Let’s talk about some of its new capabilities.

A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters. What does this mean for users? First, it means that Asterisk now comes with some additional codecs, including the 32kHz variant of the Speex codec (previous versions of Asterisk only supported the 8kHz or 16kHz variants), Skype’s Superwideband SILK codec, and pass-through support for the 44.1kHz and 48kHz variants of the CELT format.

Astute readers will note that earlier versions of Asterisk were only capable of operating on 8kHz and 16kHz sampled audio, and that the aforementioned newly-supported codecs operate at rates other than these. You’re absolutely correct. In order to support these new codecs, Asterisk 10 has also been provided with support for a variety of super and ultra-wideband sampling rates, all of which are supported as file format types for file playback or recording.

One of the best ways to show off the enhanced media capability is with an all new conferencing application. Asterisk 10 features a fully-functional software-driven conferencing application called ConfBridge. Astute readers will again note that ConfBridge was first made available in Asterisk 1.6. Again you’re correct. But, the ConfBridge of yore was sparsly featured and was only capable of operating at 8kHz (or 16kHz with a compile-time option). The ConfBridge of today (Asterisk 10), supports a full range of sampling rates (8, 12, 16, 24, 32, 44.1, 48, 96, and 192kHz) for its participants – it automatically selects the best sampling rate based on the participants’ native codec. Besides that, ConfBridge provides much greater control over in-call menus (customizable on a per-DTMF, per-caller basis), per-user profiles, as well as dynamic conference creation.

Speaking of showing off, what better way to “show” off capability than with some video? ConfBridge in Asterisk 10 provides basic video conferencing support. That’s right, if you and your friends have video-capable SIP devices, that all speak the same video codec and profile, you can create multi-party video conferences. The video played out to each participant follows a marked user; a user chosen by DTMF; or, the most useful, the user who’s currently talking.

In addition to the improvements to voice and video capabilities, Asterisk 10 can also improve your faxing experience. Asterisk 1.4 is capable of T.38 pass-through, where one T.38 capable endpoint can send a fax directly to another T.38 capable endpoint – usually a couple of SIP peers. Asterisk 1.6.X and 1.8 are capable of T.38 termination, where Asterisk can read/write TIFF files from/to T.38 endpoints. Now, with Asterisk 10, transparency between non-T.38 and T.38 is possible, with its T.38 Gateway (aka T.38 Relay) support. So, if you have a Digium E1 card terminated to the PSTN, you can now carry a fax directly from the PSTN to a T.38-enabled SIP device, without the intermediary “termination” step of TIFF conversion required in earlier versions of Asterisk. Or, if you’ve got a T.38 capable SIP provider, and a second provider that isn’t capable of T.38, you can mediate between the two; again, without any intermediary “termination” step. That means much faster and more reliable faxing in Asterisk 10.

What do we need from the community?

Testing, Testing, Testing.

Besides these great features, Asterisk 10 features a number of other important changes, many of which are under-the-hood. All of these under-the-hood changes mean that there are probably (definitely) some broken bits that Digium hasn’t found. So, to find them, we need your help. Please report any issues you find to the issue tracker.

Again, please report any issues (bugs, etc) that you find to the Issue Tracker.

It’s also useful to see successful test reports. Please post these to the asterisk-devmailing list. Or, if you IRC, then hop on over to the #asterisk-testing channel on theFreeNode network so we can work together in testing the many parts of Asterisk.

As always, we thank you for your participation.

Related posts:

1. Asterisk 10.0.0 Beta 1 Now Available
2. Asterisk 1.8, now in Beta
3. Asterisk 10.0.0-beta2 Now Available

We’ve had quite a few ‘major’ releases of Asterisk since then, including 1.2, 1.4, and most recently, 1.8. Each of these releases has included significant changes, and sometimes architecture-improving changes. Each of them has also included substantial new functionality for Asterisk users. Along the way, we’ve been asked by many people in the community when we are going to start working on (or release) “Asterisk 2.0.” Typically, we’ve responded by saying that will not happen until we can really justify such a significant change in the version number. Many open source projects have gone through similar progressions, and quite a number of them have in fact undergone complete (or nearly complete) rewrites resulting in new ‘major’ versions.

The Asterisk project, however, has tried to avoid that level of disruption to its users. Instead we’ve focused on attempting to provide as much backwards compatibility between major releases as we could. As a result, each time we’ve released a new, major version, the decision has been made that “No, this isn’t Asterisk 2.0,” and we’ve continued with the version numbering scheme that Mark Spencer started all those years ago.

Over the past few months though, as we’ve approached the first beta release of the next major version of Asterisk, we’ve been having a somewhat unexpected conversation: about just how different this release is going to be from the releases that most users in the community are using on their production Asterisk systems (primarily Asterisk 1.4, although there are still a lot of 1.2 users as well).

In fact, even though it’s been an evolutionary process, not a revolutionary one, the next major Asterisk release really will be substantially different from Asterisk 1.4 in some very noticeable ways: wideband conferencing support, basic video conferencing support, support for a number of additional VoIP technologies, full-fledged FAX support, and many others.

That has raised the question: Is this Asterisk 2.0? If not, will there ever be an Asterisk 2.0? After quite a lot of discussion, we came to the conclusion that this isnot Asterisk 2.0, but that it’s also quite unlikely that there ever will be such a release; it wouldn’t be in the community’s best interests to release something that is fundamentally different (and not compatible) but still call it ‘Asterisk.’  That then leaves the question we’ve been asked by many people: If there’s never going to be an Asterisk 2.0, why continue to call these releases “1.x”? What does the “1″ mean, if it’s never going to change?

The conclusion that we’ve reached, and that we hope you’ll agree with, is that Asterisk is always going to be Asterisk, and that you don’t need a “1.” prefix on the version number to be able to identify it. So, starting with the next major release, we’re going to drop the “1.” completely. The next major release, which was going to be Asterisk 1.10, will now be just “Asterisk 10″ and subsequent major releases will be “Asterisk 11″, “Asterisk 12″, and so forth.

We’ll continue with our plan to have both standard and long-term support releases of Asterisk, and we’ll update the Asterisk Project Wiki with this information as soon as the first Asterisk 10 beta goes out. In fact, this should occur very soon.

As always, thanks to everyone for their continued support of Asterisk. That especially includes the developer community, the people that find and report issues, the people that help test patches and the people that devote their time to answering questions on IRC channels, the mailing lists and the forums.  We hope to see everyone trying out the forthcoming beta, and we look forward to seeing you all at AstriCon 2011!

Related posts:

1. The Evolution of Asterisk (or: How We Arrived at Asterisk 10)
2. Asterisk 1.8, now in Beta
3. Asterisk Music-On-Hold Changes

Asterisk NOW

AsteriskNOW 1.7.1 has been released with a new module for FreePBX that allows installation of Digium add-on software from within the web-based interface. Now there’s no command-line work to be done to get Digium’s G729 codec, Fax for Asterisk, HPEC, or Skype for Asterisk. We’ve also made some changes to make AsteriskNOW even friendlier for newcomers to the Asterisk community.

Together with the DAHDI configuration module that began shipping with 1.7.0, these modules make Asterisk administration even easier.  Download AsteriskNOW today, and burn it to a CD or start it in a virtual machine. In minutes you can turn your computer into your next phone system!

Related posts:

1. AsteriskNOW 1.7 Hits The Streets
2. Skype for Asterisk: Public Beta available
3. Fax For Asterisk 1.1.6 Release Announcement

Digium is pleased to announce the availability of the first beta of Asterisk 1.8. Asterisk 1.8, the next major release of Asterisk, is a Long-Term Support (LTS) release and will be fully supported for 4 years, with an additional year of security-only fixes. For more information about support timelines for Asterisk releases, see the Asterisk versions page:

http://www.asterisk.org/asterisk-versions

A short list of new features that are a part of Asterisk 1.8 includes:

• Secure RTP (SRTP)
• IPv6 Support
• Connected Party Identification Support
• Calendaring Integration for CalDAV, iCal, Exchange or EWS calendars
• A new call logging system, Channel Event Logging (CEL)
• Distributed Device State, including Message Waiting Indicator using Jabber/XMPP PubSub
• Call Completion Supplementary Services (CCSS) Support, including Call Completion on Busy Subscriber (CCBS) and Call Completion on No Response (CCNR)
• Advice of Charge, including AOC-S, AOC-D, and AOC-E Support
• and much, much more!

A full list of new features can be found in the CHANGES file.

How can you help?

By testing, testing and more testing! We need your help finding bugs so that 1.8 is the best release possible. To help, download the beta:

http://www.asterisk.org/downloads/asterisk/releases/asterisk-1.8.0-beta1.tar.gz

Build and install it, and test, test, test!

Report all issues to our issue tracker:

https://issues.asterisk.org

Thank you for your assistance and for your continued support of Asterisk!

Related posts:

1. Asterisk 1.8.0-Beta1 Now Available
2. Asterisk 1.8.0-Beta2 Now Available
3. Asterisk 10.0.0 Beta 1 Now Available

Related posts:

1. Switchvox Business Phone System Wins Top Midmarket Product Award
2. Isn’t everyone looking for more ways to save money in their business these days?

Allison Smith, The Asterisk Voice

The 15 Commandments of IVR

Commandment #3: Keep Things Simple

So far in this Blog Series, we’ve covered the first two “commandments” of writing clean, easy-to-use IVR systems:

• Commandment #1: Don’t Overestimate Your Listener’s Attention Span
• Commandment #2: Thou Shalt Not Create Fake Mailboxes.

Both really key points: many writers of IVR systems feel like they have their callers “captive”, and that now that they have them listening them to their menu of options, now is a great opportunity to sell to them; to educate them in detail, and to reassure them that they’ve made the right decision by calling their company (which can be artificially made to sound bigger and more impressive than it actually may be by creating fake options and forcing callers to listen to the entire selection of options which, essentially, go nowhere.)

The customers who call into your business are busy people. They are probably over-stressed multi-taskers who simply want to accomplish what they need to accomplish in this call and move along. Your job – as the constructor of the telephone systems which “sorts” callers into appropriate departments – is to make their experience in your IVR as simplified and efficient as possible – hence Commandment #3: Keep Things Simple.

If it feels like I’m belaboring the point of simplicity, brevity, and clarity to death, I likely am. As someone who on a daily basis voices systems for a myriad of companies, I can tell you that I always have in mind – while I’m voicing the prompts – how it will feel when someone will call into this system. Will they let out a sigh, dejected, as they realize that in order to get to the department they need to speak with, they will *first* have to endure a commercial, emphasizing the benefits and wonderments of the company they’ve dialed? Will they become overwhelmed and confused by too many options – or options which are so similar as to confuse the decision of which to press? Will their selection be filtered down into too many confusing subsets?

It goes both ways: you will want to ensure that the information you’re asking for from callers is information which will not overload your organization, or make it a challenge to follow through. Just last week, I read a mailbox greeting which instructed the callers: “…For a faster response, please leave your name, number, and brief message explaining why you’re interested in partnering with us, along with your commitment level, your main passion, and the reason why you have decided to enter our industry.” You could really be inviting trouble there; most people wouldn’t likely take advantage of the situation and leave a half-hour long manifesto. But a surprisingly large number will. You need to invite that same clarity, brevity, and economy in a request for incoming information if you have any hope of boiling down the information gathered into a useful form and following through.

Reduce down the choices into the simplest options. Get callers to their needed department as quickly as possible. Don’t ask for information to be input – such as pin or account numbers – if the live agent is just going to ask for the information again. And above all: respect the caller’s time and energy.

Next blog: we’re going to be drilling deeper into the mechanics of sorting your callers into various departments, by giving them the option of not participating in the format you’ve designed at all. That’s right: we’ll be tackling Commandment #4: Always Give Callers an Opt-In.”

Watch for the next blog entry in about two weeks time! Thanks for reading, and your comments are most appreciated!

Related posts:

1. The IVR Clinic — with Allison Smith
2. The IVR Clinic — with Allison Smith
3. Conference Call Etiquette

Guest Blogger - Tim Kruse is the VP of Sales and Business Development for Incendonet, a Digium software partner.  He has been involved with speech recognition and advanced communication solutions for more than 15 years and will be presenting during an upcoming webinar on July 13th at 12PM CST.  Register Now

I have been working with speech recognition technology since 1999 and appreciate how far the technology has come.  As with many technologies we come across, the more we all use them the better they seem to work.  We all become trained by our daily interactions.  If technology is useful, affordable and easy to implement than eventually most of us will adopt it.

For many Small and Medium sized Businesses (SMBs), the idea of implementing advanced speech recognition and text-to-speech solutions hasn’t really occurred. Most aren’t aware that speech recognition based offerings created with SMBs in mind even exist. Even in today’s Internet age with websites, social networking, chat, email, and text messaging, our business phone system is still the focal point of interactions with our customers, and speech recognition has been shown to be the preferred method for interacting with self-service applications. SMBs need to do more with less, better serve their customers, project a professional image to the outside world and allow their employees to be more productive in today’s mobile world. Here are some of the ways my company has helped SMBs. use speech recognition based solutions.

• A small school district was inundated with calls first thing in the morning and when school let out in the afternoon.   A significant portion of the calls were simple call transfer requests.  The school district implemented our speech-attendant to off load call transfers and freed up help desk employees to better serve parents with more complex inquiries.

• A construction company needed our help so their field workers collaborate by calling one main number and simply stating the person or the department they needed to reach.  These workers were able to collaborate as necessary with safe hands-free and eyes-free access to each other as well as stay connected to their corporate e-mail and calendars using simple, spoken commands.

• A winery located in California, where hands-free legislation exists for mobile phone users when driving, wanted to better serve their customers by replacing their dial-by-name directory with our speech-attendant.   To improve collaboration for their field workers they also created a custom sub-directory of employee’s mobile phone numbers.

With Digium’s support of standards like SIP, companies of all sizes have the power and flexibility to easily add speech recognition based solutions to their VoIP phone systems.  Specialized VoIP appliances such as Switchvox and SpeechBridge enable SMBs to implement advanced communications solutions previously reserved for only large enterprises and call centers.

Tim Kruse, VP of Sales and Business Development for Incendonet, will be the featured speaker at the next Asterisk In-Depth Webinar.  The webinar will take place on Tuesday, July 13th at 12PMCST.  Register Now.

Related posts:

1. Introducing our new Switchvox Webinar Series
2. Conference Call Etiquette
3. The IVR Clinic — with Allison Smith

Digium’s Training Department has been working on some exciting projects, and more work remains to be done. If you know Asterisk, have an interest in training, and want to work for Digium, we’d love to hear from you! Contact us at training@digium.com. And stay tuned to digium.com/training to hear what we’re up to…

Related posts:

1. The IVR Clinic with Allison Smith
2. Digium Releases TCE400B PCI-Express Transcoder Card
3. The IVR Clinic — with Allison Smith

Related posts:

1. Isn’t everyone looking for more ways to save money in their business these days?
2. AstriCon 2010: Back to Basics
3. The Evolution of Asterisk (or: How We Arrived at Asterisk 10)

Allison Smith, The Asterisk Voice

About a year ago, I recorded an IVR for a small independent dry cleaning business – not really a Mom and Pop company; they were located in outlets across three States, and were doing very well – they prided themselves on fitting in seamlessly into the communities they served and they were at just the right size for their comfort level. When I recorded their system, a request for a total re-record came in (never a good thing) but their reason for the redo was unique and sticks with me to this day: my usual professional tone was seen as too “highbrow” for them. Created too much of a “big company” impression. They didn’t want to be “Martinizing”; they wanted to sound “local”….friendly….and accessible.

This is in sharp contrast with how *most* companies I voice for would like to come across – I would estimate that 80% of the companies who hire me to voice their systems are small and would like to sound bigger. Almost all firms have their eye on growth; the best way to do that is to create the impression that they’re already there.
A common technique to “manufacture” the impression that a company is bigger than it really is, is to invent a lengthy menu of mailboxes which technically don’t exist – an impressive, vast menu which goes on for 12, 13, 14 options or more – all in an attempt to articulate to the caller that they are legitimate; the caller has reached a well-staffed company who needs *that many* mailboxes to keep all requests organized and processed appropriately.

Many (or all) of the mailboxes will re-route to a single point of contact, but as pointed out by Matt Florell of Vicidial (who, along with Jim Van Meggelen, acts as my “IVR Senseis” for this series of articles – their input has already been invaluable) – it’s easy for the person in charge of monitoring the various mailboxes to overlook one or two of them for a couple of weeks, and “then they end up with 300 voicemails and only notice it when Asterisk hits its limit,” warns Matt. “Sending voicemails to an email address and auto-deleting from Asterisk does help with this,” continues Matt: “..but the flip side to that is that your company’s SPAM filter starts to think these messages are SPAM and deletes them.”

I submit to all IVR designers the importance of keeping the opening menu as simple as possible to navigate around – and this means to only feature the mailboxes which are actually assigned. It respects the caller’s time; it streamlines the system, and it prevents missed messages and botched follow-through. The idea that a more impressive feeling is created with a menu full of unnecessarily bloated options is counter what you’ve possibly encountered in your own telephone experiences – personally, I’m usually grateful for three or four simple options, narrowing down the likelihood in my mind that I have chosen the correct department for my inquiry.

Next installment in The IVR Clinic’s “15 Commandments of IVR” is #3: Keep Things Simple.
Thanks for reading!