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June 29, 2007

Polycom Phones on the way to Government Security Approval

Polycom, Inc. has made available SpectraLink Wireless Telephones with security features required for its pending Federal Information Processing Standard (FIPS) 140-2 certification. The Wi-Fi handsets are the first to be listed on the National Institute of Standards and Technology (NIST) pre-validation list and are in line to be the first in the industry to achieve this government-mandated recognition for security standards compliance.
Polycom's Spectralink Wireless Telephones join Polycom VSX video products and its industry-leading line of MGC/MGC+ unified conferencing bridges for voice, video and content collaboration on the FIPS 140-2 pre-validation list. Polycom says that its full range of FIPS compliant collaboration products support the company's commitment to providing users with a high degree of security, assurance, and dependability.
 
With the introduction of the VSX Release 8.5 with H.460 support for standards-based firewall, network addressing traversal as well as comprehensive Security Mode, Polycom was also the first vendor to offer a video conferencing line with pending FIPS 140-2 compliance in the U.S. and Canada. FIPS 140-2 is a U.S. government computer security standard used to accredit cryptographic modules and certify private sector vendor products for use in government departments and regulated industries that collect, store, transfer, share and disseminate sensitive, but un-classified information.
 
The standard is part of a joint effort by the NIST and the Communications Security Establishment (CSE) and has been adopted by both U.S. and Canadian governments. Four levels of security are available. The SpectraLink handsets should receive Level 2 certification, which meets all the baseline requirements of the specification and also meets requirements for physical tamper-evidence and role-based authentication. The SpectraLink NetLink e340, h340 and i640 models are pre-validated for FIPS 140-2 Level 2 security and will begin shipping next month. They range in price from $449-$699.
 
Source: Tom's Hardware 

Posted by Dal at 09:58 AM | | TrackBacks (0) Posted to Polycom

June 27, 2007

QueueMetrics 1.4.0 for Asterisk PBX Released Today

Note: After actually using this for a couple years I have to say this is one of the best 3rd party applications for the Asterisk PBX.  Adding this to a call center is like taking a breath of fresh air because of the report and real-time stats.  I want to setup a wall board at our office for fun.  
 
After a few months of developement, we are proud to release QueueMetrics 1.4.0

This release adds a very large number of new features and bug fixes:

Major New Features:
- New master engine! It should be 4x faster and 2x as memory efficient as QM 1.3, though it's tracking much more information. It's 100% compatible witth the old configuration switches. (#2, #4, #69)
- New clustering engine! clustering is now fully supported for historical, live and agent's page data. The configuration is a bit different from version 1.3.3 (#23)
- New Agent's page: an agent can log on, log off, go to pause and terminate pauses.
- Call codes tracking: your agent can associate a call code to each incoming/outgoing call and QM will report on it.
- Pause codes: an agent can mark WHY he goes on pause from the ACD, and QM will report on it.
- Multi-stint calls: if a call has been handled by multiple queues, eg it has been passed to an overflow queue, it is now possible to track its progression (#129)
New Features:
- The key pressed to exit a queue is now tracked and a graph has been added. (#65)
- Full queues list on the real-time page (#135)
- Closing time of the last call for agents is shown on the real-time page (#136, #143)
- The Realtime Live page is now fully I18N (#94)
- All new pages are fully I18N compliant (#118)
- A new field on most analyses tells from which server the data came from (useful for clustered setups) (#105)
- A new configuration switch lets you choose to use the starting or ending channel name for a call, in the case they're different (#15)
- It is possible to use a third party call monitor (e.g. OrecX) through an XML-RPC interface. (#40)
- Sometimes running Agent filter would lead to wrong results because not all data was correctly fetched from the database; now this is fixed (#64)
- Users can now change password by themselves; new user page. (#96, #100)
- Icons everywhere, and a bit lighter than the previous ones. (#123, #130)
- All icons have a popup message that appears by hovering over them. (#140)
- The MySQL Storage screen is now key-protected (#147)
- Supervisors: a supervisor holding the correct key can run a report for the team he's supervising.
- Improved DNIS filtering / Area code analisys

...plus over 50 bugs fixed and little improvements - see the changelog .In total, this release produces over 150 different results.

You can download the latest version immediately from the downloads page at http://queuemetrics.com/download.jsp together with the updated 130-page User manual. As an alternative, if you run RHEL/CentOS/TrixBox/AAH, you can install it automatically using yum - see the installation page at http://queuemetrics.com/install.jsp

I look forward to hearing from you.

Yours,
lorenzo emilitri
Partner & CEO
Loway

Posted by Dal at 10:53 AM | | TrackBacks (0) Posted to Asterisk Software

June 18, 2007

VoIP Security Fundamentals

Note:  Computer Weekly has a great Monday morning article about VoiP security and how a PBX can affect it.

Voice over IP (VoIP) security is a challenge for IT staff because IP telephony (IPT) brings with it not only the security problems of data networks but also new threats specific to VoIP. In this fundamentals guide, learn about network security threats and emerging IP telephony threats, and how to secure your VoIP systems and endpoints from them.

How to think about VoIP security

Security requires constant vigilance. Security is all about the protection of resources -- data, devices, networks, applications and people. While access to these resources is the goal of the user, securing access to these resources means the administrator of the resources wants to limit, even prevent, that access. Enterprises already have many security problems with their data network infrastructure, servers, desktops and software. Adding VoIP and IPT to the mix only compounds the security problems.

There are several security issues with VoIP networks:

  1. The VoIP/IPT devices, servers, gateways and phones share the data network and inherit the data network's security problems.
  2. There will be data attacks on voice devices such as Denial of Service (DoS) and malware.
  3. It is easier to eavesdrop on IP calls than on TDM calls.
  4. The centralized TDM PBX is gone. The VoIP/IPT resources are scattered around a network.
  5. The operating systems of the VoIP/IPT devices are less secure than the TDM operating systems of the past.
  6. Systems (PBX) administration can be located at multiple locations and can be accessed by Web browsers.

VoIP security vs. voice quality

It may not be apparent, but security tools and solutions will conflict with voice quality. The more barriers there are in the network and endpoints for security purposes, the more interference there will be with voice quality.

One of the first issues is the firewall. The firewall can block calls because it cannot process the signaling or dynamically allocate the UDP ports for the calls to pass through it. Firewalls may not read the QoS markers in the voice packet, thereby degrading the packet delivery service. Other issues include:

  1. Voice packets, when they pass through security devices, can cause added delay, jitter and packet loss during the call.
  2. Intrusion prevention systems perform considerably more processing than a firewall and have been proven to cause voice quality degradation.
  3. Encryption and decryption add delay to the calls.
  4. VPN connections encrypt the QoS markers. The routers consequently cannot deliver the needed QoS for the voice packets.

The security vs. voice quality conflict will be hard to resolve. The voice manager, obviously, does not want poor-quality calls. If the calls are poor, then why have calls travel over the data network in the first place? The security manager does not want to open the network and endpoints to security exposures that will not only compromise the voice services but weaken the data functions as well. This will require a great deal of negotiation and compromise. Security is important, but not at the cost of an unacceptable voice service.

Finding vulnerabilities

There are two sites that demonstrate the software security threats to the data functions. These lists now include VoIP/IPT vulnerabilities. Both lists are funded by the federal Homeland Security Administration. The first is hosted at Mitre. This site has a dictionary of standardized names and descriptions for Common Vulnerabilities and Exposures (CVE). The second site hosts the National Vulnerability Database at the federal National Institute of Standards and Technology (NIST).

The voice staff has not encountered many security problems with traditional TDM PBXs, but voice staff may not be prepared for the new range of security issues that will become evident as the enterprise migrates to IPT or VoIP. The VoIP personnel will either have to take on their own security responsibilities or use the existing security personnel. In either case, the new responsibilities for VoIP security will require education, possibly some organizational adjustment, and expanded job descriptions.

Click Here to Continue Reading 

Posted by Dal at 11:18 AM | | TrackBacks (0) Posted to VoIP Security

June 15, 2007

Asterisk PBX 1.2.19 and 1.4.5 Released

The Asterisk development team is proud to release Asterisk versions 1.2.19 and 1.4.5. There has been a very large number of bugs fixed since the last release, including crashes and other critical issues. There were 244 commits to the 1.4 source tree and 74 commits to the 1.2 source tree since the last set of releases. Also, keep in mind that the release branches are only changed to fix problems. In the same time frame, the development tree had 439 commits. There is a ChangeLog available in the tarball of each release with a complete list of changes.

Keep in mind that the date for moving the 1.2 release series to security fix maintenance only will be coming within the next couple of months. We strongly encourage everyone to migrate to Asterisk 1.4. Don't forget to read the UPGRADE.txt file in 1.4 for important information regarding upgrading from 1.2.

These releases are available for download from ftp.digium.com. They are distributed as both tarballs and patch sets against the previous releases. All release files have been signed with GPG keys from members of the Digium software development team to ensure authenticity. As always, thank you very much for your support!

Click Here to Download 

Posted by Dal at 04:40 PM | | TrackBacks (0) Posted to Asterisk Releases

T-Mobile Trying To Stop Truphone VoIP Calling

Note:  Andy Abramson sent this over and I would say this looks like some pretty strong-arm tactics on TM's part.  We really need to define what is open market and what can be limited by mobile providers.
 
This past week Truphone quietly introduced an advance look at version 3.0 which features a series.  At the same time Truphone is finding themselves in a sort of battle with a major UK operator T-Mobile. T-Mobile is trying a new tactic to kill off cheap mobile phone calls with a so called “T-Mobile policy decision” that denies access to cheap calls and leaves T-Mobile isolated among mobile operators, as the rest of the UK mobile operators are indeed allowing calls to reach Truphone numbers. Obviously this is more than a Truphone vs. T-Mobile issue, as it will affect all new carriers who want to play in the Mobile VoIP and Voice 2.0 arena.
Here are The Core Facts:

* T-Mobile has refused to interconnect with mobile VoIP provider Truphone: T-Mobile customers making a call to Truphone's number range (07978 8xxxxx) will not be connected.

* T-Mobile refuses to interconnect with operators offering VoIP as a matter of policy.

* However T-Online Ventures, the venture capital arm of T-Mobile's parent company Deutsche Telekom, has just invested in VoIP provider Jajah; T-Mobile connects with BT Fusion, a VoIP service; and T-Mobile has also announced a trial of a VoIP service in USA and Germany.

* T-Mobile is required to 'make calls or otherwise transmit electronic communications to every normal telephone number', which it has refused to do in the case of Truphone and other VoIP operators.

* The other four UK major mobile network operators - 3, O2, Orange and Vodafone - all interconnect with Truphone, leaving T-Mobile isolated on this issue.

* T-Mobile's current adverts display the slogan "Setting the internet free".

* Currently a 'beta' service, Truphone's is prevented from launching fully until the 07978 8xxxxx number range is fully interconnected. Beta service customers are presently unaffected by this issue.

* Other mobile operators have employed different methods to prevent VoIP uptake. There has already been the well-publicized removal of internet telephony functionality from Nokia's popular N95 handset by Vodafone and Orange, and new data tariffs published by Vodafone that mean customers using VoIP will be charged more than for web browsing or email.

Quotes:

James Tagg , Truphone's CEO, said:


"If I were a shareholder I'd be asking some tough questions about whether T-Mobile is prepared for the internet age. It looks like a company in chaos with no coherent strategy for VoIP: it is both resisting VoIP and buying it, and at the same time running ads saying it sets the internet free. Maybe the left hand simply doesn't know what the right hand is doing."

"T-Mobile's move is the most aggressive act but it isn't alone in trying to find ways to slow down mobile VoIP. Vodafone and Orange tested one way by removing internet telephony from their branded Nokia N95 handsets without telling their customers, and Vodafone is planning to charge more for VoIP traffic than for web traffic on its new mobile web service."

"T-Mobile will argue that it is not 'blocking' Truphone but is merely negotiating on price. T-Mobile receives 35p per minute from its customers but is offering only 0.21p per minute to Truphone even when Truphone's costs are 9p per minute to terminate the call."

Links to Other Content:

 
Ofcom statement on the 'Regulation of VoIP Services' <http://www.ofcom.org.uk/consult/condocs/voipregulation/voipstatement/voipstatement.pdf>
 
Telecommunications Act 2003 <http://www.opsi.gov.uk/acts/acts2003/20030021.htm>
 
Original report [from The Register] on the removal of internet telephony functionality from the Nokia N95 by Vodafone/Orange <http://www.theregister.co.uk/2007/04/18/n95_crippled/>
 
Original report [from The Register] on Vodafone's new data tariff <http://www.theregister.co.uk/2007/04/30/vodafone_data_tariff/>
 
Video of T-Mobile and Truphone <http://www.youtube.com/watch?v=eH0auDTbHx4>
 
Truphone web site <http://www.truphone.com/>
 


If you have any questions or wish to speak with James Tagg, CEO of Truphone, please let me know.

Regards,
 

Posted by Dal at 01:57 PM | | TrackBacks (0) Posted to VoIP Politics

Unyte 2.5 Desktop-Sharing Tool Released

WebDialogs, announced that the latest version of its easy-to-use real-time collaboration software, Unyte 2.5, is now available, following a successful beta-test period. The new version of Unyte features enhanced picture quality and quicker image transition times between host and viewer monitors. It also offers a new remote-assistance capability that lets users accept control of another computer, for example, to edit a document or troubleshoot their product on a customer's computer.
“The feedback to 2.5 during our beta test has been very encouraging,” said WebDialogs CEO Lou Guercia. “Users have told us that the improved image quality is readily apparent. Many also appreciate the new P2P option. But for those who don’t know P2P from B2B, Unyte continues to be the easiest-to-use desktop- and application-sharing tool anywhere.”
 
Two versions of Unyte are available. Unyte Lyte offers basic desktop and application viewing capability between the host and one viewer, and is free of charge. Unyte + is an upgraded version with additional features such as select application sharing and remote control of applications by participants. Unyte + costs as little as $30 annually. Unyte Lyte can also be used as a one-click plug-in with Skype, Yahoo! Messenger and Salesforce.com.
 
Visit www.unyte.net for more information and to download Unyte 2.5.

Posted by Dal at 01:51 PM | | TrackBacks (0) Posted to VoIP Software

Making Skype Calls from a Blackberry

Note:  Now this is what I call disruptive.  Love it.  Good read with some nice links to more information.

Today I called Alec Saunders via Skype.  Not terribly unusual, really... except that I initiated the Skype call from my Blackberry!  Back on Monday, Jim Courtney over at Skype Journal had posted a teaser about an application for the Blackberry that allowed you to use Skype IM.  Today my curiosity got the better of me and I had to IM Jim to get the info. I downloaded the app and promptly had to try it out... I guess by virtue of Alec being at the top of my contact list, he was my victim (I also know from past experience that he's open to this kind of thing).

Tonight Jim has spilled the goods for all of us in the post rather dramatically titled "Skype on the Blackberry - A New Era in the AlwaysOn World". It turns out to be an application called "IM+ for Skype" from a company called ShapeServices, which makes a variety of software products for PDAs (and specifically Blackberries). 
 
It turns out there was a news release about this back on May 7th, but I didn't see any coverage of this at all.  The app itself costs $25, but you can also download it and use it for a free 7-day trial.
 
 

Posted by Dal at 01:17 PM | | TrackBacks (0) Posted to Skype

June 14, 2007

Sugar NoteIn : SugarCRM Integration with Asterisk PBX

Note: This nice little application was emailed in  one of our readers.  Personally I use SugarCRM in the office to track leads.  It is a fine piece of open source software. 
 
Sugar NoteIn* is an open-source module for sugarCRM wich provides a VoIP integration with Asterisk. Modulis provides, with Sugar NoteIn*, an original solution to help you to save time and be more efficient when you receive a phone call.

For an incoming or outgoing call, this module provides a quick notepad and a detailed view of your sugarCRM contact. This module makes the accessibility to your CRM easier and optimizes your resources and time while using your CRM.

 

Detailed presentation : http://www.modulis-voip.com/en/sugar-notein/presentation/

Screenshots here :
1
2
3


Download : http://www.modulis-voip.com/en/sugar-notein/download/

Installation how to : http://www.modulis-voip.com/en/sugar-notein/server-installation/

Also registered in SugarForge.

Posted by Dal at 10:32 AM | | TrackBacks (0) Posted to Asterisk Software

June 12, 2007

Apple iPhone in the Enterprise?

Note:  I have been thinking the same thing about how open the iPhone will be.  To truly be adopted into enterprises they will need to have the phone open enough to get some integration in enterprise applications. 

No doubt the Apple iPhone will create a stir regarding the kind of services and applications dual-mode devices can deliver. But the success of dual-mode -- cellular plus Wi-Fi -- in the enterprise may depend on the willingness of cellular carriers to share their networks with Wi-Fi providers. One source tells me that both T-Mobile and Cingular, now AT&T, will drop any VoIP phone call originating from a handset if Skype is the service provider.

This kind of anticompetitive behavior won't last long, but at the moment, if true, it is worth considering, especially as carriers themselves begin dabbling in VoIP. T-Mobile, for one, is currently test-marketing its own VoIP service in Seattle.

If blocking VoIP calls sounds far-fetched, consider that Frank Hanzlik, managing director of the Wi-Fi Alliance, told me other carriers have VoIP capability but don’t want that fact known. He refused to mention which carriers.

Before standardizing on a dual-mode phone, enterprises should also consider the true ubiquity of the network. Uptime in airports and coffee shops is not the same as door-to-door coverage and would preclude using any serious business application on a dual-mode phone.

I spoke with Mohan Natarajan, vice president of engineering at Firetide, about this concern.

Firetide, and other mesh-networking companies, can greatly extend your Wi-Fi capabilities by deploying mesh nodes throughout a city. At present, Firetide has 3,000 nodes in a 50-square-mile section of Singapore, giving users 70Mbps access. Here in the States, however, mesh hasn't caught on, except in the public-safety sector, for which Firetide has deployed mesh networks in Dallas and Phoenix.

Click Here to Continue Reading 

Posted by Dal at 01:07 PM | | TrackBacks (0) Posted to VoIP News

June 11, 2007

AstriDevCon 2007 Summary

First, I'd like to thank everyone who participated in AstriDevCon 2007; the event was definitely a success!

We spent a great deal of time in architectural and planning discussions, and the results of those discussions will be posted here by various people over the next week or two. We also got some nice presentations on various Asterisk and Asterisk-related projects, and in addition Tom Cross from IBM ISS spent 90 minutes talking about how we can do a better job writing code and designing Asterisk to avoid security vulnerabilities.

I also want to personally thank the host organization (Georgia Institute of Technology Information Security Center - GTISC) for allowing us to use their fabulous facilities... the conference rooms were first-rate, the WiFi access was superb and they even arranged for comfortable weather for our walking trips between the Klaus building and the hotel :-)

We had some corporate sponsors that helped defray the costs of the event, including Digium (obviously), but also:

LumenVox (http://www.lumenvox.com)
NetXUSA (http://www.netxusa.com)
Aheeva (http://www.aheeva.com)
Vyatta (http://www.vyatta.com)

Finally, I'd like to thank everyone who attended, and their sponsors/employers/etc. who allowed them to join a group of Asterisk geeks and hang out for a week... (alphabetical order)

Adhearsion - Jay Phillips
Amcat - James Texter
Asteria Solutions Group - Tim Ringenbach, Jeremy Pepper
Bird's The Word Technologies - B.J. Weschke
Edvina.net - Olle Johannson
IBM Internet Security Systems - Tom Cross
INRIA - Philippe Sultan
jabber.org/Jabber, Inc. - Peter Saint-Andre
Logivox Systems - Terry Wilson
LumenVox - Randal Rand
Novell/SUSE - Karsten Keil
Nuvio - Curt Moore
OpenNMS - Jeff Gelbach
PopVox - Dave Troy
Securax - Atanas Bachvaroff, Joachim Vanheuverzwijn
Snom - Hirosh Dabui
Sokol & Associates - Steve Sokol, Jared Smith
St. Joseph's College - Brian Capouch
Sun Microsystems - Stephen Uhler
Switchvox - Brian Degenhardt
TechHat - Glenn Dalgliesh
TransNexus - Di-Shi Sun, Dmitry Isakbayev
Truphone - Ed Guy
VCCH - Howard White, Tilghman Lesher, Tom Moyer
Viagenie - Marc Blanchet
Voiceroute - Ming Guang Yong, Navin Kumar
Vyatta - Robert Bays
Westhawk - Tim Panton
Wimba - Steve Kann, Mihai Balea
Xelatec - Steven Henke
Xorcom - Tzafrir Cohen

--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

Posted by Dal at 10:46 AM | | TrackBacks (0) Posted to Asterisk Development

June 07, 2007

Asterisk iPBX Event in Brussels, Belguim Oct 9-10th 2007

Note:  The good people at Profoss emailed in about there event and made an offer to give AVN blog readers an additional discount at registration.  It is limited to 100 so do sign-up soon to get the good deal.

A Profoss event is organised the 9 and 10 october in Brussels, with the subject "Asterisk and voice over IP". This event will feature Kevin Flemming, co-maintainer of Asterisk, and several companies developing around Asterisk.

Registration is open at http://www.profoss.eu/register/. Registering before 30 june with the code: ASTVOIPNEWS gives you 30% off the registration fee. As participation is limited to 100 attendants, it is advised to register rapidly.

Profoss is geared towards IT professionals. People behind Profoss are IT professionals and will take care to position Profoss as a reliable source when you need information about Free and Open Source Software for your enterprise.
 

Posted by Dal at 11:43 AM | | TrackBacks (0) Posted to Asterisk News

June 05, 2007

Japan Plans Tighter IP Telephony Rules

The communications ministry in Japan plans to tighten regulations for Internet-Protocol (IP) telephony, after a blackout incident affected millions of customers last month, according to a report. The ministry also plans to strengthen the maintenance and security of national IP communication networks by and employing more professional engineers to deal the accidents, Japanese daily Asahi Shinbum said Monday.
In May, Nippon Telegraph and Telephone East Corp. (NTT East) and NTT West Corp., Japans main IP service providers, reported an accident where 3.18 million fiber optic circuits froze, making IP phone connections impossible for 3 1/2 hours.

The ministry regarded the accident as serious, and it began expanding the list of accidents where IP telephone service providers should notify the ministry.

Current rules require IP telephone service operators to report accidents to the ministry only when 30,000 or more subscribers cannot make phone calls for more than two hours.

For now, minor accidents such as difficulty in connecting to other phones via the Internet or delays in receiving e-mail messages through the Internet dont have to be reported to the ministry, so the ministry cannot collect data for such accidents directly.

The ministry, however, acknowledges some minor accidents might cause serious disruptions of telephony and other services. If data can be collected on a wider scale, it can be studied and the ministry can offer solutions to prevent recurrences.
 
Source: IB Times 

Posted by Dal at 01:36 PM | | TrackBacks (0) Posted to VoIP News

Coalition wants FCC to save 700MHz Spectrum for Wireless Broadband

Note:  I hope they do, it would be great to have a real wireless internet service that had some serious range. 

The FCC is currently preparing the rules for the upcoming 700MHz spectrum auction, but a coalition of nonprofit group and techies argues that simply selling the spectrum to the highest bidder could be a disaster for the US. Instead, the group wants this prime spectrum made available under special rules that could lead to a "third broadband pipe" that uses wireless technology.

With the 2009 transition to digital TV coming ever closer, the FCC needs to decide what to do with bits of the spectrum vacated by broadcasters after the changeover. 60MHz of prime spectrum in the 700MHz range will be made available in some form of auction to take place at the end of this year, but that auction could proceed in many different ways. The established players would love to see the spectrum auctioned off to the highest bidder (the usual process) as the incumbents would likely end up with it. Because spectrum in that frequency range travels so far and easily penetrates walls, it's incredibly valuable to communications companies. But even if the incumbents have no plans to use the spectrum, snapping it up could make good business sense because it would keep competition from developing.

That new competition is exactly what Lawrence Lessig, Cory Doctorow, Craig Newmark, and plenty of others want to see. An ad-hoc coalition of interest groups submitted a letter to the FCC yesterday—the last allowable day for comments on the auction—calling on the agency to "ensure that a significant portion of the newly available airwaves go to new market competitors." The dream here is that wireless broadband will provide viable competition to cable and DSL. The nightmare is that incumbents get hold of the spectrum and squat on it.

The letter puts it this way: "If the FCC simply gives the highest bidder exclusive rights over the new airwaves, phone and cable companies could become permanent gatekeepers of the airwaves—continuing their record of keeping new competition and innovation out of the marketplace."

Click Here to Continue Reading 

Posted by Dal at 01:31 PM | | TrackBacks (0) Posted to VoIP Politics

June 01, 2007

Telephone Reminders 3.0: The Free Asterisk Telephone Reminder System

Today we're updating our Telephone Reminder System for Asterisk to version 3. The original system let you schedule reminders for future events and, when the appointed date and time arrived, Asterisk swung into action and placed a call to the number you designated to deliver your customized reminder message.
Today we add the bells and whistles that just about everyone using the original application requested. Now you can set up recurring reminders that call daily or on weekdays as well as weekly, monthly, and annually.
 
This means it can be used to wake you up in the morning, or to remind Granny to take her medicine every day, or to remind your Little League team of practice times and locations, or to remind you and your customers of scheduled and recurring events.
 

Posted by Dal at 08:14 AM | | TrackBacks (0) Posted to Asterisk Help

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