I was having problems with my trixbox 1.2 installation, two of them, VMware and plain old direct install. I was not alone and I happened to stumble upon freepbx.org's Rob Thomas' post Un-trixbox your trixbox.
I have had problems with Trixbox 1.2, that queues not working, asterisk not reloading, random dropouts, stuttering, clients not being able to connect. And what do I read on freepbx article, solution to all these problems and more. The article really helped me and I wand to direct anyone with similar trixbox 1.2 woes to his article. According to Rob, scheme should work flawlessly for any trixbox-1.2 based machine.
One thing you should be aware that this will stop the usual trixbox stuff from being updated. This includes HUDlite, SugarCRM, any of the non-CentOS stuff. This isnÂt all that much of a hassle, as per Rob. It was not a hassle for me. He plans to update his article, I think after his daughters B'day!
Saturday, September 30, 2006
I was having problems with my trixbox 1.2 installation, two of them, VMware and plain old direct install. I was not alone and I happened to stumble upon freepbx.org's Rob Thomas' post Un-trixbox your trixbox.
Thursday, September 28, 2006
Via Business wire;
Ixia XXIA, a leading, global provider of IP performance test systems, announced today the launch of IxVoice Lite, a low-cost software-only version of IxVoice for testing IMS and VoIP protocol implementations. IxVoice Lite, announced at NetEvents in Faro, Portugal, contains the same features as IxVoice for testing up to 24 channels. It can easily be downloaded onto workstations and provides the same "look and feel" and operation as the traditional version. Instead of sharing a workstation, individual users can now have their own version of IxVoice running on their workstations saving time and valuable resources.
IxVoice Lite, an entry-level software platform, enables developers, QA test engineers and field engineers of service providers to perform sophisticated IMS and VoIP functionality testing. With its cost-effective test libraries, IxVoice Lite addresses all major VoIP protocols: SIP, SCCP (Skinny), H.323, MGCP and H.248/MEGACO. Specifically, its SIP library can be used to test critical CSCF functionality for IMS implementations.
Posted by ravenII at 9/28/2006 10:50:00 AM
Wednesday, September 27, 2006
I can bet you will laugh if you read his blog post or listen to the MP3 that he created of the conversation. It seems that Vonage has a regular tech support team, an elite tech support team, 500 min for 24.99 and unlimited minutes for 19.99! No I did not miss type it!!
Instead of me making textual mistakes trying to explain his experience, I suggest that you head over there and read it or listen to the conversation. That will give me enough time to finish my laughing and digest the gist of the article he had written. Basically companies try ever so harder to keep a customer, when they are cannot be arrogant! Do you know when and why a company could not be arrogant? ...
But he says that the experience was not that bad.
Posted by ravenII at 9/27/2006 04:06:00 PM
The best place to start is the new documentation site devoted to SER documentation. Since Lots of new features have been added and quite a lot of important core code has been rewritten, it is important to refer to these documents that are intended to document changes from 0.9.x to 0.10.x, as well as new features in 0.10.x.
You can also add to the documentation, if you are experienced or have experience in SER and it's various modules.
What do we have at the moment;
* Attribute-value pairs and selects
* Basic changes in configuration file
* Changes to SER modules
* New Management Interface: RPC, XML-RPC
* Optimizing the use of RTP proxy
Posted by ravenII at 9/27/2006 02:59:00 PM
From Openser news:
New XMPP module - allow straightforward interconnection of SIP networks with XMPP networks (Google Talk, Jabber) for instant messaging.
A big step to converge various IM/VoIP networks world wide has been done with the new XMPP module, developed by Andreea Spirea. It allows exchange of instant messages with any XMPP network out there (like Google Talk or Jabber), opening the way to add presence and voice support in the near future.
There is no requirement of mapping SIP addresses to XMPP addresses via database of other persistent storage, the addressing schema allows translation on the fly. Just install the SIP-to-XMPP gateway and all your SIP users become available in the XMPP network and your users can chat with anybody in XMPP world. You can be even a SIP-to-XMPP relay for SIP networks you peer with, there is no limitation that only local users can use the gateway.
The conversation will survive to restarts, the session being recovered form the messages. As a result, there is no need of a persistent storage, the footprint is very small, embedding the gateway in small devices should be straightforward.
For more technical details see:
Posted by ravenII at 9/27/2006 10:34:00 AM
Tuesday, September 26, 2006
I am putting together an article explaining how to use your Asterisk PBX (You have one don't you? Or I hope you are at least thinking og getting one!) with a popular VOIP, IP telephony free service to make calls to the world (right now only 60 countries! Hint!!) for free.
I hope to make this a good article and due to my English it is taking time!
Posted by ravenII at 9/26/2006 11:06:00 AM
Sunday, September 24, 2006
It has been told many a times that SkypeÂ® tend to disrupt regular network operations, even if you are not on a call. I have noticed that on my own network and the moment all skype tasks were removed it went away. But you reboot the computer and good old skype is up to it's tasks of being a super node. This is with only one computer being used for skype and I hate to imagine a network with multiple supernodes. If you have a large network and if you have many skype users, you will have to ban using skype use in your network. Due to the way skype operates, it is hard to stop it at the edge routers or firewalls. How skype operates is an another article.
Read my own article about universities banning skype use.
If you missed reading it when you installed skype, then read this single sentence within Skype's 4,300-word end-user license agreement:
"In order to receive the benefits provided by the Skype Software, You hereby grant permission for the Skype Software to utilize the processor and bandwidth of Your computer for the limited purpose of facilitating the communication between Skype Software users."
And I hope you knew what you were agreeing to.
How does Skype jumps through firewalls and NAT routers, The Register has the explanation.
I know only of Two solutions that is capable of filtering out skype, Verso, according to Verso's website " Verso's new carrier grade application filter solves the problem by preventing undesirable, 3rd party traffic from entering acarrierss network." But I don't like them either, I don't like their business angle.
I am talking about corporate or individual users controlling their computers and networks, Verso is providing solutions to carriers and ISP to block anything that not their own solutions. Read third party!
According to Networkworld article, "According to Skype - and validated by our research - a VoIP call will consume between 24 and 128K bit/sec. When a Skype station is functioning as a relay, the bandwidth is doubled. (We found instances when calls between adjacent stations were relayed to somewhere a continent away.)". So if your connection to internet is a DSL 384/128, your total upload space might be taken by Skype!
Yes there is another site that has tools to show you the usage of VOIP/SKYPE, IPOQUE. Their product is much better. I think this is much better suited to a network administrator than to a regular user. This is a blurb from IPOQUE site about the VOIP/SKYPE info;
" Voice over IP VoIP/Skype
Voice over IP (VoIP) has become one of the most-widely used Internet applications. The VoIP filter allows companies to gain insight in and control over their employees' VoIP usage. It allows Internet service providers to offer differentiated services and to control the VoIP bandwidth consumption in their networks.
The VoIP module supports SIP and Skype. "
But the tool can do much more than that, But be aware, it is a hardware device.
But for regular users, like you and me there are other tools. Here I am describing one of my favorite tools, skypekiller and how to use it. Read on...
Removing skype is not easy. You cannot stop it from starting up, even if you have tried to disable it from the startup group, windows firewall cannot stop it, unless you are an expert on windows, it is hard to stop or control skype. Even if you uninstall it it mighstillil be lingering somewhere in your computer.
So what does a regular user save him or herself from this application? One need to remove it, there are a couple of ways that one could do it. Do some registry hacking and follow some uninstall procedures (See the end of the article for this information). Or you can remove skype, by using skypekiller.
I think it is the most easiest way for any user. And here is how to..
So I have taken liberty of introducing the application and the procedures together with screen captures.
To stop skype, remove skype or just to list skype users, you will need first to down load the skypekiller, It is free but you have to register in order to download.
Then install it on a windows workstation or a server and make sure that a connection to your network is up and ready. If it in a domain, make sure thecomputer you are installing is a domain member. This will make procedures much easier.
Now fireup SK and Select the default clean mode or Click the "Detect only" option if you just want to list SkypeÂ® installs.
Now is time to setup Filtering
You can apply filters to determine the type of target computers on which SkypeÂ® is to be removed. For example you can choose to run SkypeKiller only on your workstations.
Once filtering is setup, you need to Select Target Computers
Select target computers. You can browse to quickly add computers from your global catalog...
or Domain controllers.
Now that you have a selection of computers, it is time for Execution
Click on Execute, a progress window will display the operation.
Now is time to go and get your other work done or have cup of coffee. You can come back to check Result analysis.
SkypeKiller will indicate success or failure of the removal for each previously selected system.
Like any other network application, you also can schedule periodic runs of the SK.
To be sure, I would run a "detect only" first and record the results. Then I will pull out my network monitoring tool (Omnipeek, a free Netwotk detection tool)to check onbandwidththth usage using omnipeek personal. It is better to run these tests during peek hours. Again run Skypekiller and remove listed installations. Once the Clean step is done run the network detection tool again. You will be able to see just roughly how much bandwidth tparticularlyarer VOIP or IP Telephony application/S was/were using.
According to Skypekiller documentation, during the clean up process, following registry keys and directories are deleted. So if you are handy with your computer, you manuallyuall perform the same functions to have a skype free computer. If you still need a VOIP application that does not violate normal network rules, look at gizmo project.
In each profile (system and user) the following registry keys/values are deleted:
The following files/folders are deleted in each profile (including the "All Users" profile):
"%ProfilUtilisateur%\MY Documents\My Skype Pictures"
The SkypeÂ® install folder is also deleted:
ou "%ProfilUtilisateur%\Local Settings\Application Data\Skype\Phone"
or other paths depending if the install folder was changed during SkypeÂ´s install.
Check regularly for updates here when SK updates its application.
If you are wondering why should I remove skype, Read the following articles;
Skype explains why security evaluation omitted bug reports
Should businesses ban Skype?
Posted by ravenII at 9/24/2006 02:13:00 AM
Saturday, September 23, 2006
Silicon Valley blog reports about San Jose State University is going to stop SKYPE within the campus network. I just finished reading the document by CERN (the birthplace of the World Wide Web.) why it does not want skype on it's net work,
excerpt from the document;
Following in the footsteps of the University of California-Santa Barbara and California State University-Dominguez Hills, SJSU plans to ban use of the pioneering VoIP system on its network because installation of the client software allows machines to serve as third-party relays for phone calls, chat and messages.
Luckily, eBay, which owns Skype, is located nearby, and the university has invited some company reps to come over on Tuesday and see if there's a way to work things out. Jennifer Caukin, a spokeswoman for eBay, said Skype was looking forward to having "a direct dialogue with SJSU officials to discuss their concerns and educate them about how Skype works." And you better believe Skype will put a lot into that lesson plan, given the stakes should this get to be a trend.
Techdirt also reports on the article. With a bunch of comments. So if you have skype installed and need to remove them;
For Linux read this weblog
For windows read this weblog. Which I will rewrite soon, due to poor presentation.
Posted by ravenII at 9/23/2006 10:42:00 AM
Friday, September 22, 2006
Here's how ZuneVoIP could kick iPod's butt by ZDNet's Russell Shaw -- Richard Ozerman of CruchGear cites the "latest rumors from people in the know" that Microsoft's upcoming Zune will be VoIP-enabled.VoIP functionality could be delivered through the device's already-revealed WiFi capability. A rumored add-on attachment that fits into a dock on the Zune would, it is believed, be able to hold a mic necessary for these [...]
Posted by ravenII at 9/22/2006 01:57:00 PM
Techdirt reports about various cases of voip arrests happening all over the world. No, not for VOIP scamming, or spamming. Simply for promoting or using VOIP or IP Telephony in day to day life. It seems where government monopolies are in controls of the Phone business, is where this is happening. I remember some time back when we tried to setup phone (VOIP hybrid) system in India, it was illegal!
Anyway techdirt has links to arrests and such.
Posted by ravenII at 9/22/2006 01:31:00 PM
Thursday, September 21, 2006
Ecamm Network has demo plug-in for the Mac OS X version of Skype called Call Recorder that lets you manually start and stop recording, or it can do it automatically. Files are saved using the AAC audio format, and are given descriptive file names that include caller ID and date and time information. Calls are saved as QuickTime files; an MP3 conversion tool is also included.
Manually start/stop/pause, or automatically record all calls.
Record and save your voicemail messages.
Control file size and quality.
State-of-the-art AAC file format saves space.
Convert your calls to MP3 format.
Uncompressed option for highest quality.
Try before you buy. Download the demo.
A demo version is available for download. System requirements call for Mac OS X v10.3 or later, Skype for Mac v1.4 or later and PowerPC or Intel processor (the plug-in is a Universal Binary).
Posted by ravenII at 9/21/2006 09:44:00 AM
Wednesday, September 20, 2006
The south Texas school is boldly moving thousands of users off a Cisco VoIP platform to an open source VoIP network based on Asterisk.
SHSU is in the process of moving its 6,000 students, faculty and staff off of Cisco CallManager IP PBXs and a legacy Nortel Meridian PBX over to Linux servers running Asterisk, which includes call processing, voicemail and PSTN gateway functionality. The driver for this project was cost, says Aaron Daniel, senior voice analyst at Sam Houston State University.
"We thought that it will be more cost effective in the long run to go with an open source solution, because of the massive amounts of licensing fees required to keep the Cisco CallManager network up and running," says Aaron Daniel, who this week gave a presentation on his migration project at the VON show in Boston. In the Cisco model, each phone attached to the CallManager required a separate annual licensing fee to operate, Daniel says. In SHSU's Asterisk/Cisco model, where it will keep its existing Cisco phones but attach them to Asterisk servers on the back end, the phone licensing costs are eliminated.
SHSU so far has moved 1,600 IP phones from Cisco CallManagers to Asterisk, which runs the IETF-standard version of SIP. The Asterisk functions are spread across six redundant Dell servers: two act as redundant PSTN gateways (and are outfitted with four-port T-1 cards from Digium, which commercially distributes Asterisk); two more servers handle call processing; another set provides voicemail.
More control over the IP PBX software and servers was another reason SHSU made the Asterisk jump, Daniel says. "We felt we were more susceptible to hacks," since only Cisco-approved servers updates and patches could be installed on the Windows Server 2000-based CallManagers, he says. "We have a lot more peace of mind with the open-source system. If a bad exploit is found in SIP, we can fix it ourselves."
New via networkworld
Posted by ravenII at 9/20/2006 08:35:00 AM
Tuesday, September 19, 2006
Philip R. Zimmermann is the creator of Pretty Good Privacy, an email encryption software package. Originally designed as a human rights tool, PGP was published for free on the Internet in 1991, Has done it again.
This time with your VOIP or your IP Telephony calls. Using Zfone, one could use existing phone connection (VOIP) to make secure calls.
if you really want to run Zfone now, you need to run a software VoIP client (such as X-Lite, Gizmo, SJphone, or perhaps a software VoIP client supplied by your VoIP service provider) on your PC or Macintosh computer. But it will not run with skype due to their closed protocol
If you want to read a full test and a great article, I can point you to one. Eric Y. Chen at Voice of VOIPSA has done a super job of testing and publishing the results. You get driven through installation, test setup, man in the middle analysis and the final results or conclusion. Visit his site and be prepared to spend a bit of time.
"Zfone is very user-friendly since it hides most of the encryption mechanism from its users. Its independence from PKI and signaling makes the technology very accessible to individuals. Zfone, being a “bump in the cord”, also allows its users to keep their favorite SIP softphones without switching to an unfamiliar one. Moreover, because only the end users are involved in the key management, the service provider does not have access to any of the keys. Eavesdropping on Zfone users seems extremely difficult as the attacker would have to be present since the first call, able to forge verbal SAS verification in real time, and preferably, able to imitate voices."
Posted by ravenII at 9/19/2006 08:40:00 AM
Sunday, September 17, 2006
I was so tired f not finding the site of iptel.org, everytime I need to get some information, I always went to the berlios site. It is the developer site for IPTEL.ORG and today I clicked on the home page link just to see what is going on.
Viola! New home and it is really nice. Good work team. Now the site is more navigatable and of course all Drupal amenities are there.
But bummer, I did not, could not find my user account. But the old site is still available until complete migration is taken place.
SER is still at Version 0.9.6 Sip Express Router
SEMS is 0.9.0 Sip Express Media Server
SERWEB is 0.9.4 WWW management interface for SER
All accessible via there own web site or sub site. Well laid out and a joy to be in compared to the old site.
I was contemplating to change "My favorite SIP Router" to OpenSER. But after visiting the new IPTEL.ORG site, I will keep it and add another site link for OpenSER.
IPTEL.ORG site overview is here.
Now may be the developer group too get their act together and get the new and better SER out.
Posted by ravenII at 9/17/2006 11:58:00 AM
From PR news.
Wildgate, Inc. a California based telecommunications service provider joins DID Exchange (DIDx.org) to sell half a million DID telephone numbers over the Internet to over 3,000 other telecom companies, USA and international.
DIDX is the world’s largest telecom DID phone number wholesale market that handles DID phone number routing, billing and Enumbering over VOIP (Voice Over Internet Protocol).
DIDX service empowers the telecom company to expand from a local telephone company to a global phone company.
By joining DIDX, Wildgate, Inc. will have instant access to over 3000 global telecom web sites who are selling VOIP phone services. Wildgate phone numbers are now available to wholesale customers around the globe, without having to provide further technical assistance.
DIDX provides telecom companies like Wildgate, Inc. one universal interop and interconnect with over 3000+ other telecom companies on the DIDXchange. Each is able to buy DID, sell DID or both.
A few companies on the DIDXchange now include Sip Phone, Yahoo, Vonage and Voice Pulse.
DIDX empowers smart start up VoIP Companies to become global telecom operators. The DIDXchange offers USA and international phone numbers to the DIDX telecom membership. With 3000 + registered service provider members, the DIDX listing totals over 2 million numbers, from Australia, Japan, Malaysia, United Kingdom, USA, Canada and 40 other countries.
DIDX creates additional revenue streams for telecom carriers who take advantage of this number exchange service, whether buying or selling DID's or both.
About Super Technologies Inc:
Super Technologies Inc was incorporated in 1999 in Delaware, USA. It began with the virtualphoneline.com concept of offering an IP-based phone number on a hardware device. The company was the first, and then later winning the Best of Show award at Spring Internet World, before the Vonage revolution began.
Posted by ravenII at 9/17/2006 01:04:00 AM
Saturday, September 16, 2006
The U.S. Senate yesterday voted to include IP-based 911 measures as part of the Safe Port Act (HR 4954) that would serve to clarify jurisdictional and liability issues surrounding voice-over-IP (VoIP) providers’ role in the 911 system.
The IP-Enabled Voice Communications and Public Safety Act of 2006 containing the E911 mandates reportedly was pushed by public-safety proponent Sen. Conrad Burns (R-Mont.) and several other lawmakers as part of a managers' amendment to the Safe Port Act (H.R. 4954). The larger bill this week also was amended to include measures on a new national emergency alert/communications network and a $1 billion allocation for interoperable emergency first-responder equipment, planning and training.
National Emergency Number Association (NENA) applauded passage of the 911 items, which the organization has been lobbying for throughout the session.
“NENA remains focused on the need for a rapid deployment of E911 for VoIP services, the need for liability protection for 911 telecommunicators, the need to preserve funding levels for PSAPs and the advancement of a modernized 911 system,” NENA President Bill Munn said in a prepared statement. “This legislation builds on the important action taken by the FCC last year and advances these critical issues.
Each of the 911 items had been part of the IP-Enabled Voice Communications and Public Safety Act (S. 1063) approved by the Senate Commerce Committee in December 2005, but it was questionable whether that legislation would reach the Senate floor, according to Beltway sources. As a result, many non-controversial 911 amendments were added to the Safe Port Act passed yesterday. The legislation now will be considered in a conference committee.
If you are not sure about E911, there is an article "Understanding e911" at PRWEB news site.
Posted by raven at 9/16/2006 12:05:00 PM
Wednesday, September 13, 2006
Google has a new dig, Googletalk and eBay that is. They have signed an agreement around text-based advertising and "click-to-call" advertising. Googletalk and skype will provide the voice and eBay will produce the goods!, literally.
According to the press release, agreement includes the following;
Search and Advertising on eBay Sites Outside the U.S.
Google will become the exclusive provider of text-based advertising on eBay outside the United States. This agreement provides Google advertisers access to one of the internet's most robust online communities while enhancing the shopping experience for eBay buyers by making it easier for them to find the products they seek.
"Click-to-Call" Advertising and Technology Integration
Google and eBay also plan to integrate and launch "click-to-call" advertising functionality within eBay's U.S. and international marketplaces and Google's search platform. The click-to-call capability will allow a user to click on a link or icon within a product or service advertisement to initiate an Internet voice call to participating eBay merchants or Google advertisers directly from either company's respective sites, using Skype or Google Talk.
You can find the full text press release here.
What made me laugh is a portion of the press release;
"Click-to-call advertising is an emerging e-commerce model that brings buyers and sellers together by opening up new ways for advertisers and merchants to generate customer leads using the Internet." We were developing this in 2002. One way or the other, this is good news for All of us, Good communication between all of us.
Posted by ravenII at 9/13/2006 04:50:00 PM
Is US falling behind in the Broadband game. Broadband is important for all internet users. VOIP IP Telephony, depends a lot on your internet connection. It is real reality check. Read the full report in PDF format here.
New Report Shows U.S. Falling Behind Rest of World in High-Speed Internet Access, as Digital Divide Persists at Home
WASHINGTON — The United States continues to lag behind the rest of the world in accessible and affordable broadband service, with no signs of closing the digital divide at home, according to a new report released today by Free Press, the Consumer Federation of America and Consumers Union.
Contradicting the rosy picture painted by the Federal Communications Commission and Congress, Broadband Reality Check II exposes the truth behind America’s digital decline: A failed broadband policy that has left Americans with higher prices, slower speeds and no meaningful competition for high-speed Internet service.
“President Bush set a goal of bringing universal, affordable high-speed Internet access to every household by 2007,” said S. Derek Turner, research director of Free Press and author of the report. “We’re nowhere close to reaching that goal. Yet the FCC seems content to ignore the problem, manipulate the data, and pretend we’re moving forward.”
Broadband Reality Check II updates last year’s report on the state of high-speed Internet in America and details new empirical research. Among its key findings:
* The United States is 16th in the world in broadband penetration, and 14 other OECD nations saw higher overall net growth in broadband adoption than the United States from 2001 to 2005.
* Consumers in other countries enjoy broadband connections that are far faster and cheaper than what is available here. U.S. consumers pay nearly twice as much as the Japanese for connections that are 20 times as slow.
* The most important factors explaining the digital divide among nations are household income and poverty — not population density.
* U.S. broadband prices aren’t dropping: Cable modem prices are holding constant or rising, and DSL customers on average are getting less bandwidth per dollar than just a year ago.
* Despite claims of “fierce competition,” cable and DSL account for 98 percent of the residential broadband market. And over 40 percent of U.S. ZIP codes have one or fewer DSL or cable modem providers reporting service.
* The market share of “third platform” alternatives like satellite, wireless and broadband over powerline technologies has actually decreased over the past five years.
* Those living in urban areas are nearly twice as likely to have home broadband access as their rural counterparts.
* Approximately one out of 10 households with incomes below $30,000 reported having high-speed Internet access, but six out of every 10 households with incomes above $100,000 had broadband.
* The price of broadband service, and not necessarily the lack of a home computer, is the key barrier to broadband adoption by low-income households.
* The FCC uses misleading and meaningless measures of broadband coverage and competition, inflating estimates of broadband availability and competition.
FCC Chairman Kevin Martin — who will testify at a confirmation hearing before the Senate today — has used the misleading FCC data to claim that consumers have numerous choices among broadband providers. Broadband Reality Check II corrects the record.
Posted by ravenII at 9/13/2006 09:28:00 AM
If you have let Skype hijack your network already (Aka if you are a Skype user), you have noticed that Skype hijacking your browser, to get quality control feedback after a call. You may be n the middle of a fantastic site, like this one! And viola you will see the Skype page instead of this nice article, if you happened to have ended a call.
You can do two things,
uninstall skype using instructions here,
Modify your browser behavior to keep both.
I hope you are sensible and have Firefox installed, if not now visit Geemodo site and click on the install firefox button. It is a great browser and you will get much more benefits that just fixing Skype annoyances.
Once you have Firefox installed and made your default browser,
Click on TOOLS menu on the Firefox,
Click on Options,
Click on TABS, tab, and you will see;
Open links from other applications in:
- A new window
- A new tab in the most recent window
- The most recent tab/window
Select the second one.
Now Skype will open in a new tab.
Your problem solved.
May be Skype should fix this in their application. Get the feedback from user in the Skype application! Duh!
Posted by ravenII at 9/13/2006 08:36:00 AM
Monday, September 11, 2006
If you want to get more than 5 people on a conference call have a good quality connection and voice quality, Vapps (voip Apps) has come out with multiple connection service for skype. The service is free to skype users. But Vapps also offers paid services that makes your conferencing a dream.
Highspeedconferencing by VAPPS allows for up to 500 participants in a Skype conference call. You can be on Skype or on the phone. All the features of a traditional conference call at a fraction of the cost.
Highspeedconferencing is business class teleconferencing for Skype users. Callers can either be on Skype or on the regular phone. All of the features of a traditional teleconference are offered including mute/unmute, lock/unlock, etc. If you register for the HIGH SPEED PLUS+ service, you receive a personal conferencing number that puts you and all Skype callers directly into the conference call, you have access to web-based controls during the conference call and you can record and download the conference to listen to later.
* Cost: FREE for Skype Users
* Languages: English
* Provided by: Vapps
* More info: Highspeedconferencing.com
Yahoo messenger conference,
The ConFreeCall plug-in from Vapps will be available for download through Yahoo Messenger with Voice service. It will allow people to hold free conference calls to complement Yahoo's existing text, voice and photo-sharing services. A Vapps representative said the plug-in will be available within the next two weeks.
The free Gizmo Project software for Apple Macintosh, Microsoft Windows
and Linux computers is developed by SIPphone. This VoIP "softphone" enables
high- quality free calling worldwide using any Internet-connected computer.
Gizmo includes free conference calling, customizable voice-mail, Instant
Messaging (IM) and a host of other convenient features. The conference calling is also provided by vapps.
Posted by ravenII at 9/11/2006 11:41:00 AM
Sunday, September 10, 2006
If you happened to be on the OPENSER site browse over to the history of openser, you will find that there is no real comparison.
I was a ardent fan of SER and have implemented a few servers using various versions of SER. But I was always wondering about the way the development was traveling, from company to company and IPTEL.ORG site was off line for days. As the OPERNSER article states, Basically I could not count on the project. I was worried about the state of servers I have deployed.
Then I found OPENSER project, installed a test server, verified connectivity that I was using. All were better than before and at that point I converted all my servers one by one to OPENSER. OPENSER behaves very well among the all the itty bitty servers that I have running, Asterisk (versions 1, 1.2 and the latest 1.4), FreeSwitch, and trixbox.
But I do visit the ser site now and then. It seems to be up for most times and there are some activities. I will watch and let you know if there are significant changes. For now choose Openser.
Just for your information, I will state the history of OPENSER here, straight from the site;
OpenSER project started on the 14th June, 2005. That looks as a pretty young project, but actually it is full of history.
Origin of OpenSER started back in 2001-2002, at FhG FOKUS research institute in Berlin, Germany. In autumn 2002, the SIP Express Router (SER) project developed to be used in different European IST projects (e.g., EVOLUTE) was released open source under GPL license and the source tree moved to BerliOS open source mediator site. The home web site of the project was http://www.iptel.org, hosted by FhG FOKUS. The core developers of SER SIP server were: Andrei Pelinescu-Onciul, Bogdan-Andrei Iancu , Daniel-Constantin Mierla , Jan Janak and Jiri Kuthan. Very soon, new contributors joined the project, among early contributors you can find Juha Heinanen , Maxim Sobolev, Elena-Ramona Modroiu, Adrian Georgescu .
The quality and flexibility of the project made it grow rapidly. It was used in other IST projects but it moved in business. Sites like FreeWorldDialup, SipPhone, SipGate, VoIPUser are well known reference points and early adopters of the project.
Unfortunately, at the end of 2004, the evolution of the public project took an undesired direction. FhG Fokus decided to start a spinoff, iptelorg.com Gmbh, that focuses on businesses with SER. Soon after, iptelorg.com Gmbh was sold to Tekelek, which had no intention to continue the development of the public project. The core development team split in two: three of them followed Iptelorg.com Gmbh (Andrei Pelinescu-Onciul, Jan Janak and Jiri Kuthan) and the other two continued with the research institute (Bogdan-Andrei Iancu , Daniel-Constantin Mierla ). After a while, Bogdan-Andrei Iancu and Daniel-Constantin Mierla left the institute and started the OpenSER project in June 2005.
The fork of the project was forced by the obstacles encountered in the collaboration with Iptelorg.com Gmbh. No contributions were accepted, releases were delayed, no interest in project's development. The team founding OpenSER project was completed with Elena-Ramona Modroiu - a main contributor of SER (xlog, avpops, diameter support, pdt, speeddial).
Other SER contributors joined the project: Juha Heinanen , Klaus Darilion , Adrian Georgescu , Cesc Santasusana , Dimitry Isakbaiev, Andreas Granig. After one year, the project counted over 60 people contributing with code, patches or documentation.
First OpenSER release happened on the 14th June 2005, versioned 0.9.4 - source code forked from SER branch 0.9.0. Since then, other releases were made: 0.9.5 patch update to 0.9.4; 1.0.0 - major release - first open source SIP server with TLS support on the 28th October 2005; 1.0.1 - patch updated to 1.0.0; and last major release at this moment, 1.1.0, on the 10th July 2006.
Links to sites and articles;
VOIP IP Telephony: Asterisk Beta 1.4 by the end of the week and an interview!
VOIP IP Telephony: Trunk trixboxes (two right now!)
VOIP IP Telephony: FreeSWITCH breaks new ground in VOIP, telephony world!
Posted by ravenII at 9/10/2006 08:08:00 PM
Friday, September 08, 2006
If you are used to your mobile phone, who isn't? And would you like to have the similar capability in for your VOIP account? Then check out BASTA. I have not tested the service but from various documents and blogs, like VOIP WEBLOG, suggest that this to be your Internet "mobile" phone and lets you make and receive phone calls from your web browser, all you need is an email address and BUSTA manages your messages and contacts making sure that they always "follow you". I checked out the rates and they are not the greatest rates, but if the service does what it says, VOIP, landline, rates are not that bad.
According to the site BASTA.COM, BASTA stands for;
Brilliant, Internet calls and messages including video mail are free and you can chat online too(1)!
U can make Calls and send SMS text messages to regular phones and they are at very low rates, and you can also link your house phone(2) if you want. And use your free call minutes to give some one a buzz on with Busta
So add a free Busta gadget into your Google, Microsoft Live or NetVibes home page or eco system so you can easily and always stay in touch.
To add your personal address books and messages including Busta video mail is easy and they all "Follow you" . You can access your Busta account from any computer with a browser. Uploading your contacts is easy and even if you can't make or take that call you can always reply to messages or just send an instant low cost text.
Also forget memory sticks, large software downloads or compromised use of your computers and broadband connection, get your Busta Internet mobile phone today and give someone a buzz!
Posted by ravenII at 9/08/2006 10:11:00 AM
Thursday, September 07, 2006
Geemodo reports on the new blackberry Pearl, or BlackBerry 8100 as it was earlier known. The phone, RIM device has a bunch of features that BlackBerry lovers will love. Read more at,
Geemodo: BlackBerry Pearl or Blackberry 8100, You can have it now!
Posted by ravenII at 9/07/2006 10:18:00 AM
Sineapps has posted a and interview with Kevin Fleming, a senior software engineer at Digium.
But what stood out in the article was;
Question 2: So, a couple of things point to the fact that the beta of Asterisk 1.4 might be coming out this week, can you confirm this?
Yes. We will be producing the first beta of Asterisk 1.4 by the end of this week, and then actively working to resolve all known issues as quickly as possible.
I think the interview itself a good one as always. Also those who is wondering what ASTERISK 1.4 may bring, here is an answer from the post;
Question 6: What are some of the changes coming in the version 1.4 release (compared to 1.2)
Well, the list is quite long, but here are about twenty that we've already identified (and we haven't done an exhaustive review yet):
Generic jitter buffer
Variable Length DTMF (proper RFC-2833 support)
Asterisk Extension Language Version 2
Shared Line Appearance support
ODBC() dialplan function
T.38 FAX passthrough support
IAX2 scalability improvements
Re-architected build system with better maintainability and portability
New high-quality sounds in English, French and Spanish
IMAP storage support for voicemail
RADIUS support for CDR storage
HTTP Asterisk Manager Interface (with AJAX components)
SIP transfer interoperability improvements
Cisco SCCP (Skinny) channel driver improvements
Better language support for speaking dates, times and numbers
IAX2 media-only transfers
Memory usage and thread locking reduction
Addition of a simple 'users.conf' configuration file for SOHO/SMB users
RTP native bridging
Posted by ravenII at 9/07/2006 09:53:00 AM
Monday, September 04, 2006
FreeSWITCH is an open source telephony application built from the ground up
and designed to take advantage of most existing voip, telephony software and libraries. FreeSWITCH paves the way for one to build an open source PBX system, an open source voip switching platform or a VOIP, Telephony gateway uniting various technologies and platforms such as SIP H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle and OpenPBX, Bayonne, YATE or Asterisk.
FreeSwitch is a multi-platforms software and it runs on Windows Linux and MAC. It is also possible to run it on other UNIX flavors.
FreeSwitch is built on existing OpenSource libraries and projects, namely SQlite, SRTP Secure RTP, Apache Portable Runtime, Lib Resample, exosip , IAXclient, Speex Codec, libsndfile.
FreeSWITCH, the telephone soft-switch, has touched upon a few milestones combining many a famous VOIP application features into it's core.
In Early April FreeSWITCH announced it's interoperability capabilities with the GoogleTalk, goggle's Voice chat program. Making it possible to gateway calls to SIP or the PSTN from Googletalk. Then again in July it brought out the capability of voice switching at 16Khz against the traditional 8Khx switching done in the VOIP world. This is a significant improvement on the quality of voice. What does that brings to a phone conversation? It brings more richness and clarity to voices, improving the overall experience of a phone call.
Now FreeSWITCH has done it again, Now it has brought the first two elements together and topped it off with a new capability that may change the way we interface to our phones.GoogleTalk has recently released a new version of their client capable of transmitting audio at 16 kilohertz making it possible to call FreeSWITCH and interact in a conference bridge or listen to a text-to-speech engine read you your favorite news story all in high definition audio.
You want a twist with that? Yes you can have all that and more, interact with the system on the phone by listening to the audio and dialing a few digits, now you can send and receive text messages with the system at the same time.
Imagine, you start your VOIP/TELEPHONY/CHAT program, and a voice asking you for your account information, then in a chat window you type your name and another person on the other end, on a phone be able to intercept the information and react accordingly. This may break the paradigm of the auto-attendant altogether. And I am sure the idea will run wild through the VOIP community. Who knows, one slime might even try to patent the idea!
Posted by ravenII at 9/04/2006 04:33:00 AM
I was reading an article on Techcrunch, Hullo To Enter VOIP War With Free Product, a well written and in depth article about current lower end voip players in the field. Mainly concentrating on the services and abilities of the HULLO.
I actually read the whole article rather than skimming and ventured out to HULLO site. It is not an easy site to navigate. The front page will make you want to get the product and register.
But I managed to get to the area in the site that had some text. As Techcrunch mentioned services offered are great, and bunch of new services are on the way. I also saw somewhere on the site saying that the service is free! Well I was happy as this was something I was looking for and I really did not want to pay for something that dials phone for me and switches to another when I want to. I am looking to make my phone life easier, not to work harder to pay for the services that make my life easier!!
Anyway I was going through the FAQ, which is also a bit obscure but if you know your way around the web, you will find a tiny little button to click that reveal some text.
One such piece of text made me leave the site, and it read;
How much does it cost to use hullo?
During the beta the service is free. Later, we expect to offer a mix of free and paid services.. So it is not really free and not what I was looking for. I think it will be helpful for HULLO to be a tad more open and truthful. They certainly aren't lying but you have to dig for the truth.
Had they stated the facts upfront, I might have at least tested the product, as it seems to be a fine product.
Posted by ravenII at 9/04/2006 03:33:00 AM
Saturday, September 02, 2006
Since many people have shown interest in the article, VOIP IP Telephony: How to kill a Skype / remove Skype installs from your network!, I thought of revisiting the article to add Linux removal information. Now that the Various Linux Distributions are finding their ways to desktops there might be some one seeking to remove skype. Here we go!
Uninstalling Skype from Linux
You can uninstall Skype the same way that you uninstall any Linux software program, depending on how the application was installed:
* If you used tar to install Skype, you can remove the Skype application and its associated directories with the rm command.
* If you installed Skype with rpm, you may want to refer to the documentation for your particular Linux distribution for specific instructions.
Uninstalling Skype with tar
If you installed Skype with tar, uninstalling it requires great caution because you need to use the rm command, possibly as the superuser (root).
Because Skype has a daemon that runs as a background process, it is best to stop it before attempting to uninstall Skype.
To stop the process, first you must identify the process identification number. To do this, follow these steps:
1. Log in as the user who is running Skype, and type ps -aux | grep skype
2. When you have found the Linux process identification for the Skype application, kill the process by issuing the following command: kill -9 skype_process_ID where skype_process_ID is the process identification number associated with Skype.
3. Change directories to the directory where tar was originally executed to install Skype. In other words, cd to the directory directly above the Skype directory.
4. When you are absolutely certain that you are in the correct directory, enter the following command to remove the Skype subdirectory and associated files forcefully: rm -rf skype-directory where skype-directory is the specific name of the directory in which Skype is installed.
Uninstalling Skype with rpm
If you installed Skype with rpm, uninstalling it is simple. You may want to check the man page for rpm on your Linux distribution to make sure that there is nothing special you need to know.
Because Skype has a daemon that runs as a background process, it is best to stop it before attempting to uninstall Skype. To stop the process, first you must identify the process identification number. To do this, follow these steps:
1. Log in as the user who is running Skype, and type ps -aux | grep skype
2. When you have found the Linux process identification number for the Skype application, kill the process by issuing the following command: kill -9 skype_process_ID where skype_process_ID is the process identification number associated with Skype.
3. Enter the following command to uninstall the Skype application software: rpm -e skype_package.rpm where skype_package is the specific name of the rpm package that installed SkypeÂfor example, skype-0_90_0_1.rpm.
Posted by ravenII at 9/02/2006 01:00:00 PM
After posting VOIP IP Telephony: Philips gives you a computer less voip phone, for Skype., I noticed that US Robotics has also released a Skype Phone.
Don't run and buy it yet! Because it is one step behind the Philips Phone, This US Robotics phone needs computer to operate. It is not much different from a wireless head phone. But it is yet useful in the sense that if you are on the skype phone and need to walk away from the computer. Now you can do it with Skype compatible USR9630 USR9631 cordless phone.
News release is here.
Posted by ravenII at 9/02/2006 10:45:00 AM
Expect a new advertisement in your TV, DirectTV has entered a Pilot program to test delivery of VOIP, IP Telephony via it's satellite service. With the competition from Phone and Cable companies offering bundles that consists of TV, PHONE and Broadband, DirecTV needs to find new avenues to retain it's customers.
The joint venture, called DirecPath, will use voice over Internet Protocol equipment from Vistula Communications Services to run a six-month pilot program of the technology. If the trial goes well, voice services could be expanded to a larger portion of DirecPath's customers, Vistula said in a statement.
DirecPath was formed by DirecTV and Hicks in May to provide bundled services such as broadband Internet access, phone service and video to multiple dwelling units (MDUs) and gated-community residents across the United States. DirecTV has formed partnerships with phone companies Verizon Communications, Qwest Communications International and BellSouth to sell a bundle of broadband, telephony and TV services to single-family homes.
Posted by ravenII at 9/02/2006 10:12:00 AM
Friday, September 01, 2006
The Agricultural Bank of China (ABC), China's largest commercial bank, with revenues exceeding $250 billion, was looking to decrease its long-distance phone expenses between its more than 50,000 domestic and international branches and affiliate offices.
FanFan Zheng, division manager of information technology of ABC, says, "We evaluated a lot of VoIP products. We like Tenor because of its ease of installation and because it doesn't require any changes to the existing voice or data networks.
These locations were in all of China's major provinces, cities and counties. ABC's goal was to consolidate multiple regional call centers to one centralized call center through a Voice over IP (VoIP) network.
Of foremost concern to ABC among its many requirements to implementing VoIP was that it had to have minimum impact on the existing IP network and voice system performance. The bank did not want to have to upgrade the existing IP network, reprogram its PBX or purchase any PBX upgrades or interface cards.
Most importantly, it wanted assurances that there would be no degradation to voice quality compared to wire line calls.
The bank required that its voice network be compatible with international standards (i.e., H.323), and also needed to be scalable for nationwide deployment and growth, and interpretable with other vendors. ABC wanted to minimize the impact on its end-users so they did not have to change their dial plans or habits. Overall, ABC required high reliability, even under extreme and unusual circumstances.
Public Information Technology Co. (PiTech), a large Chinese distributor, recommended Quintum's Tenor VoIP multiPath switches. The switches had the technologies that ABC required, the diverse applications that ABC's environment necessitated and the customer support that put ABC at ease. Read more here..
Posted by ravenII at 9/01/2006 12:44:00 PM
Introduced at the IFA 2006 World of Consumer Electronics show in Berlin, Thursday, the Philips VOIP841 (Press release is here)is scheduled to be available by the end of the year.
"We are broadening our reach to mass consumers by offering them the opportunity to communicate via Skype without having to be tied to the computer," said Stefan Oberg, Skype's general manager for desktop and hardware.
The firm said the phones will have Skype software installed enabling them to be used to make and receive traditional phone calls through landline connections. The firm uses proprietary software and eschews standard SIP. So earlier mentioned Skype network problems may still comes with the phones as well, but you might not be able to remove skype this time!
Premium Skype features, including SkypeOut, SkypeIn, and Skype Voicemail will all be accessible through the phone.
The cordless phones for Skype are accompanied by a remote DECT base station that plugs into both the broadband connection and the traditional phone line. The system handles both Skype and ordinary calls in one phone, presented through an easy-to-use, user interface. Both the Philips and NETGEAR cordless phones boast a full color graphic display, simplified calling features with an integrated contact list, and an enhanced speakerphone with great voice clarity.
Prices for the Philips and Netgear phones were not available Thursday. However, phones not adhering to the SIP standard are likely to be more expensive than SIP-certified phones. Inexpensive SIP phones like Cisco Systems' Linksys models have been on the market for months. SIP phones are cheaper because they utilize standard SIP software unlike closed Skype software.
Posted by ravenII at 9/01/2006 10:43:00 AM