Why VoIP Needs Crypto
Click Here: Why VoIP Needs Crypto?
From PBX to VoIP: Making the Change
Voice over Internet Protocol (VoIP) telephony Relevant Products/Services from , once a tool used primarily by uber techies, has matured into a viable and less-expensive alternative to the PBX systems used by businesses of all sizes. With VoIP, companies have the opportunity to discard the prepackaged offerings of traditional telecommunications and instead opt for a phone system that is customizable and highly adaptable.
According to a report released in January from the research firm Yankee Group, the VoIP market is expected to reach $3.3 billion in service revenue by 2010. The report said that businesses are favoring VoIP because the technology offers measurable savings, an excellent converged platform for voice and data, and improved ways to manage communications within the enterprise.
"Virtually all major carriers, systems integrators, and equipment vendors now offer different varieties of business VoIP services," said Taher Bouzayen, a senior analyst for telecommunication strategies at the Yankee Group. "And those that are not already exploring ways to capitalize on this revenue opportunity need to start now."
Even the U.S. military is contemplating a move to VoIP. Avaya, a VoIP vendor, recently announced that the Air Force is testing the technology with an eye toward deploying it for military communications in the field. According to Avaya client executive Vic Galante, the Air Force realized that it would have to turn to commercial technologies to save money.
Click Here for the Full Article
According to a report released in January from the research firm Yankee Group, the VoIP market is expected to reach $3.3 billion in service revenue by 2010. The report said that businesses are favoring VoIP because the technology offers measurable savings, an excellent converged platform for voice and data, and improved ways to manage communications within the enterprise.
"Virtually all major carriers, systems integrators, and equipment vendors now offer different varieties of business VoIP services," said Taher Bouzayen, a senior analyst for telecommunication strategies at the Yankee Group. "And those that are not already exploring ways to capitalize on this revenue opportunity need to start now."
Even the U.S. military is contemplating a move to VoIP. Avaya, a VoIP vendor, recently announced that the Air Force is testing the technology with an eye toward deploying it for military communications in the field. According to Avaya client executive Vic Galante, the Air Force realized that it would have to turn to commercial technologies to save money.
Click Here for the Full Article
GoogleTalk Gets a Facelift for Business with Asterisk
Talk about mind-boggling changes. A new project will allow businesses to connect GoogleTalk users to their Asterisk, telephony servers, Mark Spencer told me yesterday. He should know. Spencer is the author of the Asterisk open source, IP PBX and CEO of Digium, the company packaging Asterisk as a business-grade solution
The project is currently in beta with availability set for some time around June, smack in between the May release of Digium's Business edition of Asterisk B.1 and the next rev of the public Asterisk 1.4 release in July.
I'm so incredibly excited about this project for lots and lots of reasons. Inter-company collaboration will become a lot easier now for Asterisk users. Extending out the Asterisk network is a cinch and think of all of the cool applications one could create for GoogleTalk. Heck, just real termination becomes easier.
But I get ahead of myself.
One of the things that's always puzzled me is why Google didn't just support SIP or even Mark's own IAX protocol used in Asterisk. Mark had the same question until he studied Google's SIP replacement, Jingle
Click Here for the Full Article
The project is currently in beta with availability set for some time around June, smack in between the May release of Digium's Business edition of Asterisk B.1 and the next rev of the public Asterisk 1.4 release in July.
I'm so incredibly excited about this project for lots and lots of reasons. Inter-company collaboration will become a lot easier now for Asterisk users. Extending out the Asterisk network is a cinch and think of all of the cool applications one could create for GoogleTalk. Heck, just real termination becomes easier.
But I get ahead of myself.
One of the things that's always puzzled me is why Google didn't just support SIP or even Mark's own IAX protocol used in Asterisk. Mark had the same question until he studied Google's SIP replacement, Jingle
Click Here for the Full Article
Uplink Connects SIP & Skype
Note: After helping break the story about SIP to Skype progress it is nice to see products coming out in a more refined form.
SIP protocol is regarded as the industry standard for VoIP. A number of telephone companies, IP phone manufacturers and virtual IP based PBX systems have been using this protocol for connecting calls. On the other hand, Skype has been using a proprietary protocol and whenever one wants to call from a SIP based VoIP service to Skype, one requires an adapter which acts as a gateway so that calls be made as SIP and Skype are not interoperable.
NCH Swift Sound has come with Uplink which connects SIP based voice calls to Skype and it works in both directions. It can be easily configured and used. Once the software has been installed, go to the options to give the path to Skype executable, enter SIP service setting and dial options when Skype calls SIP and vice versa. If configured in the right manner, it works perfectly and provides a good voice quality.
Click Here for more Informaion
SIP protocol is regarded as the industry standard for VoIP. A number of telephone companies, IP phone manufacturers and virtual IP based PBX systems have been using this protocol for connecting calls. On the other hand, Skype has been using a proprietary protocol and whenever one wants to call from a SIP based VoIP service to Skype, one requires an adapter which acts as a gateway so that calls be made as SIP and Skype are not interoperable.
NCH Swift Sound has come with Uplink which connects SIP based voice calls to Skype and it works in both directions. It can be easily configured and used. Once the software has been installed, go to the options to give the path to Skype executable, enter SIP service setting and dial options when Skype calls SIP and vice versa. If configured in the right manner, it works perfectly and provides a good voice quality.
Click Here for more Informaion
iotum helps makes Asterisk relevant to more Enterprise Users
For the more than 250,000 Asterisk IP-PBX installations around-the-world, getting the right phone call, at the right time, and on the right device just got easier thanks to iotum. Today, at VON Canada in Toronto, iotum announced the beta availability of the iotum Asterisk integration module; two relevance-enabled call management applications for Asterisk and a rich new set of developer API's.
iotum's Asterisk integration module connects its award-winning Web 2.0 call management applications to Asterisk IP-PBX's to help users prioritize which calls are important, and which can wait, based on who's calling, and what the user is currently doing. The iotum Asterisk integration module consists of source code, installation instructions and access to an iotum test account.
"Today's announcement represents the next step in iotum's platform strategy", said CEO Alec Saunders. "As of today, members of the world-wide community of Asterisk developers and integrators can use our platform to build new and compelling relevance-enabled revenue generating applications."
Once the integration is complete, this non-commercial beta will allow Asterisk users to filter, rank, and prioritize incoming calls using iotum as well as offer users the ability to easily schedule conference calls from within Microsoft Outlook using iotum's Pronto Conference Calling feature. With, Pronto Conference Calling participants join a conference call by dialing the organizer's usual phone number eliminating the need for everyone to dial into a conference bridge or remember pass codes.
The iotum Relevance Engine API allows Asterisk developers to embed advanced, contextually driven call management capabilities into their own applications. The API allows services to be built which use the iotum Relevance Engine, the world's first smart platform to intelligently assess the relevance of a phone call, and route it to the most appropriate device on any network.. The API is for use in a variety of different kinds of applications, including collaboration, CRM, and personal productivity applications.
For more information about iotum, visit the company's web site:
http://www.iotum.com
iotum's Asterisk integration module connects its award-winning Web 2.0 call management applications to Asterisk IP-PBX's to help users prioritize which calls are important, and which can wait, based on who's calling, and what the user is currently doing. The iotum Asterisk integration module consists of source code, installation instructions and access to an iotum test account.
"Today's announcement represents the next step in iotum's platform strategy", said CEO Alec Saunders. "As of today, members of the world-wide community of Asterisk developers and integrators can use our platform to build new and compelling relevance-enabled revenue generating applications."
Once the integration is complete, this non-commercial beta will allow Asterisk users to filter, rank, and prioritize incoming calls using iotum as well as offer users the ability to easily schedule conference calls from within Microsoft Outlook using iotum's Pronto Conference Calling feature. With, Pronto Conference Calling participants join a conference call by dialing the organizer's usual phone number eliminating the need for everyone to dial into a conference bridge or remember pass codes.
The iotum Relevance Engine API allows Asterisk developers to embed advanced, contextually driven call management capabilities into their own applications. The API allows services to be built which use the iotum Relevance Engine, the world's first smart platform to intelligently assess the relevance of a phone call, and route it to the most appropriate device on any network.. The API is for use in a variety of different kinds of applications, including collaboration, CRM, and personal productivity applications.
For more information about iotum, visit the company's web site:
http://www.iotum.com
A new twist on VoIP? - A new Jajah killer and serious Skype competition?
Since early 2006 movement has come into the VoIP industry. New VoIP providers are now launching all over the world with each one of them hoping and expecting a share of the ever growing popularity and income stream. At the last count the research company www.myvoipprovider.com had almost 650 VoIP phone providers listed.
This has had one distinct advantage for the consumer - VoIP costs internationally are dropping at an alarming rate. A few VoIP providers in Europe have taken their marketing activities to the extreme by offering free calls to a wide range of up to 50 international destinations. VoipBuster was the pioneer early 2005 and has since then launched a barrage of sister companies offering exactly the same type of service. Time will tell if this "Free VoIP" campaign has any long term merit.
Even in this highly competitive enviroment some companies still manage to stand out of the masses. In mid March 2006 two companies launched, in one case, relaunced their services. Lycos decided that it is time to join the race with the likes of Yahoo and possibly in the very near future Google and MSN. Using Globe7's technology Lycos launced an interesting softphone with a free US phone number, 100 free minutes and an integrated mp3 player and video.
Click Here for the Full Article
This has had one distinct advantage for the consumer - VoIP costs internationally are dropping at an alarming rate. A few VoIP providers in Europe have taken their marketing activities to the extreme by offering free calls to a wide range of up to 50 international destinations. VoipBuster was the pioneer early 2005 and has since then launched a barrage of sister companies offering exactly the same type of service. Time will tell if this "Free VoIP" campaign has any long term merit.
Even in this highly competitive enviroment some companies still manage to stand out of the masses. In mid March 2006 two companies launched, in one case, relaunced their services. Lycos decided that it is time to join the race with the likes of Yahoo and possibly in the very near future Google and MSN. Using Globe7's technology Lycos launced an interesting softphone with a free US phone number, 100 free minutes and an integrated mp3 player and video.
Click Here for the Full Article
AstManProxy 1.20pre Released
Hi Folks,
I have done a bunch of new work on AstManProxy, including 1) adding in the Action: Challenge authentication mechanism (basically done), 2) adding in support for SSL, 3) added patch from Steve Davies that will do basic user authentication.
The SSL support is based on code from John Todd at Tello (digium #6812). I do not have the SSL stuff functional yet but it is compiling and I am hooked into their underlying routines. I just need to figure out how to hook all the functions in now, which I'll be working on over the next few days.
Also, I have setup a proper branches/tags/trunk structure on svncommunity.digium.com (thanks to Kevin Fleming for his help). 1.13 is currently in tags & trunk, and branches contains the 1.20 development work.
Anyway, please take a look at the 1.20pre branch and let me know what you think:
Click Here for Download of 1.20pre
Also I have turned on moderation on the astmanproxy list, so we should not have any more spammers... sorry about that.
Best,
Dave
I have done a bunch of new work on AstManProxy, including 1) adding in the Action: Challenge authentication mechanism (basically done), 2) adding in support for SSL, 3) added patch from Steve Davies that will do basic user authentication.
The SSL support is based on code from John Todd at Tello (digium #6812). I do not have the SSL stuff functional yet but it is compiling and I am hooked into their underlying routines. I just need to figure out how to hook all the functions in now, which I'll be working on over the next few days.
Also, I have setup a proper branches/tags/trunk structure on svncommunity.digium.com (thanks to Kevin Fleming for his help). 1.13 is currently in tags & trunk, and branches contains the 1.20 development work.
Anyway, please take a look at the 1.20pre branch and let me know what you think:
Click Here for Download of 1.20pre
Also I have turned on moderation on the astmanproxy list, so we should not have any more spammers... sorry about that.
Best,
Dave
VoIP Caller ID Spoofing - Still Dangerous
Many in the VoIP service industry have known for years that caller ID can be spoofed (that is, misrepresented) relatively easily. In fact, one need not be an expert at using Asterix's Linux PBX software or know the other tricks of the trade - he can simply pay a few dollars for an Internet telephone caller ID spoofing service. (We're not going to provide free advertising for these services here) While this may seem harmless, it opens up the door to a number of serious vulnerabilities.
More and more caller ID is being used to authenticate people's identity. Credit card companies have long been using caller ID in the card activation process. Financial institutions such as Citibank and American Express are now using it to authenticate identity of account holders who dial in to their telephone service. In business, caller ID is used to signal whether a caller is calling from inside or outside the firm. 911 call centers use it to determine who is calling and where to send emergency responders. Voicemail systems, particularly cell phone voicemail systems, automatically playback messages based on caller ID.
Click Here for the Full Article
More and more caller ID is being used to authenticate people's identity. Credit card companies have long been using caller ID in the card activation process. Financial institutions such as Citibank and American Express are now using it to authenticate identity of account holders who dial in to their telephone service. In business, caller ID is used to signal whether a caller is calling from inside or outside the firm. 911 call centers use it to determine who is calling and where to send emergency responders. Voicemail systems, particularly cell phone voicemail systems, automatically playback messages based on caller ID.
Click Here for the Full Article
Connect GoogleTalk to your Telephone with FreeSwitch
The FreeSwitch project announces the immediate availability of a brand new Open Source Jingle XMPP signaling library as well as an endpoint module enabling a Jingle telephony gateway. The library dubbed "libDingaLing", written in C, creates a layer of abstraction to allow for an easier transition as the Jingle protocol evolves and eliminates the need to deal with XMPP or XML and supports many concurrent instances within 1 application.
The library is currently considered to be in Alpha stage, has been compiled and tested on many computer platforms including Windows XP, Solaris, Linux and MacOS X. The only other existing implementation of this protocol released thus far is the GoogleTalk instant messenger application therefore the library has been designed with interoperability with this particular client in mind but also anticipates changes in the protocol to come along as it becomes more widely accepted.
The new endpoint module appropriately named "mod_dingaling" couples FreeSWITCH to libDingaLing and allows both inbound and outbound communication. With this technology, GoogleTalk calls can gateway to the PSTN or to other VoIP protocols such as SIP or H323.
FreeSWITCH, http://www.freeswitch.org is a new open source telephony project started in early 2006 designed to provide a modular platform on which to merge various technologies. Both libDingaLing and FreeSWITCH were written by Anthony Minessale II, a developer who after contributing to other telephony related open source projects, decided to start a new initiative that focuses on abstraction, modularity and cross-platform crossarchitectural design.
The library is currently considered to be in Alpha stage, has been compiled and tested on many computer platforms including Windows XP, Solaris, Linux and MacOS X. The only other existing implementation of this protocol released thus far is the GoogleTalk instant messenger application therefore the library has been designed with interoperability with this particular client in mind but also anticipates changes in the protocol to come along as it becomes more widely accepted.
The new endpoint module appropriately named "mod_dingaling" couples FreeSWITCH to libDingaLing and allows both inbound and outbound communication. With this technology, GoogleTalk calls can gateway to the PSTN or to other VoIP protocols such as SIP or H323.
FreeSWITCH, http://www.freeswitch.org is a new open source telephony project started in early 2006 designed to provide a modular platform on which to merge various technologies. Both libDingaLing and FreeSWITCH were written by Anthony Minessale II, a developer who after contributing to other telephony related open source projects, decided to start a new initiative that focuses on abstraction, modularity and cross-platform crossarchitectural design.
Arisng Group Evaluates Asterisk for International Clients
Arising Group, Inc. has begun rigorous testing of the open source VoIP PBX, Asterisk. It is hoped that the new communication platform will provide a robust and low cost telephone solution for their international clients who need to stay connected with their traveling field representatives.
George Karshner, Director of Business Development at Arising Group said that what attracted them to Asterisk is the versatility and functionality of the program. "It appears that Asterisk can perform all of the functions of a premium office phone system like voicemail, conference bridging, call queuing, and call detail records, plus higher functions usually required by trading firms, like talk detection, call monitoring, and remote call pickup" Karshner said. "Its flexible feature set is very promising", he added, “but we want to run a battery of situation tests before we deploy it".
Arising Group has several multinational clients who need to stay in touch with their workers in Europe and Asia. One China based client has reps in the U.S., Japan, Korea, Vietnam, Hong Kong, Taiwan, Singapore, Malaysia and Indonesia. All of these people must communicate frequently from places where Internet service is more reliable than cell phone connections. They are also looking to cut down their international calling costs. With this new VOIP technology, all the inter-office phone bills will be one low, flat rate, Internet connection call.
Click Here for the Full Release
George Karshner, Director of Business Development at Arising Group said that what attracted them to Asterisk is the versatility and functionality of the program. "It appears that Asterisk can perform all of the functions of a premium office phone system like voicemail, conference bridging, call queuing, and call detail records, plus higher functions usually required by trading firms, like talk detection, call monitoring, and remote call pickup" Karshner said. "Its flexible feature set is very promising", he added, “but we want to run a battery of situation tests before we deploy it".
Arising Group has several multinational clients who need to stay in touch with their workers in Europe and Asia. One China based client has reps in the U.S., Japan, Korea, Vietnam, Hong Kong, Taiwan, Singapore, Malaysia and Indonesia. All of these people must communicate frequently from places where Internet service is more reliable than cell phone connections. They are also looking to cut down their international calling costs. With this new VOIP technology, all the inter-office phone bills will be one low, flat rate, Internet connection call.
Click Here for the Full Release