Any Word from Astricon?

[kenn10]

I was curious if anything breathtaking has been announced at Astricon. I had hoped some new and cool features of Asterisk 1.8 or 2.0 would be revealed.

I'd like to hear some feedback from the convention if anyone has any.

 

[Linetux]

Actually, yes.

There's a new feature - actually a whole new project - announced at the keynote this AM. It even came with a live demo - more on that in a minute.

So the new announcement is for Asterisk Scalable Communications Framework. Basically it's a way to handle redundancy and scaling of Asterisk into areas that have been very difficult to do so far given the current code base. Essentially the idea is that you setup a cluster of servers that participate in the new SCF framework, and your Asterisk boxes participate in it as well. It then allows for full redundancy from the call all the way down to presence.

The live demo was with the CEO & CTO of Digium, along with 10 volunteers from the audience. They had the 10 folks call into a conference bridge, and then the Digium employees joined in as well. In this setup on stage, they had 3 "data centers" - one where the conference bridge was, and another two for each the CEO & CTO. Then they shut power down to one of the data centers, but everyone stayed on the call. In fact, they were even able to transfer one of the calls over to another data center after its data center had died.

VERY cool stuff. This is what I look forward to - not some half-assed implementation of Skype like they did last year.

Of course, none of this has been released (yet) and I'm sure it's still a couple of years away from real use, but it's really good to see Asterisk growing up.

More than likely this will become an integrated part into what Asterisk 2.0 might be. They did announce Asterisk 1.10, but I haven't seen details there... there's also a Q&A session this afternoon about the new SCF and how it will integrate with other things.

 

[wardmundy]

Coming Nov. 1 to Incredible PBX

Once folks see Google Voice calling via Gtalk using Asterisk 1.8, you'll drop Asterisk 1.4 and 1.6 like a hot potato. This isn't the Rube Goldberg edition by joining two calls via an intermediate provider. Jabber and Jingle are functionally equivalent to a direct SIP to SIP call. We think the instantaneous connections and voice quality will make a believer out of just about everybody. Did we mention all the calls in the U.S. and Canada are still free including an inbound DID in your favorite city. Tom and I have been working 12 hour days to make this happen sooner rather than later, and Asterisk 1.8.0 appears to be rock-solid! So we missed Astricon.

But... I agree. Asterisk is making impressive progress. You can't help but think some of it is inspired by looking in the rear view mirror at FreeSwitch. But that's a good thing for all of us.

 

[Linetux]

New SCF is taking advantage of ICE, or the Internet Communications Engine. This is another Open Source project that allows for fast and efficient communication over IP. This is the main core on which they built, and it's what they use to keep everything in sync in order to provide realtime information about calls - and more importantly not drop calls

Now, if only the carrier would talk to a SCF cluster so you'd never have a failed call unless the endpoint died!

 

[kenn10]

Now if they would just address the need for shared call appearances, Asterisk would be able to knock Cisco and Avaya out of the way.

 

[Linetux]

What do you mean by that?

 

[tm1000]

What do you mean by that?

I think Kenn10 means it's doesnt work like it's suppose too.

 

[Linetux]

What I'm asking is what is a "shared call appearance"?

 

[hkgonra]

I hate to sound too consipracy minded but it seems like if they wanted to fix shared line appearance they could. Maybe someone doesn't want them to fix it ?

 

[malcolmd]

I'll quell any suspicion. There's no conspiracy. There's not an evil agent who doesn't want us to fix it.

As far as Digium has been concerned, Broadsoft-style SCA and Sylantro BLA extensions just haven't been at the top of our priority list, internally, for things to do.

If you and several community members feel strongly about BLA/SCA, we'd love for you to organize it into a plan of action that results in code.

We're not against new features and improvements, we're simply not staffed to address everything we'd like to, so we have to choose the projects we tackle.

P.S. to Ward: I'm looking to the future, and I see happy PIAF users on November 1st.

 

[wardmundy]

What I'm asking is what is a "shared call appearance"?

It's blinky lights on every phone so that you can pick up a call that someone else has already answered. In the "old days" the (almost always) female secretary put the call on hold (hence blinky light), announced the call to her (almost always) male boss, who then attempted to use a phone by picking up the handset and pressing the blinky light. Great inventions die hard.

[youtube]k9e3dTOJi0o[/youtube]

 

[kenn10]

Background on Shared Call Appearances

Shared call appearances are typically used in boss/secretary or shared office spaces where several extensions appear on a pool of phones. It gives the ability for one person to answer a call, put it on hold, and allow for another user to pick up the line.

Note that this is not shared trunk appearances but actual extensions that appear on several phones and provide lighted buttons to indicate busy, hold or ringing status.

I've been in telecom for 30 years and believe me, every major installation of enterprise class phone systems supports it and the users of those systems want it.

Along with shared call appearances come a bevy of issues that must be addressed such as exclusion (ability to keep other phones from picking up on your active call unless you want them to), hold indication so other phones know when the extension is on hold and the buttons blink accordingly, ringing notification so other phones indicate ringing lines, audible ring selection to determine which lines audibly ring on which phone. A lot of work to program I'm sure.

This is what distinguishes the big boys from Asterisk. All I can say is that since the majority of Asterisk installs seem to be smaller deployments, I think to make the next leap to Fortune 500 or Fortune 100, some changes must occur. Even Cisco added shared call appearances after their early system deployments.

The fact that so many people on Asterisk and other forums have no idea what shared call appearances are is indicative of a lack of understanding of what potential customers expect. Lack of experience in the marketplace. Missed expectations. Missed opportunities. Perhaps ignorance is bliss. Freeswitch has addressed shared extensions and many other enterprise class features but lacks the ease of implementation and administration that we enjoy with Asterisk.

The folks at Digium and the Asterisk community at large have done so much to make Asterisk a "real world" product and I respect just how far things have come in a few short years. Large enterprise and most government (local, State and Fed) expect more than what the current product offers. It sounds like true redundancy and voice encryption are here (almost).

I see Asterisk being only a few steps from a drop-in replacement to Avaya, Cisco, Nortel or Siemens if the few feature omissions are addressed.

 

[kenn10]

I'm Willing To Spearhead Some Discussions on SCA

I'm not a programmer but I can specify functionality if someone has interest in tackling Shared Call Appearance in Asterisk.

I'll toss this out as known parts of the puzzle:

1) Because SIP is not really intended for multi-line functionality, you must have an intermediate process or proxy between the Asterisk extension registration and the telephone. This agent front-ends requests between Asterisk and the phones.

2) The intermediate agent must allow multiple phones to register to it, rather than to Asterisk. Phones would have unique registrations to the proxy and the proxy would handle individual call appearances registered on the phones.

3) Study needs to be done to see if any multiline SIP phone could work in this capacity or if the Broadsoft method would be utilized (thus limiting which phones could use it).

Here is an example of how I see it working from an average Sip phone:

a) Sip phone registers to proxy as (for example) SIP1
b) Line-1 of the phone registers at SIP1-1, Line 2 as SIP1-2, etc.
c) The Sip proxy is then told that SIP1-1 is Asterisk extension 201 and SIP1-2 is extension 202, etc.
d) Another phone could register as SIP2 with lines of SIP2-1, SIP2-2, etc.

As many buttons of 201 as you want to register to the proxy would look like one extension 201 to Asterisk. This would almost be a mini-Asterisk within an Asterisk. Several of the big boy systems use a separate applications processor to do this so the proxy process would need to be scalable from running on the same box with Asterisk to the ability to move to other boxes as required.

Baby steps first, I'd do a process on the same Asterisk box and branch out from there. The design needs to work from small to large and on the same box to multiple boxes.

Any code developers for Asterisk want to pick up on this and run with it? If this takes, off, we need to move this to a new thread since I have hijacked my original post.

 

[islandtech]

The majority of my small to midsize companies prefer and demand Shared Line (trunk) appearances and thus they won't move to Asterisk.

 

[wardmundy]

I've been in telecom for 30 years and believe me, every major installation of enterprise class phone systems supports it and the users of those systems want it.

Amen, Brother. All I can say is that, if you are ever responsible for swapping out a phone system that had it with a new phone system that doesn't, you'll probably be standing in the unemployment line shortly thereafter.

"Users want it" doesn't begin to describe the firestorm you'll have on your hands. Ask me how I know.

 

[TheShniz]

Alrighty, I like your mini-Asterisk idea, and I've got a good & fresh writeup to maybe get your hampster wheel moving (but first some description)... the Asterisk engine (after many years) still remains incapable of performing some very basic tasks. These tasks are common expectations of customers both SMALL and large (small = key-system, i.e. Shared Line Appearances - SLA; large = corporate, i.e. Shared Call Appearances - SCA). Freeswitch (very roughly speaking) is the culmination of people that are tired of being unable to provide such basic abilities which Asterisk lacks, and these folks have the ability to effectively rewrite their own engine. Unfortunately, Freeswitch is still in its infancy, and has absolutely nothing written for it (with any level of sophistication) in terms of features and interface (remember, it wasn't till the Asterisk Management Portal aka AMP was written that Asterisk really took off).

SOOOooooooooooooo....... what IF you could run both Asterisk AND Freeswitch on the SAME system, get them to play nice with each other, and do everything in Freeswitch that Asterisk is incapable, while still retaining the interface and application development for Asterisk?

This article was written, so Asterisk could actually do Google Voice the right way, through a direct connection... http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to_add_google.html ... but I see no reason why this concept couldn't also be applied to SLA, SCA, or any other three-letter acronym that you can't search for on these forums.

Yeah, after all these years w/ Asterisk, I have a real hard time believing it's a prioritization issue of 'things to do'; and yet, it continues to be the canned response (whether an honest one or not, it's hard to accept with so much time having passed). I don't know if it's Asterisk growing up faster than is reasonably expected, if the engine needs to go over some rewrite (much like FreePBX is) to clean up a messy backend, or what. I also tend to agree that seeing Freeswitch in the rear-view mirror may be just what Asterisk needed. I for one am looking forward to 1.8 and some of its new-found goodness (so long as there's no more syntax-craziness).

 

[lowno]

With SLA I see it more as a give an take. There are so many benefits with asterisk when implemented right.

 

[malcolmd]

Yeah, after all these years w/ Asterisk, I have a real hard time believing it's a prioritization issue of 'things to do'; and yet, it continues to be the canned response (whether an honest one or not, it's hard to accept with so much time having passed). I don't know if it's Asterisk growing up faster than is reasonably expected, if the engine needs to go over some rewrite (much like FreePBX is) to clean up a messy backend, or what. I also tend to agree that seeing Freeswitch in the rear-view mirror may be just what Asterisk needed. I for one am looking forward to 1.8 and some of its new-found goodness (so long as there's no more syntax-craziness).

We're open for contributions.

Cheers.

 

[kenn10]

Sounds Like A Cold Shoulder

I guess from the remark, the Asterisk developers are still working on things they consider more important than shared call appearances. I'm not sure how far we can get trying to do community development at the speed things are changing with Asterisk.

Its a shame because I believe this is a vital feature to compete in the real world of big business. Having to install a Panasonic key system or Avaya Partner system to provide this service to certain areas of a business precludes the feasibility of using Asterisk.

 

[malcolmd]

I guess from the remark, the Asterisk developers are still working on things they consider more important than shared call appearances. I'm not sure how far we can get trying to do community development at the speed things are changing with Asterisk.

Its a shame because I believe this is a vital feature to compete in the real world of big business. Having to install a Panasonic key system or Avaya Partner system to provide this service to certain areas of a business precludes the feasibility of using Asterisk.

Howdy,

It's not the cold shoulder. My response was and is, also to you, the most reasonable one I can provide.

The things that were deemed most important following the most recent AstriDevCon are documented here:

https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010

The AstriDevCon is an open event held before or after AstriCon each year, so, PIAF developers are welcome to attend and to voice their concerns and ideas. Certainly, we at Digium can do a better job of publicizing that moving forward, as word probably didn't escape past the asterisk-dev mailing list this year.

The Asterisk development community would be thrilled for someone to put forward a workable solution using the dialog-info method.

We're wary of using the Broadsoft extensions, because those extensions are not provided without license from Broadsoft, so to use them, we'd have to be 100% certain that someone either received them under a license that Broadsoft doesn't use, meaning it ain't gonna happen, or that they reverse engineered the extensions in a clean manner (slightly less unlikely). We have to be good stewards of Asterisk.

Most phones support dialog-info these days.

Based upon the discussions at the recent AstriDevCon, doing work on any SCA/BLA features was put in the category of items that no one's really sure we want to invest our time in.

From Digium's perspective, we've got a series of other items we're already signed up for, and I fully expect it'll consume us completely between now and 1.10. The items we're signing up for are items that no one else is going to be able to develop, due to complexity.

Based on the other developers in attendance, no one was really sure they wanted to do it either. No one, for example, wanted to attach their name to getting it done.

So, this is the point where someone else out there among the user and developer-space of Asterisk steps up to the challenge.

Maybe the PIAF community gets together, finds a developer, and sponsors the feature?

Cheers