="012afffb",realm="asterisk",uri="sip:[email protected]",algorithm=MD5,response="d19ca5b98450b4be7bd4045edb8a3a2f" Via: SIP/2.0/UDP hostName.hn:5060 To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Content-Length: 257 v=0 o=310 7108915969559970847 7108915969559970847 IN IP4 xxx.xxx.x.xxx s=- i=Nu-Art Software - TacB0sS VoIP information c=IN IP4 xxx.xxx.x.xxx m=audio 3312 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Incoming Response 6: SIP/2.0 100 Trying Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 Incoming Response 7: SIP/2.0 180 Ringing Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 Call to: [email protected] is Ringing Incoming Response 8: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 264 v=0 o=root 27669 27669 IN IP4 yy.yy.yy.yy s=session c=IN IP4 yy.yy.yy.yy t=0 0 m=audio 10914 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Incoming Response 9: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 I do not respond to the invite, that is why all this is happening, but why am I getting a ringing if I'm not the one sending it. Thanks, Adam.

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  • Why are SIP calls via my server silent?

    - by Archcode
    I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually). I can connect to it, make a call to other endpoint (namely, my android device to my pc and vice versa) and signals are send with no problems (call, answer, hangup, etc). Unfortunately, and what drives me crazy, that's all: no audio gets through, no video either. Server does not throw errors, it reports many retransmission though, looks like this: switch_rtp.c:915 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=15 Codecs are set up correctly (same config worked locally on my LAN). NAT/firewall on client side may be a problem, signals do get through (perhaps due to fixed port, data streaming runs on random one, that is currently my best bet). STUN/TURN/ICE setting on client seem to have no effect. Endpoints sit behind symmetric NAT. On server there are no iptables rules, security group is set as suggested there: http://wiki.freeswitch.org/wiki/Firewall Help, please. How to make it work or at least diagnose what's wrong?

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  • Build a ruby daemon that integrates my rails environement

    - by jjmartres
    Hi guys, I need to build a ruby daemon that will use the freeswitcher eventmachine library for freeswitch. Since few days I as looking the web for the best solution to build a ruby daemon that will integrate my rails environment, specailly my active record models. I've take a look to the excellent Ryan Bates screencast (episodes 129 custom daemon) but I'm not sure that is still an actual solution. Does anyone known a good way to do that ? Thanks all for your help.

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  • IP telephony open source systems

    - by danke
    I'm trying to pick an IP telephony technology to learn. I heard of Asterisk, trixbox, freePBX, and my head was already spinning being not sure what to learn. Then I came across this article listing some more like Kamailio, Yate, CallWeaver, FreeSWITCH, SipXecs and now my head REALLY is spinning http://www.cio.com.au/article/323016/five_open_source_ip_telephony_projects_watch . Can someone give me a run down of how all these technologies tie together? What is the trend now, because I'd like to start learning. Note: Anyone please re-tag this question if you know better, because I'm new to this field and not sure about the best tags.

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  • Moving from Analogue PBX to digital VoIP?

    - by saint
    I don't even know if this belongs here?. If not, do let me know. So we have an analogue Alkatel PABX system in our little office. We have extensions, direct lines and PBX lines. We are trying to move to a more digital/flexible way of handling the phones and I've heard good things about FreeSwitch. I have zero knowledge about it. My biggest question is how would one handle existing phone lines with such a system. Surely there must be a way to make and receive calls from outside. Just a help in the right direction would be fine. Thanks.

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  • rsync server side limit bandwidth/connection

    - by c2h2
    In a VOIP application, I have upto 3000 clients rsync audio files from there linux server in a daily, server is placed at a data center (10Mbps in/out bound), the server works as a VOIP sip server running FreeSWITCH (low ping latency should be ensured.) Therefore I would like to have server side control of rsync which controls: Limit total outbound bandwidth. Limit total number of connections. (Reject clients while at max number of connection and let it retry after a specific time frame.) OPTIONAL: list/kill individual connections. Normally I would use ssh + rsync + pem_keys with some extra options, but above requirements are not feasible by simple command lines. Can anyone point me some direction. or show some scripts/tools? I would also probably integrate them and release on github. Thanks!

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  • SIP and NAT routers?

    - by OverTheRainbow
    Hello SIP was not built with NAT routers in mind, and I'd like to get to the bottom of this issue to check what needs to be done on all devices so it works with NAT routers, and understand in what context it just can't be used and I should check more NAT-friendly alternatives like IAX. A picture being worth a thousand words, here's the layout I need to use: http://img62.imageshack.us/img62/4077/sipandnatrouters.jpg The PBX server is located in the private LAN behind a NAT router connected to the Internet (I know it'd be easier if it were located in the public network, but this router doesn't support DMZ's so the server has to be in the private network) A couple of (soft|hard)phones are located on the same LAN and connected to the PBX server, along with a PSTN gateway (Linksys 3102 or a Digium PCI card) Remote users using (soft|hard)phones are located somewhere on the Net with dynamic IP's and are also located behind NAT routers I may or may not have control over the local NAT router where the PBX server is located, but I have no control over the remote NAT routers, either because the users don't have the computer knowledge to map ports or because the routers are off-limit (eg. web cafés, hotel LAN's, etc.) Is it possible to configure the PBX server, the (soft|hard)phones, and the PSTN gateway so that the all conversations work fine, no matter the endpoints (POTS caller/local phone, POTS caller/remote phone, local phones, remote phone/local phone)? In which cases may I expect problems, and are there solutions? FWIW, I'm leaning toward using Freeswitch, but I could end up using Asterisk if there are technical advantages to it in this context. Thank you for any info.

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  • SIP UAS asks for OPTIONS

    - by TacB0sS
    Hey, I have UAC that registers to a UAS, after registration the UAS sends me an OPTIONS request, what should I answer it? only the audio media streams? Update I: Allow me to explain myself better... if I want to invite someone to a session I USE the INVITE method and negotiate the media then, for that specific session. But once I register to the server, and it asks me for OPTIONS, then what should I supply, everything my client supports? once I answer it would it deduce that every INVITE I would request from now on would use these medias? or would I need to supply new media with every request? Update II: Hi Wiz, I was in the process of building a negotiation system, so i tried it out and replied the UAS here is the sort dialog we had: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK45b197cb;rport=5060;received=xx.xx.xx.xx From: "Unknown" <sip:[email protected]>;tag=as66cf26df To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Freeswitch 1.2.3 Max-Forwards: 70 Date: Sat, 05 Jun 2010 12:06:43 GMT Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 OPTIONS In Response To 102: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK45b197cb;rport=5060;received=xx.xx.xx.xx From: "Unknown" <sip:[email protected]>;tag=as66cf26df To: <sip:[email protected]> CSeq: 102 OPTIONS Call-ID: [email protected] Allow: INVITE,CANCEL,ACK,BYE,OPTIONS Content-Type: application/sdp Content-Length: 248 v=0 o=310 4515233118481497946 4515233118481497946 IN IP4 10.0.0.1 s=- i=Nu-Art Software - TacB0sS VoIP information c=IN IP4 10.0.0.1 m=audio 40000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 This response caused the server to stop sending me the options request, does this means I can only use these parameters with the server now? or as you said, it does not matter? Thanks, Adam.

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  • Audio/video streaming on Windows platform

    - by bushtucker
    I'm building an interactive language learning application to be used in a classroom environment. The idea is that a teacher should be able to talk to the students (=audio stream to all students), let students talk to each other (= audio P2P) in groups of two or more, let students watch a video coming from a the DVD player or coming from a media server. It should be possible to save the audio/video streams. The teacher should also be able to monitor, take-over or block the desktop of the students. The platform is Windows and it's a desktop application, no web application. The audio delay should be as minimal as poosible. Optionally a student sitting at home should be supported, but it's not a high priority. I am now finished with the classroom control part of the application (login, monitor, block, ...) and want to start the audio and video part. I've been evaluating several options like DirectX, GStreamer and SIP but now I have to make a decision. DirectX seems an obvious choice for the Windows platform, but it only lets me capture and playback audio and video. The encoding/decoding/network part I should do myself. GStreamer contains all kinds of options to capture/encode/stream/save audio and video streams. I've experimented a bit with it (ossbuild) and it does seem to involve a lot of trial and error to make something work: - microphone capture (via directsoundsrc) produces cracking noises on some computers - rtpL16 payloader didn't work well - streaming raw audio over the network only working at a sampling rate of 8000, no higher - there are a lot of errors when receiving mpeg4 video (bad I-frame), on some computers worse than others It is my impression that gstreamer is primary targetted at linux platforms. Development and support for the Windows platform seems to be a little behind. Nevertheless it's a powerful framework that could save me months and years of work. SIP seems to be able to do everything I want, but it is targeted towards telephony and IM. I don't know how flexible SIP is. It seems to me that the SIP layer would just be overhead as I already have a central (teacher) application that can control and setup all the streams. The interesting parts of frameworks like opalvoip and freeswitch are the actual audio/video capture, the encoding and transmission. Does anyone know how these interesting parts relate a framework like gstreamer? Are they easy to integrate into a custom application? Are they flexible enough? Does anyone have experience with all or one of these technologies? Maybe there are even other options I can look at? Many thanks for your advice

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  • CodePlex Daily Summary for Sunday, April 11, 2010

    CodePlex Daily Summary for Sunday, April 11, 2010New ProjectsArkzia: This Silverlight game.CodePlex Wiki Editor: CodePlex Wiki Editor makes it easier for CodePlex users to create their wiki documentations. This project offer a rich interface for the edition...Evaluate: Evaluate & Comet DWR like .NET library with powerfull Evaluate and Ajax Comet support. Also, you may use Evaluate library in your own .Net applicat...FamAccountor: 家庭记账薄Horadric: This is common tools freamwork!K8061.Managed: This is a solution to use the Velleman Extended K8061 USB Interface board with .net and to have a nice wrapper handling most of the overhead for us...Latent semantic analysis: all you need is to use!: Baggr is feed aggregator with web interface, user rating and LSA filter. Enjoy it!LIF7 ==> RISK : TOWER DEFENSE: Université Lyon 1, L2 MATH-INFO 2009-2010 Semestre de printemps Projet RISK : TOWER DEFENSE Membres : Jessica El Melhem, Vincent Sébille, et Jonat...Managed ESL Suite: Managed ESL Suite using C# for FreeSWITCH Omni-Tool - A program version concept of the tool used in Mass Effect.: A program version concept of the tool used in Mass Effect. It will support little apps (plugins) that run inside the UI. Its talor mainly at develo...PdxCodeCamp: Web application for Portland Code CampProjeto Vírus: Desenvolvimento do Jogo Virus em XNAsilverlight control - stars with rounded corners: Draw stars and cogs including rounded cornersSilverlight MathParser: Implementation of mathematical expressions parser to compute and functions.turing machine simulator: Project for JCE in course SW engeenering. Turing Machine simulator with GUI.WpD - Wallpapers Downloader: You can easy download wallpapers to your computer without any advertising or registration. On 5 minutes you can download so many wallpapers!New ReleasesAJAX Control Framework: v1.0.0.0: New AJAX project that helps you create AJAX enabled controls. Make use of control level AJAX methods, a Script Manager that works like you'd expect...AutoFixture: Version 1.1: This is version 1.1 of AutoFixture. This release contains no known bugs. Compared to Release Candidate 1 for version 1.1, there are no changes. Ho...AutoPoco: AutoPoco 0.3: Method Invocation in configuration Custom type providers during configuration Method invocation for generationBacicworx (Basic Services Framework): 3.0.10.410 (Beta): Major update, winnowing, and recode of the library. Removed redundant classses and methods which have similar functionality to those available in ...Bluetooth Radar: Version 1.5: Mostly UI and Animation Changes.BUtil: BUtil 4.9: 1. Icons of kneo are almost removed 2. Deployment was moved to codeplex.com 3. Adding of storages was unavailable when any of storages are used FIXEDcrudwork library: crudwork 2.2.0.3: few bug fixes new object viewer - allow the user to view and change an object through the property grid and/or the simple XML editor pivot table ...EnhSim: Release v1.9.8.5: Release v1.9.8.5Removed the debugging output from the Armor Penetration change.EnhSim: Release v1.9.8.6: Release v1.9.8.6Updated release to include the correct version of EnhSimGUIEvaluate: Evaluate Library: This file contains Evaluate library source code under Visual Studio project. Also, there is a sample project to see the use.ExcelDna: ExcelDna Version 0.25: This is an important bugfix release, with the following changes: Fix case where unpacked .config temp file might not be deleted. Fix compiler pro...FamAccountor: 家庭账薄 预览版v0.0.1: 家庭账薄 预览版v0.0.1 该版本提供基本功能，还有待扩展! Feature: 实现基本添加、编辑、删除功能。FamAccountor: 家庭账薄 预览版v0.0.2: 家庭账薄 预览版v0.0.2 该版本提供基本功能，还有待扩展! Feature: 添加账户管理功能。Folder Bookmarks: Folder Bookmarks 1.4.2: This is the latest version of Folder Bookmarks (1.4.2), with general improvements. It has an installer - it will create a directory 'CPascoe' in My...GKO Libraries: GKO Libraries 0.3 Beta: Added Silverlight support for Gko.Utils Added ExtensionsHash Calculator: HashCalculator 1.2: HashCalculator 1.2HD-Trailers.NET Downloader: Version: TrailersOnly if set to 'true' only titles with 'trailer' in the title will be download MinTrailerSize Added a minimum trailer size, this avoids t...Home Access Plus+: v3.2.6.0: v3.2.5.1 Release Change Log: Add lesson naming Fixed a bug in the help desk which was rendering the wrong URL for tickets Planning has started ...HTML Ruby: 6.20.0: All new concept, all new code. Because this release does not support complex ruby annotations, "Furigana Injector" is not supported by this release...HTML Ruby: 6.20.1: Fixed problem where ruby with closed tags but no rb tag will result in empty page Added support for complex ruby annotation (limited single ruby)...K8061.Managed: K8061.Managed: This is a pre-compilled K8061.Managed.DLL file release 1.0.Kooboo CMS: Kooboo CMS 2.1.0.0: Users of Kooboo CMS 2.0, please use the "Check updates" feature under System to upgrade New featuresWebDav support You can now use tools like w...Kooboo forum: Kooboo Forum Module for 2.1.0.0: Compatible with Kooboo cms 2.1.0.0 Upgrade to MVC 2Kooboo GoogleAnalytics: Kooboo GoogleAnalytics Module for 2.1.0.0: Compatible with Kooboo cms 2.1.0.0 Upgrade to MVC 2Kooboo wiki: Kooboo CMS Wiki module for 2.1.0.0: Compatible with Kooboo cms 2.1.0.0 Upgrade to MVC 2Mavention: Mavention Simple SiteMapPath: Mavention Simple SiteMapPath is a custom control that renders breadcrumbs as an unordered list what makes it a perfect solution for breadcrumbs on ...MetaSharp: MetaSharp v0.3: MetaSharp v0.3 Roadmap: Oslo Independence Custom Grammar library Improved build environment dogfooding Project structure simplificationsRoTwee: RoTwee (10.0.0.7): New feature of this version is support for mouse wheel. You can rotate tweets rotating mouse wheel.silverlight control - stars with rounded corners: first step: These are the first examples.Silverlight MathParser: Silverlight MathParser 1.0: Implementation of mathematical expressions parser to compute and functions.SimpleGeo.NET: SimpleGeo.NET example website project: ConfigurationYou must change these three configuration values in AppSettings.config: Google Maps API key: for the maps on the test site. Get one he...StickyTweets: 0.6.0: Version 0.6.0 Code - PERFORMANCE Hook into Async WinInet to perform async requests without adding an additional thread Code - Verify that async r...System.Html: Version 1.3; fixed bugs and improved performance: This release incorporates bug fixes, a new normalize method proposed by RudolfHenning of Codeplex.VCC: Latest build, v2.1.30410.0: Automatic drop of latest buildVFPX: FoxTabs 0.9.2: The following issues were addressed: 26744 24954 24767Visual Studio DSite: Advanced Guessing Number Game (Visual C++ 2008): A guessing number game made in visual c 2008.WpD - Wallpapers Downloader: WpD v0.1: My first release, I hope you enjoyMost Popular ProjectsWBFS ManagerRawrASP.NET Ajax LibraryMicrosoft SQL Server Product Samples: DatabaseAJAX Control ToolkitSilverlight ToolkitWindows Presentation Foundation (WPF)ASP.NETMicrosoft SQL Server Community & SamplesFacebook Developer ToolkitMost Active ProjectsRawrnopCommerce. Open Source online shop e-commerce solution.AutoPocopatterns & practices – Enterprise LibraryShweet: SharePoint 2010 Team Messaging built with PexFarseer Physics EngineNB_Store - Free DotNetNuke Ecommerce Catalog ModuleIonics Isapi Rewrite FilterBlogEngine.NETBeanProxy

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