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CallerID is cool, but CallerID with Name (CNAM), -- also known as Calling NAMe -- is where it's at. When you go with a traditional phone provider (non-VoIP) they'll often offer you CallerID with Name for an additional fee. But penny-pinching Asterisk & VoIP fans want CallerID with Name too. Unfortunately, too often the SIP trunks or traditional PSTN trunks (analog, T1/E1) connected to your Asterisk IP-PBX don't provide CallerID with Name - just regular CallerID (number only). So how do we solve this dilemma?

Well, Asterisk sits on an IP network, which of course means it can access the Internet. With access to the Internet, "in theory" a special Asterisk script can take the CallerID number, perform a reverse lookup on AnyWho, Google, and 411.com and then change the CallerID data string before it gets passed to the Asterisk extension.

Well, theory is all well and good, but has it been done? Oh yes it has!

Check out this Perl script I found on Team Forrest's website that leverages AGI (Asterisk Gateway Interface), a powerful interface that lets your programmatically control Asterisk.

Calling the CallerID with Name Perl script (calleridname.pl) is as simple as calling this single line of code:
exten => s,n(getname),AGI(calleridname.pl,${CALLERID(NUM)})

Here's the calleridname.pl script:
#!/usr/bin/perl -w
use strict;
use LWP::UserAgent;
$|=1;
sub trim($);

my %AGI; my $tests = 0; my $fail = 0; my $pass = 0; my $result = ""; my $cidnum = ""; my $cidname = "";
my $npa = ""; my $nxx = ""; my $station = ""; my $name = "";

$cidnum = $ARGV[0];

while(<STDIN>) {
    chomp;
    last unless length($_);
    if (/^agi_(\w+)\:\s+(.*)$/) {
        $AGI{$1} = $2;
    }
}

my $AnyWho = '1' ;
my $Google = '1' ;
my $www411 = '1' ;


if(substr($cidnum,0,1) eq '1'){
$cidnum=substr($cidnum,1);
}

if(substr($cidnum,0,2) eq '+1'){
$cidnum=substr($cidnum,2);
}

if ($cidnum =~ /^(\d{3})(\d{3})(\d{4})$/) {
    $npa = $1;
    $nxx = $2;
    $station = $3;
    }
elsif($cidnum =~/\<(\d{3})(\d{3})(\d{4})\>/){
    $npa = $1;
    $nxx = $2;
    $station = $3;
    }
else {
    print qq(VERBOSE "ERROR: unable to parse caller id" 2\n);
    exit(0);
}


if ($AnyWho > '0') {
    print qq(VERBOSE "STATUS: checking AnyWho for name lookup" 2\n);
    if ($name = &anywho_lookup ($npa, $nxx, $station)) {
        $cidname = $name;
        print qq(SET VARIABLE CALLERID\(name\) "$cidname"\n);
        print qq(VERBOSE "STATUS: AnyWho said name was $cidname " 2\n);
        exit(0);
        }
    else {
        print qq(VERBOSE "STATUS: unable to find name with AnyWho" 2\n);
        }
    }
else {
    print qq(VERBOSE "STATUS: AnyWho lookup disabled" 2\n);
}

if ($Google > '0') {
    print qq(VERBOSE "STATUS: checking Google for name lookup" 2\n);
    if ($name = &google_lookup ($npa, $nxx, $station)) {
        $cidname = $name;
        print qq(SET VARIABLE CALLERID\(name\) "$cidname"\n);
        print qq(VERBOSE "STATUS: Google said name was $cidname " 2\n);
        exit(0);
        }
    else {
        print qq(VERBOSE "STATUS: unable to find name with Google" 2\n);
        }
    }
else {
    print qq(VERBOSE "STATUS: Google lookup disabled" 2\n);
}

if ($www411 > '0') {
    print qq(VERBOSE "STATUS: checking www411 for name lookup" 2\n);
    if ($name = &www411_lookup ($npa, $nxx, $station)) {
        $cidname = $name;
        print qq(SET VARIABLE CALLERID\(name\) "$cidname"\n);
        print qq(VERBOSE "STATUS: www411 said name was $cidname " 2\n);
        exit(0);
        }
    else {
        print qq(VERBOSE "STATUS: unable to find name with www411" 2\n);
        }
    }
else {
    print qq(VERBOSE "STATUS: www411 lookup disabled" 2\n);
}

print qq(SET VARIABLE CALLERID\(name\) "$cidnum"\n);
print qq(VERBOSE "STATUS: Unknown name for $cidnum " 2\n);
exit(0);

sub anywho_lookup {
    my ($npa, $nxx, $station) = @_;
    my $ua = LWP::UserAgent->new( timeout => 45);
    my $URL = 'http://www.anywho.com/qry/wp_rl';
    $URL .= '?npa=' . $npa . '&telephone=' . $nxx . $station;
    $ua->agent('AsteriskAGIQuery/1');
    my $req = new HTTP::Request GET => $URL;
    my $res = $ua->request($req);
    if ($res->is_success()) {
        if ($res->content =~ /<!-- listing -->(.*)<!-- \/listing -->/s) {
            my $listing = $1;
            if ($listing =~ /<B>(.*)<\/B>/) {
                my $clidname = $1;
                return $clidname;
            }
        }
    }
    return "";
}

sub google_lookup {
  my ($npa, $nxx, $station) = @_;
  my $ua = LWP::UserAgent->new( timeout => 45);
  my $URL = 'http://www.google.com/search?rls=en&q=phonebook:' .  $npa . $nxx . $station . '&ie=UTF-8&oe=UTF-8';
  $ua->agent('AsteriskAGIQuery/1');
  my $req = new HTTP::Request GET => $URL;
  my $res = $ua->request($req);
  if ($res->is_success()) {
    if ($res->content =~ /<font size=-2><br><\/font><font size=-1>(.+)<font color=green>/) {
      my $temp = $1;
      my $clidname = "";
      if ( $temp =~ /(.+)<font color=green>/o ) {
        $clidname = substr($1, 0, -3);
      } else {
        $clidname = substr($temp, 0, -3);
      }
      if ($clidname =~ /<a href(.+)\//) {
        $clidname = $1 ;
        if ($clidname =~ />(.+)</) {
          $clidname = $1 ;
        }
      }
      return $clidname;
    }
  }
  return "";
}

sub www411_lookup {
  my ($npa, $nxx, $station) = @_;
  my $ua = LWP::UserAgent->new( timeout => 45);
  my $URL = 'http://www.411.com/search/Reverse_Phone?phone=' .  $npa . $nxx . $station;
  $ua->agent('AsteriskAGIQuery/1');
  my $req = new HTTP::Request GET => $URL;
  my $res = $ua->request($req);
  if ($res->is_success()) {
    if ($res->content =~ /Location: <strong>(.*)<\/strong>/s) {
      my $temp = $1;
      my $clidname = "";
      $temp =~ s/&amp\;/&/g;
      $temp =~ s/%20/ /g;
                        $clidname = $temp;
            return $clidname;
     }
  }
  return "";
}


For more details, head on over here.

trixbox CE vs. Asterisk Downloads

December 23, 2008 6:00 AM | 4 Comments
As you may have read, Digium announced a sharp rise in Asterisk downloads for 2008 registering in a whopping 1.5 million downloads for 2008. That got me thinking just how does Asterisk & AsteriskNOW compare with Fonality's trixbox CE distro? Arguably trixbox CE is the either the #1 or the #2 downloaded Asterisk-based distribution. But are they #1 or #2? Let the Fonality vs. Digium download battle commence...

First, we head to Sourceforge, which tracks trixbox CE downloads and see how trixbox CE stacks up against Digium's claimed 1.5 million downloads.
trixbox-ce-downloads-2008.jpg

Date (UTC) Downloads Bytes Served
Dec 2008 * 18,050 6.8 TB
Nov 2008 31,000 12.2 TB
Oct 2008 35,187 15.6 TB
Sep 2008 19,449 9.6 TB
Aug 2008 6,948 3.3 TB
Jul 2008 8,329 3.7 TB
Jun 2008 19,993 9.3 TB
May 2008 21,822 13.3 TB
Apr 2008 18,238 11.0 TB
Mar 2008 41,085 25.2 TB
Feb 2008 27,725 16.9 TB
Jan 2008 43,942 26.4 TB
     
Total 291,768 153.8 TB

* Partial data: End of month not yet reached.
Taking a daily average, the final 2008 total downloads should be ~292,587.

That's 292,587 (year-end estimate) vs. 1.5 million (so far) for Asterisk. So it would appear Digium's Asterisk beats Fonality trixbox CE by over 5X!

Not so fast though. I was pretty sure the Digium download numbers were for Asterisk and AsteriskNOW combined. AsteriskNOW and trixbox CE are more comparable to each other, since both have FreePBX installed, both are .iso bootable CDs, they're both nearly plug-and-play distros, etc.

So to be fair to Fonality's trixbox CE, I should compare AsteriskNOW to trixbox CE. I contacted Digium's Bill Miller, VP, Product Management, and asked him for the breakdown in their 1.5 million downloads.

I told Bill my intentions of making a side-by-side download comparison, but also pointed out I had to wear my skepticism hat:

"The only caveat to any comparison I make between Asterisk/AsteriskNOW & trixbox CE is that you host your own downloads - not a 3rd party like Sourceforge. So there I have to have a healthy level of skepticism for any numbers Digium cites. Nevertheless, I think it'll be an interesting discussion if I post these comparative stats in an article."


Bill explained, "19.8% are AsteriskNOW. We are close to par with them.". Indeed at 19.8% of 1.5 million equates to 297,000 downloads or roughly 6500 more than trixbox CE. Although, I said above I had to wear my skepticism hat, I think the numbers Bill gave me sound credible.

Bill Miller ads, "Due to the other open source packages they have integrated, and the head start they had, we suspect that they have more in trixbox than AsteriskNOW in production but we are taking downloads and share from them."

I would agree with Bill that there are indeed more trixbox CE boxes in production, but it does appear from these numbers that Digium's AsteriskNOW is closing fast in production numbers and already exceeding Fonality's trixbox CE in downloads.

To be perfectly frank, I'm a little bit surprised AsteriskNOW has surpassed trixbox CE, since I'm sure even Digium would admit that trixbox CE, which was formerly Asterisk@Home, had a huge head start and developed a strong community (see trixbox forums and the sheer number of posts & threads). In fact, Bill Miller mentioned the head start in our conversation. Still, the number of downloads between the two is very close and they are neck-in-neck.

Of course, if you add in regular Asterisk downloads it's no contest. Digium wins by a landslide.
Some good news from my company TMC and Digium, the founder of Asterisk that I thought I'd share. If you're an Asterisk fan, you need to be at Asterisk World, which is co-located with the #1 VoIP tradeshow in the U.S., namely TMC's ITEXPO.

digium-asterisk-world.jpgTechnology Marketing Corporation (TMC) and Digium, the Asterisk Company, today announced the educational program for Digium|Asterisk World, scheduled to take place February 2-4, 2009 in Miami. The event is collocated with TMC's ITEXPO East 2009.

TMC and Digium also announced that although the conference content and exhibiting opportunities are more than 90 percent complete, additional opportunities remain for interested Digium partners.
Digium|Asterisk World at ITEXPO is the conference that addresses "Everything Asterisk" for business users, resellers and executive decision-makers. The initial conference sessions featured in the program include:

1. Web-Aware Unified Communications with Switchvox
2. Asterisk for Enterprise
3. Enterprise Pitfalls: Lessons Learned
4. The Case for Asterisk: Call Center App Integration with .NET
5. Multi-Site Open Source Call Center Deployment: A European Case Study
6. Asterisk as a Regulatory Compliance Toolkit
7. Website Identity Management and Authentication Using Asterisk
8. VoIP Transparency: Asterisk and the Economics of Monitoring
9. The Asterisk VoIP Conversion and the Opportunity for Substantial ROI
10. Druid: Case Study for Selling UC Solutions
11. Ingredients for Successful Asterisk PBX Sales

Limited booth space and sponsorship opportunities are still available for companies to showcase their offerings and provide training to Digium|Asterisk World and ITEXPO attendees. Companies exhibiting in the showcase include: Asteria Solutions; Camrivox; Freeside Internet Services; High Powered Help; Interlink Communications Systems; OpenLine Networks; Orecx; TransNexus; Voice Pulse; and Xorcom.

Registration for Digium|Asterisk World and ITEXPO is now open.
"Now in our third year, Digium|Asterisk World's content is evolving to reflect the maturity of the world's most widely used open source telephony software," said Mark Spencer, CTO and founder of Digium and the creator of Asterisk. "Our three-day event at ITEXPO will provide the hands-on training and education that Asterisk developers need to build more advanced and more flexible telephony solutions."

"We're proud of several new and innovative programs that will make the 19th iteration of ITEXPO one of the best ever," said Rich Tehrani, TMC president and ITEXPO East 2009 conference chairman. "Digium|Asterisk World bolsters an already strong developer focus, and other programs like TMC University, the 4GWireless Evolution event, TMC Editors' Week and the balance of the conference content make ITEXPO the one-and-only 'must-attend' event in the communications industry."

ITEXPO East 2009 is the world's largest and most significant communications technology event, featuring hundreds of companies exhibiting on the EXPO floor and hundreds of sessions led by the industry's most prominent thought leaders. The show helps attendees identify the issues and challenges affecting the deployment of communications technologies. It provides a comprehensive forum for evaluating the latest products and services and delivers a face-to-face networking opportunity that service providers, carriers, resellers, distributors, equipment manufacturers and IT executives from enterprise and SMB companies need to cultivate new business relationships. For more information on ITEXPO East, please visit: http://www.tmcnet.com/voip/conference/.
fonality-uae-01.jpg
Back in September I wrote about Fonality's trixbox Unified Agent Edition (UAE) and how it can automatically match all inbound and outbound calls with the corresponding record in salesforce.com's AppExchange, and call data is captured and logged. Apparently, this was still a yet-to-be-announced product I wrote about that resulted in a call from Fonality's CEO Chris Lyman asking how I found out about it. Woops, my bad, Chris.

Well, today, Fonality has officially launched UAE. I spoke with Chris Lyman to learn more about this product. Chris told me, "What we've done is allow managers and agents using CRM and a phone system to make some pretty serious decisions about how they run their business and what we consider some major cost savings."

Chris explained that they pumped the CDR (call data records) from Fonality's PBXtra into Salesforce.com. Once the CDR data is housed in Salesforce.com obvious business benefits can be realized. You can for instance know when customers called you or when you last called a particular customer. Importantly, with the reporting mechanisms you can understand how long it takes to call a lead or opportunity and how that affects your close rate. The reports allow you to for example easily see "neglected opportunities" such as 'opportunities that have not been contacted in XX minutes'.  This lets you see how long does it takes an agent to make a first contact for a lead/customer/opportunity that been created inside the CRM system. Here's a screenshot of some of the reports:
[click for larger image]
fonality-uae-02.jpg

Fonality UAE brings a whole new level of CRM integration with PBXtra, an Asterisk-based system. This tight-level of integration with an Asterisk-based system with the popular Salesforce.com hosted CRM platform is great news for the Asterisk community. While Asterisk-based systems have made in-roads in the contact center space, heavyweights like Nortel and Avaya still win a huge share of the contact center space because of their advanced CRM integration. So the more Asterisk-based solutions offer advanced CRM applications, the better for the Asterisk ecosystem. We'll see more Asterisk deployments in contact centers as a result.

It's worth mentioning that Digium's Switchvox IP-PBX also offers integration with CRM apps, including Salesforce.com and SugarCRM. I saw a demo of Switchvox's web-based panels down in Huntsville, Alabama. I remember it supported querying Salesforce.com on an inbound call (screenpop), but don't recall if it supported querying Salesforce.com on an outbound call. I'm also not sure to what level Switchvox offers reporting on inbound & outbound calls tied to the CRM contact records. I don't believe Switchvox offers contact-based reporting based on CDR information. I've actually been meaning to do a full-review of Switchvox.

In any event, getting back to UAE, I should point out that Fonality queries a local cache of the Salesforce.com database for it's instant when it performs a contact record lookup and is not affected by any sort of Internet outage or Salesforce.com outage. Importantly, inbound calls can be automatically directed to the contact or opportunity owner. This prevents sales personnel from 'stealing' a lead.

"The Frankenstein, bolt-on products from other telephony vendors provide only basic screen pops or click-to-call features. Fonality has a much more sophisticated approach for delivering feature-rich, end-to-end contact center solutions," said Corey Brundage, vice president of marketing and product management at Fonality. "PBXtra Unified Agent is the only offering available today that immediately impacts a contact center's top and bottom line without the need for professional services or a six-figure capital expenditure."

Overview:
  • PBXtra Unified Agent from Fonality combines phone system, call center, and CRM capabilities to provide a complete view of contact center operations. With Salesforce CRM and PBXtra Unified Agent, companies can increase close rates, revenue and profits, while providing much better customer service and dramatically reducing sales and support costs.
  • For less than $20/user per month, PBXtra Unified Agent delivers an immediate return on investment. Agents benefit from productivity enhancements and timely access to more accurate and complete information so they can do a better job selling to and supporting customers. Managers gain deep visibility into real-time and historical agent call activity within Salesforce CRM so they can make more intelligent business decisions based on accurate data.
"Today, contact center managers are often faced with inaccurate or missing data and agents struggle with cumbersome CTI applications, resulting in a poor customer experience," said Richard Gonzales, president of Buvelo Solutions, a salesforce.com consulting partner. "PBXtra Unified Agent solves these problems with a unified solution that helps managers and agents use Salesforce CRM more effectively than ever before."

With PBXtra Unified Agent managers can easily measure, test, and improve sales and support processes, view the effectiveness of human resources, salvage opportunities or identify lapses in service level agreements. PBXtra Unified Agent includes the following features for managers:
  • Track and record all call data in Salesforce CRM automatically
  • View customer call histories
  • Understand how call follow-up times impact profitability
  • Find out the number of calls it takes to convert a lead to a sale
  • Determine which agents are the most responsive
  • Visually rank agent calling frequency and duration
  • Instantly report all leads not called within a specified period
With HUD, Fonality's communications application, you can view desktop alerts with inbound caller name, number, even deal size. In the sample below, the sales agent can see an inbound $120,000 opportunity:
fonality-uae-03.jpg

"PBXtra Unified Agent's deep integration with Salesforce gives our customers even more value from their CRM installations," said Clarence So, CMO, salesforce.com. "Thousands of customers have installed applications from the Force.com AppExchange because partners like Fonality are offering innovative tools that expand the capabilities of Salesforce CRM."

Chris explained that one of reasons why they called it Unified Agent Edition is because they created a unified agent creation and logon. When you make an agent on PBXtra, it displays a drop-down box of your Salesforce.com agents allowing you to map your extension on Salesforce.com account. So it matches username and password across the two platforms. That's how you know the name of the person making a call as opposed to the extension number.

For current PBXtra customers it's a $4,000 upgrade for unlimited agents. New customers the pricing starts at $6,995 for server hardware and software.  Interested parties may also participate in a Fonality-hosted informational webinar taking place on Friday, December 12 at 10 a.m. PT. To register for this webinar, head to www.fonality.com/UAEwebinar.
fbi-ic3.jpgA few days ago the FBI's Internet Crimes Complaint Center (IC3) issued an unclear warning that says versions of Asterisk software are vulnerable to vhishing (voice phishing) attacks, but didn't say which versions, but causing a flurry of news activity on VoIP news sites, tech sites, and blogs.

It all started with this warning from the IC3:
New Technique Utilizing Private Branch Exchange (PBX) Systems To Conduct Vishing Attacks

The FBI has received information concerning a new technique used to conduct vishing attacks. The recent attacks were conducted by hackers exploiting a security vulnerability in Asterisk software.
My esteemed colleagues Rich Tehrani and Greg Galitzine did some research to find out what the story was, including contacting Digium's John Todd.

Here's Rich's take:

Before commenting I waited to hear back from Digium's John Todd who explained that there were some methodology and editorial process issues in this alert - basically no one checked with Digium before going public. As it turns out, after checking with Digium, the FBI quickly revised their statement and everything is fine.

The details are that there was a bug which Digium found in March of 2008 and subsequently patched in version 1.2 and 1.4. Version 1.6 is not affected. Besides, according to Todd, the security issue would arise if system administrators basically disregarded logical security measures like using numerals in passwords.

Read more...

Greg Galitzine also writes about the FBI's warning about Asterisk in an article titled Digium Defends Asterisk Against Fed Warning: "Tempest in a Teapot"

In it, Greg writes:

Todd writes in a blog entry titled SIP Security and Asterisk:

That bug allowed in some cases unauthorized callers to make calls through an unprotected "context" in Asterisk. Due to the nature of the bug there was fairly limited exposure - it would have required a fairly unusual set of configurations to permit fraud, and there was both a simple config file change that would provide protection, as well as an actual patch to the code which we have every reason to believe has been widely implemented by the very proactive Open-Source community using Asterisk in production environments. The bug didn't allow arbitrary setting of caller ID, and would only work in a limited set of circumstances that personally I think would be unusual, though possible.

Early on, Todd had a sense that this might just be a misunderstanding: Sorry for the fuss, and I suspect this is just a tempest in a teapot. Use good passwords, keep your packet filters up, and I'll update things here as we hear more.

Digium's John Todd wrote an excellent blog post describing what happened after he was able to contact the FBI in charge of the security warning. While there was indeed a security vulnerability in Asterisk, it was patched in 1.2 and 1.4 and doesn't exist in 1.6. Thus, someone would have to be using a very old version of Asterisk. And as for the security vulnerability itself which seemed to enable vhishing attacks, John indicates that it was a relatively obscure exploit and "an administrator would have to consciously configure their system in what I believe to be an extremely unusual way in order to be victimized by this particular vulnerability." So indeed in John's own words, it seems to be a tempest in a teapot after all.

John Todd wrote:
As we had surmised, the warning from the IC3/FBI on Friday was just a re-hash of a bug that was fixed back in March of this year.  I was in touch with the agent in charge of this release this morning (after contact attempts on Friday failed) and he understood quickly that the wording was lacking in ways that created questions in the minds of readers, and this was being amplified by bloggers who more clearly outlined the set of questions raised by the advisory/release.  To his credit, the IC3 agent quickly pushed through a set of changes today to the posting which more specifically describes the issue, which indeed is the AST-2008-003 SIP guest permissions problem.

 

John Todd also wrote:
This bug was discovered and patched for 1.2 and 1.4 versions of the software, and 1.6 releases were not vulnerable.  Simple changes to site-specific configurations typically would be all that would be required even on systems that did not get patched or upgraded.  The bug that is described is relatively obscure, and was found by Jason Parker here at Digium.  We didn't know of any "in the wild" exploits back then, though of course there may be some now.  I'm still somewhat surprised that anyone has been able to use this bug to the extent that they were able to mount "vishing" attacks.  While I won't get into the details of configuration specifics, I would say that an administrator would have to consciously configure their system in what I believe to be an extremely unusual way in order to be victimized by this particular vulnerability.
John Todd complains about the "vagueness" of the warning but in an update after speaking to the IC3 agemt, John Todd says "To his credit, the IC3 agent quickly pushed through a set of changes today to the posting which more specifically describes the issue, which indeed is the AST-2008-003 SIP guest permissions problem." (an old issue)

So my Asterisk-loving friends. If you are indeed running patched 1.2/1.4 Asterisk or v1.6 you have nothing to worry about. And if you aren't running these versions, what the heck is wrong with you? And you call yourself an Asterisk fan. Per shame!

P.S. As Rich said, "I am sure by the time Asterisk World rolls around in a few months in Miami, we will all be laughing about this incident and marveling at the opportunity that is open source communications."
vpn-in-a-flash-AspireOne.jpg
Nerd Vittles (Ward Mundy) is the man! He just combined the popular Acer Aspire One NetBook featuring the powerful Intel Atom N270(1.60GHz) processor along with Fedora 10 and his Asterisk-based distro. Nerd just may have cooked up the perfect low-cost, powerful, GUI-rich (low overhead KDE), and portable Asterisk-based appliance! Now you can take your phone system on the road while driving, on the plane, etc. Imagine sticking a 3G/EVDO card in the system while driving and your Asterisk-based PBX is still able to make/receive VoIP over EVDO calls and route them accordingly. He writes about the challenges of building a portable Asterisk-based platform:
Rich has a great interview with Sangoma's VP Sales and Marketing, Doug Vilim about their new B600 Series of Analog voice cards, which can be used in Asterisk-based systems. Apparently, Sangoma's über customizable & modular cards can be a slight cost detriment for those that don't need such modularity.
sangoma-pci-b600.jpg
                        B600D PCI model.

Here's a teaser from Rich's blog:

What is the reason for these new cards to be introduced?

Some of our OEM customers;  those who are building IP PBX appliances Empowered by Sangoma, often do not require the modularity, port customization and single slot expandability that our A-Series analog cards provide.  Also, four PSTN lines and one fax connection is a common configuration that creates economies of scale, which can lead to a substantial cost savings for these customers when purchasing in larger volumes.

Read full story

Also, here are the specs:
  • 4 ports of FXO and 1 Port of FXS.
  • Support for Asterisk®, FreeSWITCH, Yate, trixbox, and PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR, and VoIP gateway applications.
  • Single synchronous PCI interface for all ports.
  • Dimensions: 2U Form factor: 140 mm x 55 mm for use in restricted chassis.
  • Includes both standard and short half-height compatible mounting clips for installation in 2U rack-mount servers.
  • 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
  • Autosense compatibility with 5 V and 3.3 V PCI busses.
  • Fully PCI 2.2 compliant, compatible with all commercially available motherboards, and shares PCI interrupts.
  • Power: 3.6 W (0.3A @ 12 V) and 1.65 W (0.5A @ 3.3 V).
  • Temperature range: 0 - 50°C.
  • Cables are included.
  • Suports Fax: Single FXS Port designed for optimum fax support.
  • Sangoma B600 cards synchronize perfectly with Sangoma's T1/E1 cards and the PSTN clock for error-free fax and modem passthrough over T1 lines.
  • Optional: DSP Echo Canceller
  • G.168-2002 echo cancellation in the hardware.
  • 1024 taps/128 ms tail per channel on all channel densities.
  • DTMF decoding and tone recognition.
  • Voice quality enhancement: music protection, acoustic echo control and adaptive noise reduction.
  • No CPU load as a result of echo cancellation.
  • No additional slot is required.
Thumbnail image for pika-warp-appliance-1.jpg
When I met with PIKA Technologies  at ITEXPO they told me support for FreePBX was coming. Well today, PIKA Technologies announced that PIKA WARP the Appliance is now compatible with the Asterisk-based FreePBX GUI (Graphical User Interface) application. I reviewed the PIKA Appliance recently and was pretty impressed with it. Having FreePBX support is a huge milestone for the PIKA Appliance. FreePBX is a popular user-friendly web application  that makes it easy to setup and configure Asterisk.
freepbx-logo.png
According to PIKA, "While customers often develop their own GUIs, many have told PIKA that if WARP were compatible with industry-standard GUIs like FreePBX, they would be much more inclined to adopt the portfolio. With today's announcement, PIKA has once again demonstrated its responsiveness to the needs of its user base."

"We are very pleased to have supported the effort to adopt the FreePBX application to run in the PIKA Warp environment," said Terry Atwood, vice president of sales, marketing and customer care at PIKA Technologies. "Used in many Asterisk implementations around the world, including Trixbox, FreePBX has proven its value, time and again. When the FreePBX team expressed their willingness to work with us to port to the Warp Appliance, we jumped on the opportunity."

"FreePBX has become the de facto standard for enterprise grade PBX functionality delivered to the SMB business and includes a very rich set of functionality and customization potential," said Philippe Lindheimer, open source community director of Bandwidth.com and leader of the FreePBX project. "But no GUI is complete without a wide range of hardware options to complete the package. We are delighted that PIKA can now include FreePBX in the PIKA WARP and bring our two eco-systems together."

Today's announcement from PIKA follows news of a new partnership between FreePBX and Bandwidth.com, a complete business communications provider offering advanced VoIP, Internet services and managed network services to small and medium businesses. Bandwidth.com will devote significant resources to expand the scope of FreePBX while protecting its charter to remain open source and free.

"The partnership with Bandwidth.com is great news as it gives FreePBX the support it needs to grow while ensuring it remains a free GUI for the entire open source eco-system," said David Clarke, business development manager at PIKA and director of the PIKA Warp Community. "I know the choice of Bandwidth.com was a decision that Philippe made only after months of consideration and sound input from the key developers and contributors to the FreePBX project."

Out of the box, FreePBX provides a long list of features including many typically found only in an enterprise-grade PBX, some examples are:

• Unlimited number of voicemail boxes
• "Follow me" functionality
• Ring groups and call queues
• Unlimited number of conference bridges
• Paging and intercom functionality
• and much more

The PIKA WARP Appliance product portfolio is ideal for deploying small- to medium-sized IP-PBX systems, IVR self-service systems, predictive dialling systems, fax servers and many other features typical of a traditional, purpose-built business telephone system that are often lacking in a computerized system. Compatible with a variety of open-source development platforms, including Asterisk and Linux, the Appliance offers a cost-effective alternative to traditional off-the-shelf computers and plug-in-card network connectivity in a smaller footprint.

Bandwidth.com invests in FreePBX

November 14, 2008 4:06 PM | 2 Comments
freepbx-logo.pngbandwidth-logo.gifBandwidth.com has just made an investment in FreePBX, the popular front-end interface to Asterisk-based distros. I discussed this news with Philippe Lindheimer just a couple hours ago. One of the questions I asked was if Bandwidth.com would get "preferred treatment" within the FreePBX interface, since Bandwidth.com offers SIP trunking. Obviously, if FreePBX gives Bandwidth.com a prominent position in the GUI or they make it "easier" to configure FreePBX (i.e. plug-n-play) that could be a huge boon to Bandwidth.com Philippe said that that isn't part of the investment announcement being made today, however, that is something they are looking at.

As for the purpose of the investment, Philippe said it was mostly due to Bandwidth.com's desire to grow the market and help build the FreePBX community. The idea is that the more IP-PBXs out there, the more SIP trunks, and hence more revenue for Bandwidth.com. I have some further thoughts on this, but I'm pretty busy today and wanted to share the news.

Philippe Lindheimer said, "Part of assuring the success of FreePBX is to make sure that we continue to have strong leadership, community participation and a thriving eco-system of users and partners. I would like to announce a new partnership that will help the project tremendously. I have joined forces with Bandwidth.com as their Open Source Community Director, where we will be devoting significant resources and effort to expand the scope of FreePBX while protecting its charter to make sure it remains open and strong."

One significant piece of news is that Bandwidth.com helped protect the FreePBX's project several months ago when the FreePBX trademark (which FreePBX.org nor Phillipe never owned) was "being shopped around to parties that did not have this project's best interest in mind" according to Phillipe. Thus, Bandwidth.com preemptively purchased the trademark with Phillipe's blessing in order to assure FreePBX was not jeopardized.

You can read Phillipe's blog post about this here which has more details.
AudioCodes has now entered the IP phone arena with a phone that supports wideband codecs (HD) for superior sound quality. Seems a bit odd for a company that makes VoIP PCI and cPCI communication boards and VoIP media gateway modules (PMC form factor) and Analog Media Gateways (2/4/8/24 ports) to be entering the crowded VoIP arena, but enterprise IP phone market is expected to quadruple, from $2.1 billion in 2007 to $8.4 billion in 2001, with some 63 million endpoints being shipped by that time, according the Synergy Research Group. So there is a pretty big pie to go after. Traditional network hardware vendor Adtran has also recently entered the IP phone arena with their Adtran IP700 series (see Adtran IP706 review).

The AudioCodes 300HD Series includes three models: The 310 HD entry level phone with a basic display and interface; the 320HD premium endpoint with a larger screen, and the 350HD executive phone with a color LCD screen. All three models are based on AudioCodes newly announced VoIPerfect software, and include the most commonly used wideband codecs. Of course they are SIP-based so they should work on standards-based IP-PBXs such as Asterisk and these phones are also PoE (Power over Ethernet) compliant.

For more details, check out TMCnet reporter Erik Linask's article. One interesting take Erik has is when he writes, "First and foremost, it owns the DSPs that enable the higher quality, which means it can provide its HD VOIP-enabled handsets at a price point comparable to other high-end non-HD devices".

Now I just have to get my hands on on to review. Stay tuned...
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