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Tag: asterisk

  • VoicePulse Announces SIP Trunking Interoperability with IPitomy PBX Products

    VoicePulse and IPitomy announced that they have successfully completed interoperability testing between SIP products and services. VoicePulse is now interoperable with IPitomy’s Pure IP PBX platform.

    IPitomy designs and manufactures IP telephony equipment for businesses. VoicePulse delivers SIP trunking, worldwide termination, origination, and phone service to residential and business customers.

    According to the companies, IPitomy’s PBX combined with VoicePulse’s SIP origination and termination services “create a complete phone solution for businesses of all sizes.”

    VoicePulse can now provide new customers using IPitomy’s IP PBX with official configuration guides when setting up VoicePulse VoIP services on their PBX. Businesses using IP PBXs such as Digium’s Asterisk, AsteriskNow, Fonality’s PBXtra, trixbox, FreePBX, FreeSwitch, Switchvox.

    “And now IPitomy can benefit from VoicePulse’s competitive international rates, toll-free services, and failover features all on a “Tier 1” back bone network,” as the company claims.

    “We are confident that with this new relationship, we will see an increase of resellers and new customers combining IPitomy’s IP PBX products with VoicePulse VoIP services,” said Monica Haley, Marketing Associate at VoicePulse.

    “IPitomy sets itself apart from its competitors by providing enhanced features beyond the typical key systems and PBXs of the past,” she added.

    Nick Branica, President and CEO of IPitomy, said: “IPitomy is happy to add VoicePulse to our growing list of qualified SIP providers. SIP providers are a very important component to the growth of the IP PBX market and VoicePulse adds value to our total overall product offering. Open Standards SIP based systems are all about choice.”

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  • Digium and Aumtech Certify a Multi-Language Speech Recognition Server Solution with Asterisk

    Digium and Aumtech, Speech and Computer Telephony Integration company, announced a partner relationship between the companies.

    According to them, Asterisk users now have a high-quality, low-cost Speech alternative, featuring server-based licenses that support 48+ ports of Automated Speech Recognition for less than the cost of one or two competitive ASR licenses.

    Aumtech’s solution provides 48+ ports of multi-language speech recognition for the Asterisk platform with support for $1,975 per port.

    The firm is introducing its Media Resource Control Protocol (MRCP) Connector utility, which enables Asterisk customers to access the high-quality Microsoft Speech Services 2007 ASR for a fraction of the cost of competitors’ ASR engines, as the company claims.

    This includes unlimited grammars and multiple languages of ASR. Aumtech says the list of 14 available languages and voices is constantly growing, and there are successful installations in North America, South America, Europe and the Far East.

    Asterisk users can access this speech solution on Asterisk and on a variety of Interactive Voice Response solutions, including Aumtech’s VoiceXML 2.1 Certified IVR platform, or other standards-based IVR platforms.

    The Aumtech MRCP Connector for the Microsoft Office Communications Server 2007 Speech Server allows companies using Linux to build speech-driven applications with the Microsoft ASR product.

    Since the MRCP Connector is based on the WC3 Forum’s open standard, applications running on open platforms such as Asterisk can access Microsoft’s Speech Recognition functions through Aumtech’s Connector. Existing Speech applications are also candidates for deployment on this solution, since industry standards ensure code portability.

    Bill Miller, Digium’s vice president of product management, said, “Our Asterisk customers have long recognized the value of Speech Recognition applications; however, most have delayed implementing these technologies due to the high entry costs. Aumtech’s solution is the lowest-cost at higher capacities for world-class Speech technology and may prove to be the ‘tipping point,’ finally bringing speech to the masses, and will benefit our customers worldwide.”

    Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company’s software and hardware products enable businesses to implement turnkey unified communications solutions or to design their own VoIP systems.

    Aumtech is the Automatic Speech Recognition systems provider for large enterprises, governments, and global carriers. Its innovations include the first SIP-based IVR, Automated Outbound IVR Notification, and a standalone CTI product, called ESP (“Economical Screen Pop”).

  • Kolmisoft Releases Free Version of VoIP Billing and Routing Platform MOR

    Kolmisoft, a creator and developer of all in one solution – softwich with billing and routing functionality, has released a free community edition of its platform MOR focused on the startups and entrepreneurs who are willing to start a VoIP business, the company announced.

    Kolmisoft’s versatile application can be used for various VoIP business models (wholesale, retail, prepaid and postpaid), branded with provider’s logo and integrated with the provider’s back-end or toolbox.

    Running on Asterisk, MOR easily handles even 300-500 simultaneous calls on a single server, the company claims.

    The free version has the same features and functionality as the commercial edition, just limited to ten concurrent calls.

    “We had a free version in the past along with first release of MOR, then went to a trial version, but our customers did not like it because they knew the next call they would get was from a sales representative, or the trial period would expire before they could even finish testing,” Kolmisoft CEO Mindaugas Kezys said.

    “With the new release, Kolmisoft is hoping to help companies with low budget to start VoIP business and upgrade the software to commercial edition only when their business begins to grow.”

    Apart of 15 new futures, MOR 8 comes with two new modules: Mobile Number Portability add-on enables mobile telephone users to retain their mobile telephone numbers when changing from one mobile network operator to another and Recordings add-on allows to record selected users’ calls for monitoring purposes.

    According to Kezys, by using MOR 8, telecom companies can effectively provide VoIP services sparing more time for marketing their product instead of worrying about infrastructure. “Kolmisoft support team can easily take care of VoIP related problems by client’s wish,” he said.

    “This version of MOR is the most reliable and powerful in Kolmisoft history”, CEO of Kolmisoft stated.

    He added that the platform includes a “how to make a first call” guide and has default provider Kolmisoft so users could instantly test the system, see how call is billed and start using MOR system for their VoIP business.

  • Skype for Asterisk Now Available


    Digium, the Asterisk Company, and Skype announced the general availability of Skype for Asterisk.

    Skype for Asterisk is an add-on channel driver for Asterisk-based PBX systems. The software is compatible with the free and open source Asterisk versions 1.4, 1.6 and AsteriskNOW, as well as the commercially licensed Asterisk Business Edition.

    It enables multiple concurrent Skype calls from a single Skype account, and supports both G.711 and G.729a calling.

    According to Digium, with the new software, customers can:

    • Manage business Skype accounts with the Business Control Panel
    • Get low Skype global rates on outbound calls (as low as 2.1US¢ per minute)
    • Receive inbound calls to online numbers
    • Route calls according to Skype profile fields, online status and privacy settings
    • Streamline customer contact by allowing Web site visitors to place free Skype calls directly to their business with global click-to-call buttons

    The companies promote the software as a solution for connecting Asterisk-based business phone systems to Skype.

    “We created Skype accounts such as Digium Sales and Digium Support—a convention I suspect many companies will quickly adopt. Now, our customers all over the world can call us for free using Skype and our Asterisk PBX processes the inbound call just like it would a normal call,” said Danny Windham, CEO of Digium.

    “Skype for Asterisk allows businesses to access the 400 million community of people communicating over the Internet, natively encrypts all voice calls and lets companies manage their Skype user accounts via Skype’s Web-based Business Control Panel,” the company says.

    Businesses already using an Asterisk-based phone system can add Skype as another complementary form of communications by downloading Skype for Asterisk, without additional costly hardware.

    Skype for Asterisk is available to download for $66 per concurrent call. It comes with 90 days of installation support from the time of purchase.

  • Positive Signs For Interoperability Between VOIP Systems


    It would appear that efforts to address problems of compatability and interoperability between the various VoIP protocols, packages and services are making some headway.

    For users – and especially small businesses – the issue has been of growing concern as the popularity of VoIP has led to a huge increase in the number of VoIP services.

    As Erica Stewart points out in her NetSquared blog, these potential users may want to adopt VoIP for their communication needs – but compatibility issues prevent them from fully entrusting their operations to such a service.

    However, she goes on to say that developments have been made since Stefan Oberg, Skype’s Vice-President for Business, announced the launch of a development program with Digium (the developers of Asterisk) last autumn.

    The partnership aims to incorporate Skype with the Asterisk system in a way that will allow the Asterisk PBX handle Skype calls more clearly and efficiently.

    Stewart says clients using this service now have the capability to make and receive Skype calls from within their existing Asterisk software and hardware systems.

    She says the partnership between the two well-known VoIP providers points to the growing interoperability between the various VoIP systems.

    "A small business owner can become more at ease in knowing that despite the multitude of both open source and proprietary voice codecs available in the market, there are efforts to be able to connect them to one another," she says.

    "These moves towards partnerships and idea and technology sharing, only bodes well for the goal of interoperability between Voice over Internet Protocol systems."

  • Gizmo5 CEO Challenges Skype For SIP


    The CEO of Gizmo5 Michael Robertson has responded to last week’s announcement of Skype for SIP by posting a comparison (see below) of the new service and his own company’s OpenSky.

    While welcoming Skype’s initiative, he described it as a "vaporware announcement" with "murky pricing details".

    Writing on his blog, Robertson said he has been a vocal advocate for open standards, both in music with my company MP3.com and in VOIP with Gizmo5.

    He said open standards have always give consumers more choices and ultimately better value.

    "V0IP standards got a huge boost this week with two announcements," he said.

    Roberston said these were Gizmo5’s launch of its SIP for Skype service called OpenSky, which lets any SIP device call Skype and receive their Skype calls, and Ebay’s announcement of Skype for SIP.

    "These announcements are a huge boost for SIP as the open standard which will let calls move freely from any calling device or network," he said.

    "It’s great to see Skype inching towards a more interoperable world. Even if this is a vaporware announcement at least their heart is in the right direction."

    Robertson compared Skype For SIP with Skype for Asterisk, announced last year, saying that Skype’s business offering is not yet available and pricing details are murky.

    In response to Robertson’s blog comments, Skype said its SIP offering is available now.

    While there are other details that will undoubtedly be challenged by Skype, Robertson’s riposte will certainly give any enterprise pondering the services something to chew over.

  • Skype For Asterisk Version Announced


    Skype and Digium, creator and primary developer of Asterisk, the open source telephony platform, have announced the beta version of Skype For Asterisk.

    The move will allow the integration of Skype functionality into Digium’s Asterisk software and enable customers to make, receive and transfer Skype calls from within their Asterisk phone systems.

    Stefan Öberg, vice president and general manager for Skype Telecom and Skype for Business, said: “Throughout our individual histories, Skype and Asterisk have each disrupted conventional communication methods through innovative, cost-effective solutions.

    “We are excited to be working together with Digium to offer small and mid-sized businesses an even more powerful communications solution to conduct business worldwide.”

    Specifically, the beta version of Skype For Asterisk is an add-on channel driver module that integrates Skype Internet calling with Asterisk-based telephony products.

    Skype For Asterisk also complements small and mid-sized business users’ existing services by providing low rates for calling landline and mobile phones around the world.

    Danny Windham, CEO of Digium, said: “Working together with Skype, our goal is to help businesses boost productivity and reap the rewards of feature-rich telephony software, all while saving a substantial amount of money.

    “The Skype For Asterisk beta program is a first step towards adding Skype capabilities to Asterisk-based phone systems and enabling them to reach more than 338 million Skype users.”

    The beta version of Skype For Asterisk will enable business users to:

    • Make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware.
    • Complement existing services with low Skype global rates (as low as 2.1US¢ per minute to more than 35 countries worldwide).
    • Save money on inbound calling solutions such as free click-to-call from a website, as well as receive inbound calling from the PSTN through Skype’s online numbers.
    • Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

    Following the beta period when the product is released, Skype For Asterisk will be sold and distributed by Digium and its worldwide network of resellers.