Source: snapvoip.blogspot.com

22/11/2007

.uk registry chosen to manage the unified telephone numbering system

Nominet, the .uk domain name registry, has been awarded the contract to run the Tier 1 registry for UK ENUM, a standard that will unify the telephone numbering system with the Internet’s Domain Name System.

ENUM is potentially an important part of the convergence of regular telephone services and Internet telephony and it may enable businesses to benefit from cheaper or even free telephone calls in the near future. It is a suite of protocols that makes it possible to relate a domain name to a telephone number, and then use that domain name to identify various communications devices such as fax, mobile phone, voice-mail, email, IP telephony addresses or web pages.

The Tier 1 registry contract for UK ENUM was awarded to Nominet by the UK ENUM Consortium (UKEC), a limited company set up with the recognition of the BERR (formerly the DTI) to administer the UK ENUM top level domain. Nominet won the contract based on its understanding of the issues and challenges relating to best practice management of a top level domain. As part of the process of establishing the registry Nominet will be joining the International Telecommunication Union, Telecommunication Standardization Sector, ITU-T.

ENUM – What is it and how does it work?

ENUM will make it possible to link different VoIP servers so that telephones within businesses that use VoIP technology can connect to other businesses who use other VoIP providers via the Internet. As a result, telephone calls between businesses using this technology will be much cheaper or possibly even free, because there will be no need to connect to the telephone network for the call.

Lesley Cowley, Nominet’s CEO says: "We are delighted that we have been given the opportunity to run the UK ENUM Tier 1 registry. Now the real work begins and we hope that this exciting development will result in cost reduction benefits for businesses and consumers".

Source: snapvoip.blogspot.com

Press release;

Rebranding reflects breadth of telecom offerings, peering and inter-network intelligence

FOSTER CITY, Calif.–(BUSINESS WIRE)–VoEX™ Inc., a major VoIP managed-services provider, announced today it has renamed itself IntelePeer™ Inc. to better reflect the company’s comprehensiveness in next-generation communications services.

IntelePeer’s phone-number-to-IP address registry, global IP peering infrastructure, media transcoding and routing intelligence allow carriers, cable companies, wireless and other voice services providers, universities, call centers, enterprises and eCommunities – such as AIM, Yahoo! and MSN – to reduce their communications costs dramatically. Its infrastructure allows participating peers to send traffic around the world without making costly interim hops to public switched telephone networks, and to build and run intelligent voice-and-data communications applications from end to end.

“We felt we needed to change our name because ‘VoEX’ implies that we’re simply a voice exchange service using VoIP,” said company CEO F. Terry Kremian. “We needed a brand name that better reflects the broader range of our communications offerings and future direction of the company.

“We chose IntelePeer because it stands for Intelligent Telecommunications and Peering,” he added. “Many of our would-be competitors offer voice peering and minutes exchange, or registry services, or gateway functions, as well as long-haul IP trunking services. We differentiate ourselves by offering all of these services combined with the infrastructure to develop next-generation intelligent communications applications.” Kremian said he could foresee, for example, such applications as location-aware (or presence-aware) enhanced person-to-person interactions.

Leveraging industry standards, IntelePeer’s intelligent peering navigation technology assures that every call is delivered in the right format across the most cost-effective, highest quality route possible – providing one-stop shopping for state-of-the-art IP services at a fraction of the cost. These services incorporate:

• Intelligent Least Cost Routing – maximizing quality and cost savings across all networks;
• SIP-based Session Management – reducing termination costs and eliminating payments to long distance and local carriers;
• Any-to-Any Network Interoperability – real-time transcoding, protocol translation and media conversion between disparate networks;
• Phone Number Mapping – linking phone numbers to IP-based addresses using ENUM and other protocols;
• Device Discovery – dynamically analyzing network traffic and call statistics to determine the best route and method to deliver calls while reducing the complexity of building device-specific applications.

“These are the core services that define IntelePeer and will drive future growth,” Kremian said.

He called IntelePeer’s SuperRegistry™ capability – the addressing database and routing algorithms essential to the peering community — a key differentiator, because the wide number of destinations it reaches makes the company a “one-stop shop” for realizing the many cost and Quality of Service benefits of VoIP and direct network-to-network interworking. It also sets the groundwork for support of IP Multimedia Subsystem (IMS) and other IP-based applications.

The IntelePeer SuperRegistry combines its open-standards carrier ENUM directory technology with a global carrier-grade IP peering and TDM interconnect routing infrastructure. Together, this enables customers to originate, terminate and share calls or sessions for mobile, fixed and broadband communications.

In addition, the SuperRegistry platform will allow service providers to create new sources of revenue by deploying innovative SIP-based services such as video, presence and location-awareness.

IntelePeer Inc. (www.IntelePeer.com)

Source: goebel.net

Many of us travel a lot. This means high roaming charges and no local phone number in the country where you are. Some people yet decided to buy local SIM cards and put them in their cell phone whenever they arrive at the airport. For them Truphone has just presented their new Multi-SIM capability, which supports travellers who take international SIM cards with them abroad. Calls to their Truphone number will reach them whichever SIM they’re using at the time.

That’s nice, but even cooler is Roam4Free’s "Get a local fixed line number for any one of 28 countries on your SIM". Instead of exotic numbers from Estonia, Liechtenstein or the Isle of Man they will point local fixed line numbers for any one of 28 countries to their SIM cards when the new version of Roam4Free comes out. The customers can then be called on mobile phones with a local fixed line number.

Local numbers for global SIMs seem to be the new trend. The German company GlobalSIM has also started recently to give local fixed line numbers from 43 countries to their SIM card customers. That’s even better than Truphone’s Multi-SIM capability. But I see the disadvantage that this call forward will surely cost extra and the owner of the mobile phone has to remember many local numbers.

So I think that an entire new number range is missing for worldwide mobile telephony. The best thing would be a cheap interconnect to the ++882 or ++858 number range, or something similar. These are international codes that don’t belong to any particular country, but to ENUM services. It would be great if people could call them from every country for local prices. So you would never have to change SIM card or number for travel. You just had a virtual number, similar to German 032 numbers which don’t belong to a particular city but to VoIP.

Source: snapvoip.blogspot.com

Following an article on Asterisk blog by Russell, "Sneak peek at new features" and provided a link to SVN reository. Following is a part of it (85 lines, there are 235 lines of description). I was surprised that so many features had sneaked by and yet we are happy to use…!

UPDATE!
Seems like formatting makes this impossible to read;

So you have to go Asterisk developer site (SVN repository) see the complete document.Thanks

1 -------------------------------------------------------------------------------    2 --- Functionality changes since Asterisk 1.4-beta was branched ----------------    3 -------------------------------------------------------------------------------    4      5 AMI - The manager (TCP/TLS/HTTP)    6 --------------------------------    7   * Added the URI redirect option for the built-in HTTP server    8   * The output of CallerID in Manager events is now more consistent.    9      CallerIDNum is used for number and CallerIDName for name.   10   * enable https support for builtin web server.   11      See configs/http.conf.sample for details.   12   * Added a new action, GetConfigJSON, which can return the contents of an   13      Asterisk configuration file in JSON format.  This is intended to help   14      improve the performance of AJAX applications using the manager interface   15      over HTTP.   16   * SIP and IAX manager events now use "ChannelType" in all cases where we   17      indicate channel driver. Previously, we used a mixture of "Channel"   18      and "ChannelDriver" headers.   19   * Added a "Bridge" action which allows you to bridge any two channels that   20      are currently active on the system.   21   * Added a "ListAllVoicemailUsers" action that allows you to get a list of all   22      the voicemail users setup.   23     24 Dialplan functions   25 ------------------   26   * Added the DEVSTATE() dialplan function which allows retrieving any device   27     state in the dialplan, as well as creating custom device states that are   28     controllable from the dialplan.   29   * Extend CALLERID() function with "pres" and "ton" parameters to   30      fetch string representation of calling number presentation indicator   31      and numeric representation of type of calling number value.   32   * MailboxExists converted to dialplan function   33     34 CLI Changes   35 -----------   36   * New CLI command "core show settings"   37   * Added 'core show channels count' CLI command.   38     39 SIP changes   40 -----------   41   * The default SIP useragent= identifier now includes the Asterisk version   42   * A new option, match_auth_username in sip.conf changes the matching of incoming requests.   43      If set, and the incoming request carries authentication info,   44      the username to match in the users list is taken from the Digest header   45      rather than from the From: field. This feature is considered experimental.   46   * The "musiconhold" and "musicclass" settings in sip.conf are now removed,   47      since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4   48   * The "localmask" setting was removed in version 1.2 and the reminder about it   49      being removed is now also removed.   50   * A new option "busy-level" for setting a level of calls where asterisk reports   51      a device as busy, to separate it from call-limit   52   * A new realtime family called "sipregs" is now supported to store SIP registration   53      data. If this family is defined, "sippeers" will be used for configuration and   54      "sipregs" for registrations. If it's not defined, "sippeers" will be used for   55      registration data, as before.   56   * The SIPPEER function have new options for port address, call and pickup groups   57   * Added support for T.140 realtime text in SIP/RTP   58   * The "checkmwi" option has been removed from sip.conf, as it is no longer   59      required due to the restructuring of how MWI is handled.  See the descriptions   60      in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf   61      for more information.   62   * Added rtpdest option to CHANNEL() dialplan function.   63   * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.   64     65 IAX2 changes   66 ------------   67   * Added the trunkmaxsize configuration option to chan_iax2.   68   * Added the srvlookup option to iax.conf   69   * Added support for OSP.  The token is set and retrieved through the CHANNEL()   70      dialplan function.   71     72 DUNDi changes   73 -------------   74   * Added the ability to specify arguments to the Dial application when using   75      the DUNDi switch in the dialplan.   76   * Added the ability to set weights for responses dynamically.  This can be   77      done using a global variable or a dialplan function.  Using the SHELL()   78      function would allow you to have an external script set the weight for   79      each response.   80   * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These   81      functions will allow you to initiate a DUNDi query from the dialplan,   82      find out how many results there are, and access each one.   83     84 ENUM changes85....235.....So you have to go Asterisk developer site (SVN repository) see the rest of the lines."Sneak peek at new features"

Source: goebel.net

The Gizmo Project works on a new feature that seems to be still in heavy beta, but sounds quite interesting: Free Local Numbers for their website Gizmo Call. As I am always looking for a free inbound number from Peru I browsed their forum and thought that this could be a solution.

What are Free Local Numbers and how do they work?
A "Free Local Number" is a new feature of Gizmo Call that allows people around the world to call you by dialing a local number from their mobile phone or land line which you can answer on your computer at no cost to you.
[...]
Here’s how it works:
When you call someone using Gizmo Call, a number will display on your screen. This number is what we refer to as a Free Local Number, which will be a local phone number in either the same region of the person you are calling or a number in the least expensive neighboring region for them to call. The next step for you is to tell the person you called to call you back at the Free Local Number on your screen. Optionally, you can email the number or send an SMS to the person’s mobile using Gizmo SMS. The Free Local Number will also appear in the caller ID history of the person you called. Once the person you called has this Free Local Number, they can call you anytime at no charge or at a low cost. You can receive calls on your computer, mobile, or land line phone. To receive calls on your computer, you must first be logged in to Gizmo Call. …

There is a little more information at the Free Local Numbers FAQ. But I have to admit that yesterday I did not get it running. I called a friend in Lima from Gizmo Call. But on the website no free local number appeared, as described, that he could have called back.

Gizmo says The Free Local Number will also appear in the caller ID history of the person you called. To prove this I dialed my German Sipgate number from Gizmo Call. The incoming caller ID was 00858XXXXXXX. What kind of number is this? Which country has 00858 as international code? Definitely not Germany. I suppose it is a net only number blog, similar to the numbers you get from the ENUM provider e164.org. At least I did not find a country for this code. North Korea uses 00850, but I don’t think that Gizmo has a branch in Kim Il Sung’s country.

So I went on trying and called my US-american Truphone number from Gizmo Call to have a little talk with myself. The protocol says that the caller (Me, but from the Gizmo Call website, where I was logged in with my login data from GizmoProject) had the number 00185886XXXXX.

What number is this?

When I called it back I heard only advertising. A female voice said that the caller had used Gizmo Call to dial my Truphone number. Then she spelled the word Gizmo Call before she started again with the same advertising text. Is 00185886XXXXX my free local number for my US number at Truphone? Why can’t I call it then? I was still logged in at Gizmo Call when I dialed it.

To me it seems that Gizmo’s new feature has a great potential but I don’t understand it yet. Also it seems that they want it to work only with their website application Gizmo Call. As we know Gizmo Call allows only a few minutes of free calls from their website. This time shall be used to tell the callee your free local number so that he can call when the time is over.

But maybe this stuff will also work as inbound number for the Gizmo Project? How do they want to distinguish between Gizmo Project and Gizmo Call, which have the same login data? I have Gizmo Project installed in my Voxalot account. My aim is to use this Free Local Numbers feature to ring my Voxalot account, which I can answer on my mobile and my landline phone. A normal Peruvian Call In Number costs 233 dollars annually at Gizmo Project. But maybe the new feature can be a workaround? Other people are also courious about the free local numbers, as you can see in this answer from jfinlayson in Gizmo’s forum:

They indicated that the would be rolling out a beta in "the next few days". I had assumed that when it is ready for testing that it would:

1) Accompany another announcement to that effect, and
2) Either be deployed on a separate server or require you to set a switch somewhere to enable it, so that other Gizmo Call users are unaffected.

That’s traditionally what "beta" has meant, anyway. On the other hand, the software industry’s definition of "beta" seems to have shifted in the past few years, thanks largely to Google, where most offerings are "beta" in perpetuity.

Meanwhile, the strange caller id you’re seeing has me curious, too.

Even more interesting seems what Martin from Voxalot says:

We are currently working with the SIPphone guys on GizmoCall.

All I can say is watch this space.
__________________
Martin

He is the moderator of Voxalot’s forum and a member of the staff. For instance he announces Voxalot’s new features. So his voice is quite important and his words make me even more courious.

Stay tuned!

More info: Markus Gbel’s Tech News Comments: How Gizmo Project’s free local numbers save me 230 dollars annually

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com
It is coming in to light that big name companies are using more and more Open Source products. It is no difference when it comes to VoIP. I remember a bunch of companies that used OpenH323 before SIP came to be the leading protocol.
It is revealed that Cisco is using OpenSER as SIP proxy for Service node in Linksys one communication platform.
OpenSER is not the only Open Source Package in the Service Node;
The Cisco Service Node servers run a collection of open-source and Linksys One software:

• FreeBSD-This is the open-source operating system that runs on all Cisco Service Node servers. FreeBSD provides a mechanism that allows multiple virtual instances of the OS to be spawned and run on the same server, with each virtual OS completely isolated from all other instances. This is the partitioning mechanism used to implement the brand-level services.

• PostgreSQL-This open-source package is used to provide database services on the Cisco Service Node.

• OpenSER-This open-source package is used as the Cisco Service Node SIP proxy.

• BIND-This open-source package is used for Domain Name System {DNS) services. The Cisco Service Node runs its own DNS servers. DNS is used for several functions on the Cisco Service Node, including ENUM-based call routing of SIP calls and branding (each brand is known to the outside world as a separate DNS domain name).

• BIND DLZ-This open-source package allows BIND to use the PostgreSQL database to store its zone information. Dynamically loadable zones (DLZ) allows DNS updates to be reflected immediately when a change is made to zone data in the database. This feature is important because CPE that uses Dynamic Host Configuration Protocol (DHCP) can change its IP address at any time. When this happens, DNS must be updated immediately for the ENUM-based call routing to be able to successfully route calls to the CPE.

• NET-SNMP-This open-source SNMP package runs as an agent on the servers and implements several MIBs.
Thank you Cisco for letting us know so openly and congrats to OpenSER people for providing such quality product.

Links;
The OpenSER mail archive entry and discussion

OpenSER gets CISCOs vote of confidence
Cisco Service Node for Linksys One SN-10 and SN-100 Data Sheet

Source: goebel.net

GigaOM tells some news that seem interesting at first sight:

GrandCentral, the single-phone-number Web-based service launched last fall, is adding support for the free Gizmo Project Internet VoIP service, which may open up a whole new way to decrease spending on international or long-distance calling.

Still in beta, the GrandCentral service is the latest entrant in the often-attempted single phone number scheme. The Fremont, Calif.-based startup uses a combination of VoIP technology and softswitch-based applications to give users ways to tie multiple phone numbers, services (voice mail, etc.) and devices to a single inbound number.

This news seems more like a good piece of Public Relation work than a technical breaktrought to me and I want to tell you why:

If we all would be using ENUM the deal between GrandCentral and Gizmo wouldnt be big news. With ENUM I can already route my PSTN number to my Gizmo account. VoIP users that call this number and do an ENUM lookup can call me for free.

Why bother for another number from GrandCentral?

I can implement the same call routing features of GrandCentral in my analog telephony adapter (ATA). I actually do. So whenever somebody calls my years old PSTN number I can let it ring wherever I want (for instance on my Gizmo Project soft phone or my mobile phone) and I can also filter the callers like GrandCentral offers.

Why pay extra for a service which I can already use at no extra cost at home?

Well, some might say that ENUM is difficult or that they just don’t know it. But the point is that its totally easy to do an ENUM lookup before every call. SNOM VoIP phones do it automatically before they start the call. The VoIP providers could do it automatically before the VoIP call starts. Also they could enlist all their numbers in ENUM. The client wouldnt even notice that ENUM is working in the background. It would just be all over IP and for free. But the VoIP providers dont do it because the earn very well not doing ENUM and charge for calls that technically could be for free.

Others say that we should at least give GrandCentral credit for is their rules processing.

Well, the newest versions of the Fritz!Box do this out of the box. No complicated acronyms needed for installation. Just some clicks in the browser.

Incoming calls can be treated individually. You can for instance block unwanted calls or pass them to an answering machine. Friends and business partners can be redirected to a mobile phone, even if the call signal of the phone thats connected to the Fritz!Box is switched off.

I am using such a Fritz!Box at home. Its connected between my old telephone and the DSL connection. So all my calls go for free or for very modest prices over the internet. I do the configuration in my browser and its very easy. You can see an example here.

In fact the Fritz!Box is an entire PBX and cost me only 30 Euros, because I bought it used. It can do everything that GrandCentral does. Even an ENUM lookup before every call. But this is a hidden function for which you have to tweak the Linux that runs on the box. The Fritz!Box is getting better and better with every firmware update.

So to me GrandCentral is no big news and no necessary service. But I am always courious about new stuff.

Source: snapvoip.blogspot.com

Open source SIP stack OpenSIPStack has been rereleased under a triple licensing scheme to ensure that it can be used by the largest possible number of individuals and development communities. This tri-license aims to address the perceived incompatibilities between Mozilla Public License (MPL), GNU General public license (GPL) and GNU Lesser Public License (LGPL). The stack was previously distributed under MPL 1.0.

Open Source SIP project, openSBC is based on the OpenSIPStack, a fully compliant (RFC 3261) SIP stack designed for stability and scalability, and with a heritage of commercial usage. The project currently contains reference implementations of a session border controller (OpenSBC), Yeya and several components that are useful to developers wishing to use Solegy’s service deployment platform.
OpenSBC

OpenSBC is a reference implementation of a hybrid SIP proxy and B2BUA (back to back user agent) created from the Open SIP Stack core. It is well suited for a number of VoIP implementations. Among other things, it can be used as a Registrar for SIP endpoints, as an entry/egress point for SIP trunking applications, or as a far-end NAT traversal solution.

OpenSBC has been designed for scalability and flexibility. Deployments can grow incrementally with traffic needs because a primary instance can be configured to load balnce sessions across other instances. Each instance may be run on separate servers, or multiple instances may be run on a single server.

OpenSBC can perform the following functions:

Session Border Controller: Full back-to-back user agent (B2BUA) hides network topology with:
- Integrated web UI for basic configuration tasks
- far-end NAT traversal with RTP proxy
- Complete transparency for end-nodes with support for pass-thru of non-standard SDPs,
- Routing using static rules, ENUM or Solegy RTBE
- Comprehensive logs using syslog server
- Encryption of SIP and RTP packets with simple hash

Registrar: Fully standards-compliant with support for pass-through registrations (also referred to as upper registration) and integrated support for presence using SIP/SIMPLE or XMPP.

Proxy: Fully standard-compliant with multi-protocol support (UDP, TCP, TLS*), processing and relaying signals from remote (SIP) and local endpoints.

Presence: Compliant with SIP/SIMPLE and XMPP standards with support for PUBLISH as well as SUBSCRIBE/NOTIFY events.

Event Packages: Support for message-summary information about waiting messages (voicemail) and presence

Solegy™ Offers Free VoIP Softphone for Microsoft Windows — Customized Softphones Available to Service Providers with a Full Range of Calling Features and Back-Office Functionality in a Hosted Environment.

Links;
Open Source Sip website
OpenSIPStack web site
Solegy website

Source: voipcentral.org

The nationwide nationwide residential VoIP service provider my people (yeah, they spell it that way!) has entered into a deal with INFONOXX is the world’s largest independent provider of directory assistance and enhanced information services.

Under the agreement, my people subscribers will have access to residential and business directory assistance provided by INFONOXX. Besides, users will also have access to other INFONXX information services such as DiningSource, HoroscopeSource, MovieAssist, StockSource, WeatherAssist along with lottery results, sports scores, time of day information etc.

The best feature in this arrangement is that INFONXX also connects my people’s users through VoIP call completion. In this process can do away with having to remember, write down the queried number. Instead, one can remain on the line as the directory assistance operator connects you to the requested listing without you having to disconnect and redial.

Read more on this here.

Source: voipcentral.org

A few days back I was having an IM chat with Aswath Rao, perhaps the most technically learned guy around in blogosphere on telephony, especially VoIP, and asked him to enlighten me about ENUM (or Telephone Number Mapping) for that seemed to be in everyones mind these days. He told me that he will send me a short note and he did oblige me.

Though many of us are aware of what the technical story behind ENUM is all about, it is very difficult to put it in simple words and uncomplicated language that non-geeks can understand. It was amazing how easily Aswath could manage it.

For the benefit of all I am reproducing the part of Aswaths mail to me (hope he doesn’t mind me doing this) that deals with the technical aspect of ENUM in simple language:

The primary purpose of ENUM is to terminate calls from PSTN in VoIP. Currently this can be done only at the Class 5 switch that is serving the VoIP customer. I mean if I am a VoIP customer and you dial my PSTN number, PSTN will deliver it my class 5 switch, which will hand it to my SIP Proxy, which in turn will alert me. This may be inefficient, because I may be closer to you rather than my class 5 switch.

So the idea is to have a globally accessible database that maps PSTN numbers to VoIP reachable addresses so that any PSTN switch to look it up and route it accordingly. The mechanics of ENUM is simple: map the telephone number to a hierarchical domain name and then use DNS technology to do the mapping. Since tel nos are hierarchical this is easy. Each digit in the tel no from the left (country code first) becomes the top level domain. So 1234567890 will become 0.9.8.7.6.5.4.3.2.1. Of course this has to be placed under some other root in the domain name tree. There is a controversy here. Some place it under e164.org and others argue against it for political reasons.

Aswath also wrote about his views on the revenue opportunities ENUM databases might offer. However, I wont discuss that here for he should be the one to post his opinions(that I would really love to read more often)in his blog than someone else.

The e164.org is a free service and they are the ones who took this up for what they claim in the About section of their website:

We recognize a need to faciliate the adoption of Voice over IP, and thus we offer our service to the public at large by validating telephone numbers and allowing number owners to determine how Internet communication protocols should interoperate against their phone number.

In my opinion, the ones like e164.org are aggressively asking people to register do have plans apart from providing public service alone. Though not immediate, there are mid-term strategic commercial interests behind it.

The commercial viability may be diminished by the sprouting of many databases, still just consider when it gradually becomes a true VoIP directory, what the telemarketers can pay to get/use the databases most of which are going to be in the hands of private entrepreneurs. (Will there be anything like Do Not Call lists then?) And that is just one way to make money out of many, many more.

The early birds as usual will be the biggest beneficiaries because they would be having the biggest dadabases and would be able to make money when it is there to be made. Anyway, it would not last forever. With multiple ENUM directories, the usefulness of the databases would be substantially diminished.

If there arent billions to be made (obviously there isnt) there are certainly millions for the taking.

I will be discussing more on ENUM both from the technical and business angles in the coming days.

Thank you Aswath. You can read Aswath Rao’s blogs here

Source: voipcentral.org

Russell Shaw in his blog today discussed about ENUM, which is actually the acronym for Telephone Number Mapping. A few days back I was thinking what will be the solution in the future that will take care of the problem of finding a persons number when VoIP will be the most preferred mode of telephony. Given the astronomical numbers of VoIP operators and the problems with integrating E.164 phone numbers with IP addressing using the Domain Name System it is still some distance away in my opinion.

Anyway, talking about the system itself, it works by publishing a DNS zone that can be used by various Internet applications that includes SER, Gnome Meeting and Asterisk.

I shall write more on this soon. This is a subject we just cant miss.

Please read Russells post that discusses in-depth on this.

Source: gigaom.com