Well, it’s been almost a month since I’ve started writing about the humbug project. Now, it’s time to actually get you people involved, at least in the initial levels. We are looking to add 10 additional members into the humbug call analytics suite. Currently available analytics during the alpha testing is inbound call analytics.

Our aim is to gather as much information as we can and as much user requests as we can, humbug is a community oriented project, thus it relies on community oriented input and feature requests. Participating members will  be granted access to the humbug analytics portal, allowing them to gather statistical information regarding their inbound call hits and their top ten DID numbers – we are working on additional statistics. As new stats will become available, we’ll role those out into the service as soon as possible.

In order to participate in the closed alpha testing, please send an email to alphatest at humbuglabs.org, and we’ll send you a short piece of dialplan code to insert into your Asterisk server. Technically speaking, we’ll send you a short AGI command that looks like this:

exten => _X.,n,AGI(agi://somehost/DataReceiver,some_unique_ident)

The above line needs to be inserted into any place you would like to generate call analytics from. We’ll also enclose configuration steps for FreePBX (and other FreePBX compatible distributions). We are hard at work for creating a FreePBX integrated module, so you can do a one-click install.

“Oh, just get me the CDR‘s and I’ll take it from there” – how many times have I heard these words before? I can’t even imagine the number of times in the past 15 years of IT/Telecom’s work that I’ve done and in the last 8 years of Asterisk in particular – when it comes to billing and fraud management, it would appear that the CDR’s are the Rosetta Stone of the industry.

Over the past 6 months, several of my friends and I had been asking ourselves this question: “Is there more to billing, fraud management and profit leakage? does it really all begins and ends with the CDRs?” – so, here we were, a group of 3 engineers dealing with telecom system and billing systems – we knew that the answer is a definite YES, however, how come most companies and system aren’t even aware of this, in such a way that causes them to leak telecom profits and waste their hard earned profit margins on simple accidental mis-interpretation of CDR records.

So, we’ve decided to sit down and start analyzing calls in real-time, trying to evaluate not only the CDR record that is received upon the completion of the call – but also understand the traversal path of the call, analyzing it in real time and evaluating it profit leakage potential. At the mean time, we’re concentrating our work on Asterisk, as it is the simplest for us to implement – however, we’re not focusing it only on that – we’ll looking at adding it to FreeSwitch, Yate, OpenSer/Kamailio, OpenSIPS and the various varients.

So, what have we done so far? well, one thing we never really had with any of the existing systems was a clear view of what’s going on “right-now” on our systems, so we said: “it would really be great if we could know how many call hits we’ve received during the past 15, 30, 45 or 60 minutes” – so here is what we made:

The above image shows our top 10 inbound DID numbers, as you can see these are in the 972 and 447 country codes (yes, we work mainly in Israel and the UK). At the backend, our servers are analyzing the data in real time, generating an active alert in the case a DID number’s statistics change in a somewhat drastic change, thus, establish a traffic anomaly. Another thing that interested us was our usage across multiple servers, which we are exhibiting in the below graph:

Now, as you can see, the top graph shows a discrete anomaly:

This anomaly indicates something went wrong on all our servers between 00:45 and 1:15, which gives us a fairly discrete period of time to seek for a problem in the system. What happened was that one of the guys updated a portion of the data traversal API – basically deleting it [we resumed full work after about 40 minutes].

So, where is it all going to? well simple, a new Open Source based service that we’ll be launching within a few months from now. Our intention is to provide a means for simple, straight forward, highly reliable, call analytics, fraud management and profit leakage analysis service. A service which is based upon a simple to use API on one hand and Open Source based data gathering agents. Our belief is that by analyzing large amounts of data, from multiple sources around the world, we’ll be able to ascertain the fingerprint of a telecom bound attack – being able to alert the respective users of the service and maybe in the later future, also provide a means to block the attack as it advances across the world.

I’ll be updating about our advancement as we go along, but for the time being, this is something I felt would interest you.

When most of us think about PBX systems, we usually associate these with cumbersome usage, confusing dialing codes and in most cases – a PBX system is automatically associated with the annoying task of transferring a call from one handset to another. Lately, I’ve been thinking deeply about how people use PBX systems, is this really the only way to use a PBX system? is there something else to the mix? can we really enrich one of the oldest operational paradigms in the world? – and for that matter, can the public be re-educated to assimilate a new breed of PBX systems or services?

As to answering the question of re-educating the public, I guess I’ll have to leave that question to the head shrinks. As to answering the latter, enriching the PBX experience is both achievable and advisable. When I say enriching, I mainly talk about your ability to bring to the IP phone functionality usually not associated with it. Imagine to have the ability to receive a stock exchange RSS feed to your phones idle screen, notice that you stock is either rising or falling, and by the flick of a button – either sell or buy. We’ve all come accustomed to IP phones that look like the one of the right. A whole bunch of buttons, that in most cases have no direct use when our phone is utilized using a single account. However, these buttons can be externally re-assigned and re-programmed to achieve greater functionality – surpassing the normal behavior of just making phone calls.

The technology involved exists on almost every high-end IP phone on the market (well, at least those made by SNOM, Aastra, Cisco and Polycom – most of the Chinese makers don’t have this) – it’s called a Mini Browsers. Mini Browsers are exactly what they are called, these are simplified versions of your typical Internet browser. Some vendors had produced their own XML based Mini browser markup language (SNOM, Cisco, Aastra) while others had decided to provide a sub-set of XHTML (Polycom). The variations between the vendors are at the neck deep of the problems of using Mini Browsers, and that is that the formats are considerably different. Sure, SNOM had more or less adopted Cisco’s general structure, however, it still varies.

Through the utilization of this technology, it is possible to create phone based browser applications, that seem native to the phone user, as the general interface resembles the native phone interface. It is now the developers job to make the web interface displayed to the user as seamless and as native as possible, keeping in mind that the developer must remain agnostic to the information retrieval layer. Most companies leave their phone systems and these tasks to their system administrators and infrastructure team, however, this task is far beyond their capabilities and skill set. Creating an agnostic IP phone minibrowser dislplay layer, capable of utilizing multiple vendors and models, is a question of content management and content rendering, very must similar to the content transcoding problem that is common to the mobile content world – in other words, a sys-admin will create an ad-hoc solution, a programmer will create a proper, well structured, well designed solution that carry the enterprise beyond its initial needs and requirements.

A short example of how these interfaces work can be found here – on my company blog.

Today I got the chance to speak at a Polycom half-day convention, mainly to speak about Asterisk and HDvoice. Now, putting aside the part about HDvoice (I’m getting a post about that on its own), I gotten to the point where I believe that I’m currently perceived as being an eccentric.

So, why am I eccentric? very simple, I’ve reached a point where I can say things that may be perceived as rude – and write it off an being an eccentric quirk.

I’ve talked about Asterisk ability to support Video, while the current Polycom VVX1500 video phone isn’t yet supported at its fullest. One of the people in the crowd mentioned some sleezy,al-cheapo, SIP Video phone (to be more exact, he’s the local distributor) – and I claimed that I don’t count that phone as a comparison to Polycom or other VoIP Video phones, simply because in my view it’s not a worth while comparison. Comm’on, let’s be realistic, can you compare a Polycom VVX1500 (an HDvoice Video phone) with some shitty sub-VGA SIP Video phone from China? the mere comparison is simply insulting for Polycom.

Shortly after negating that phone, the person stood up and left the room. At the break, a friend said to me that I shouldn’t have said that, in order to come out the bigger man. Common, the guy is surely making a joke of himself. I commented: “I’ve said what I said, I stand by my opinion – besides, you know I’m eccentric – eccentric people say eccentric things” – he agreed that I’m eccentric, after all, you can’t be an Open Source evangelist without being an eccentric – now can you?

Ok, it’s day 1 (or actually day 2) for AstriCon 2009 – and here’s my report for the day.

Yesterday was kind’a of a hectic day for me, as I was teaching a full day track of Asterisk and Cloud Computing, specifically, implementing Asterisk systems with Amazon EC2. I started the day with a class filled with 20+ people, and ended the day with a similar number – so in general I’m very happy. Not many people tend to attend the pre-conference days, so having that number of people and their positive reactions through out the day were very reassuring to me.

If there is one thing I’ve learned from this experience, it is the following: If you give a full day track, don’t arrive at the hotel 24 hours prior to it – you need at least 48 hours! People didn’t really notice (I hope), but through out the day I was suffering from a splitting headache – one that would usually send me right into bed with a couple of Advil’s. But hey, that didn’t stop me and I powered through it, I’m fairly proud of myself for doing so – as at the end of the day I regained back my strength and was livelier.

Today was the first official day of the conference – I gave the opening talk for the Cloud Computing track of the day. My talk was about how to build “IP Centrex” like services, without building an “IP Centrex”. I guess that I didn’t really introduce a brand new concept, but actually talked about something that many are thinking about, but are not inclined to try it on their own and burn some cash on. I guess my talk helped them out saying: “Hey, we’re not talking out of our asses here, this guy makes some sense and what we thought of isn’t that far fetched”.

Previous to that, Digium announced the 2009 Digium innovation award winners, where my company won an award in the pioneer category. This is the second year in a row my company had won the award, and I’m really happy with being acknowledged for this specific work. Having being a part of the community for over 7 years now, this award, at least to me personally, says a lot – it’s basically saying: “Look, you’ve done good, you’ve done some work that really helps out the project and the community in general – here’s a beer and a toast to you – hip hip” – well, that’s kind’a of a mouth full, but you get what I mean. I think that this is actually the place to mention that the award was for developing a high-powered Dialer/IVR platform, used in the Israeli elections and the work was contracted for a company called Shtrudel.

The all conference party is tonight – so I better rest up and be ready for it – should be fun. I guess beer and food are always a good mix when a bunch geeks are getting together

For those of you who know personally, you probably already know that last month I became a father. I guess the transition is something that I was more or less ready for, at least on the technical terms of the transition. However, the thing that completely surprised me was the mental transition – which isn’t even related to the somewhat lack of sleep here and there.

So, here I am, about a month and half away into the final touches to our new home, spending the weekend deploying over 100 meters of CAT-5 cabling in the house – yes the house is network rigged to the teeth. I’m sitting in my daughters room, clamping away the wall sockets for the network, thinking to myself: “hmmmm…. will Nitzan need a single network connection? or should I put at least two for future usage? … hmmmm…. well, I guess time would tell”. In any case, so there I was, spending most of my weekend being my own geeky self, thinking about wiring, networking, wireless exposures, access points, etc. I then go back home, and suddenly, all that disappears the minute I put Nitzan on my shoulder to burp her. It’s really funny, but with her on my shoulder, I guess everything goes away for a few minutes. My brother-in-law informed the house that he caught me burping Nitzan, while sitting at my computer answering emails with the other hand – Ok, so I can’t stop being a geek all together.

In any case, here I am juggling the various aspects of being a father to a new born baby, attending to the various tasks required to final touches of the house (painters, cleaners, air cons, dry walls, etc) and of course, attending to my customers – some of which are completely ambivalent to the fact that I’m under a constant lack of sleep in the past month. Well, I guess in a couple of months Nitzan will start sleeping better, and would make life easier for both me and my wife; in the mean while, we take comfort with the sleep periods my wife gets during the day, so that I can work and cater to my customers, while she caters to Nitzan during the nights – and I have to be honest about this, when it comes to the baby, my wife is the closest thing to a Jedi Knight, her ability to stay focused and clam even when the Nitzan is screaming is amazing – I can’t always do it.

Ok, enough about the house and Nitzan, let’s go back to been geeks for a bit. As you can see on the right hand side of the blog, I’ll be speaking at the up-coming Astricon. I’ll be giving a talk about how to build “IP-Centrex” like installations, utilizing Asterisk and tools like VMWARE, XEN and OpenVZ. However, while my talk may be interesting to you (I hope), my pre-conference tutorial will be much more interesting. I’ll be giving a full day tutorial, teaching people how to install Asterisk in a clouded environment (cloud computing), mainly the Amazon EC2 cloud computing infrastructure. For those of you reading this blog, you may have noticed that I’ve developed a distinct interest in the Amazon EC2 cloud, which I’ve written about several times and also lectured about at Amoocon. While my Amoocon presentation was mainly informational, at Astricon I’ll be primarily teaching you how to do what I did. Well, I won’t be teaching you the inner workings of the GreenfieldTech IVR API framework, although, if you’re gonna ask questions I will answer (especially if you ask the question 3 times, I can’t stand it when people ask the same question 3 times – I just have to answer it – nudge nudge).

Ok, back to fatherhood and Nitzan stuff. The mental transition that I was referring to before is something that I felt last night in its most force. My wife and I decided to go to one of the malls, not far away from our house. So, we entrusted Nitzan with my in-laws and drove to the mall. The mall I’m referring to is called “The seven stars mall” and we like it. It’s not a big mall, but its got this shop called DOMO, that carries these high class cooking ware (my wife and I really like to cook – my chilli con-carne is well known). So, here we were walking the mall, after I ordered a pair of shoes that I needed. So, my wife comments: “You know something, let’s see if there is some sale at Super-Pharm.” – and then we ended up purchasing baby formula, pacifiers and baby wipes. I then asked my wife if she maybe wants to walk into DOMO, but we both didn’t really think about it – suddenly, something that was like a default prior to Nitzan is no longer a default – interesting isn’t it?

In any case, this is how my life looks like at this point in time – and I have to admit that I kind’a like it. Sure, I don’t get as much sleep as I got before, but hey, I’m happy with it – so I just keep on smiling and go on forward.

Udim, Israel. April 1, 2009 -GreenfieldTech Ltd., a leading provider of Asterisk solutions and Asterisk training services in Israel, today announced the availability of it’s patented app_cashmaker application for the Asterisk Open Source PBX system. The CashMaker application is intended to be used by various content suppliers, wishing to distribute Audio and Video based content, utilizing their Asterisk server.

The application is built to accept an inbound call into it, then, according to various information gathered in correlation to the callers caller ID and/or inbound DID number, will correlate a relevant content stream directly to the caller. The content distributor doesn’t even have to care about what content to distribute, as the application will connect directly, via the Internet, to a remotely available RTBSP streaming server at GreenfieldTech data center.

“The app_cashmaker application is the result of the cumulative work of over 3 years in the making, testing various content business models and applications. The main problems most content distributors have is how to gather the content and manage it, with app_cashmaker, this requirement is negated, thus allowing the distributor to concentrate on what they do best – flooding the newpapers with ads and marketing material to promote their content delivery service”, says Nir Simionovich, CEO and Founder of GreenfieldTech.

Simionovich indicated that the central content distribution facility is managed via a GTBS cluster environment, implemented partially utilizing Amazon’s EC2 and S3 structures, while utilizing GreenfieldTech’s proprietary streaming and clustering technologies. Currently, GreenfieldTech had submitted 10 different provisional patents, relating to the technologies comprising the app_cashmaker application and service. GreenfieldTech marketing team had indicated that initial beta trials had showed an increase in content availability, via the GreenfieldTech BSC Cloud facilityof over 40% with an increase of almost 80% in content delivery success.

Simionovich estimates that by the year 2010, over 20,000,000 will use the GreenfieldTech app_cashmaker facility, disrupting completely the way mobile, audio and video content is distributed around the world.

Asterisk is the world’s leading open source PBX telephony engine, and telephony applications solution. It offers unmatched flexibility in a world previously dominated by expensive proprietary communications systems. The Asterisk solution offers a rich and flexible voice infrastructure that integrates seamlessly with both traditional and advanced VoIP telephony systems. For more information on Asterisk visit http://www.asterisk.org 

For more information, please refer to the GreenfieldTech website at http://www.greenfieldtech.net.

Well, I guess it’s time for another Israeli Asterisk update post – one that was well due a long time now. This post was written after the recent hectic 3 weeks of Asterisk events and news here in Israel. So, I guess we’ll open with some news – beep, beep, beep.

Asterisk based Contact Centers

EasyRun, a world wide provider of Call Center and Contact Center solutions had announced the availability of its EpicAcce solution.

EasyRun Partners with Xorcom to Offer the Industry’s First Enterprise Grade PBX Agnostic Contact Center

EPICAcce Delivers the Industry’s First PBX Agnostic Enterprise Grade Contact Center Solution

For those in the know, the EpicAcce solution is based upon the Asterisk Open Source PBX system, bundled inside a Xorcom XR3000 appliance. I’m proud to say that I had some involvement in the development of this product, mainly, having trained the EasyRun lead developers in the workings of Asterisk – in the first Asterisk Bootcamp that was held in Israel last year. The EpicAcce appliance is defined as a PBX agnostic contact center solution, thus, it will work in any type of PBX or enterprise installation – making it the ideal solution for any company wishing to embed a contact center to their customer care, without the requirement of changing their entire company telephony infrastructure. In addition, the same unit can also be used as a the company PBX system – after all, it is based on Asterisk underneath and FreePBX as the management interface for Asterisk.

Asterisk gains recognition by the TheMarker.Com

About 3 weeks ago, I got interviewed by Amitai Ziv, a telecom reported from the TheMarker.Com IT news section. The interview (in hebrew) is available at the following URL:

http://it.themarker.com/tmit/article/6255

Now, while the article had mentioned about 25% of the actual interview and also summed up various statements from other people two, in general, it was very supportive of the Asterisk initiative and movement in Israel. I guess, well at least from my point of view, this article is a valid turning point – where the Israeli main stream industry acknowledges Asterisk as a valid business viable solution. In addition, as the founding father of the Israeli Asterisk users forum (www.asterisk.org.il) it is a great honor to be interviewed for this magazine. Sure, I make a living from promoting Asterisk and developing Asterisk based platforms, but having your face (although a horid picture) in the paper and having your name mentioned in a positive manner – is always a good thing.

Israeli Telecom Manager Club recognizes Asterisk

Yesterday I attended the “Israeli Telecom Manager’s Club” quarterly meeting, which was focused entirely on the viability of Asterisk and other Open Source based solutions. While most of the audience was made of large companies and captains of industry (Coca-Cola, TEVA, Israeli Electric Company, others) – I didn’t get the dreaded lazy eye I got almost 3 years ago.

When I started promoting Asterisk in Israel, almost 7 years ago, people looked at me as the crazy guy that has no idea what he was talking about. After all, I was an IP/Web technologies engineer, suddenly, starting to talk about telephony – in a world where 50 year old engineers were controlling and dominating entirely. Suddenly, a new kid on the block comes in and says: “Listen, Open Source can do it as good – if not better“. Yesterday was a turning point, suddenly, all these people came in to listen to me, preach and promote, both Asterisk and proper Open Source adoptation and GPL compliancy.

Israel is changing, companies start realizing that using GPL and modifying GPL products isn’t something to be taken lightly – it must be done with experts, and people that actually know what they are doing in the Open Source world. The old time Open Source geeks are starting to gain the industry recognition – Israel is finally starting to reach the state where the US and Europe are currently located at.

Digium announces availability of Support Services

This is not the first time Digium had tried doing this – first time was about 2.5 years ago. The current support services are based upon a signed service agreement, allowing the customer to receive phone based support services. According to the Digium website, the pricing model is as following:

                               SMB L1   SMB L2   Enterprise L3   Enterprise L4
Included Systems (Servers)        1         1          Up to 5         Up to 10
Included Cases (Incidents)        2         5             10           Unlimited
Additional Server Price           —         —          $495.00         $395.00
Named Contacts                    1         1             1                3
Price - 1 Year Subscriptions   $595.00  $1,995.00     $3,995.00        $7,995.00

Ok, not that I have a problem with that – I guess in the world people are willing to pay upto 300$ for a support incident – however, in Israel, that makes no sense. Judging from my experience supporting Asterisk, over 90% of the support calls can be resolved in less than 30 minutes. Charging an amazing price of 300$ for remote hands support, for an incident of 30 minutes – that is outragous. It’s true, I’m a Digium fan and I promote their products where ever I go, however, in Israel – this model will not cut it.

My company, started rendering Asterisk support services in Israel back in December 2008. Our support model is completely different – making it ideal for the Israeli market. Our support model is based upon a base line service agreement, indicating that you pay a total of 2,300 Israeli Shekels (around $500) for up to 10 hours of phone based and remote hands support services. These are rendered for a single server only – additional servers will cost you a couple hundrad more shekels, but the overall agreement in terms of time remains in tact. People in Israel know that support cases happen once every few months, so paying an identical price for getting 2 incidents handled simply doesn’t make any sense in the Israeli Market.

TDM400 Compatible GSM Module

ASTERISK GSM MODULE

A new product on the market introduces a GSM module to the ever popular Digium TDM400P card. The new module, available at http://www.asteriskgsmmodule.com/index.html is a plug-in for the TDM400P card, allowing it to accept a GSM SIM card – instead of the standard FXO module.

Finally, a plug-in for Asterisk that negates the need to work with a GSM converter. The bad thing is that it requires a patch to the wctdm.c Zaptel driver, and aparently, isn’t yet available for DAHDI at all – but I guess this will be fixed in the short future. I surely hope that these guys will contact Digium and maybe introduce the driver into the main stream driver distro, after all, Digium doesn’t make GSM modules – so it’s no competing with any Digium product.

Today I sat down with a friend, a telecom consultant like myself. The main difference between the two of us is that, while I’m purely focused on Open Source technologies, he’s focused on proprietary technologies. Well, I can’t really blame him, after all, he used to be the CTO of a Tier-1 long-distance carrier in Israel and his track record of building large scale systems can’t be negated or dispersed at ease.

In any case, we were discussing the financial validity of obtaining an IP centrex platform vs. building one of your own based on Asterisk. On one hand of the scale, we had companies such as Broadsoft and the like, proprietary technologies for building IP centrex environments. In addition, we’ve discussed the various FlatPlanetPhone type companies, rendering IP centrex type services via Asterisk, in a hosted model solution. It is obvious that if you are going into the “retail” business, then you simply sign up as a reseller of anyone that hosts their own platform and go on from there – however, this was not the case.

Our discussion was based on the following assumption: a projected customer wants to host IP centrex like services to a total of 15,000 business subscribers, each one consisting of anything from 5 users at the low end to 60 users at the high end. The average calculation was 33 users and multiplied that by 7500 (half of the project users case) – giving us a total 247,500 connected phones. Ok, a Broadsoft of that size is WAY WAY WAY more expensive than our projected budget, and building our own IP centrex platform based on Asterisk would take too much time.

Recently, advances on the implementation of Asterisk utilizing Amazon EC2 had resulted with the implementation of a full office PBX system can be implemented within an EC2 instance. While the PBX is based on Asterisk + FreePBX, it surely can provide any of the required IP Centrex type of services a normal PBX platform will provide – in addition to the hosting environment which is highly reliable.

Now, a normal office works anything between 9 to 11 hours per day. Doing some Amazon math shows that running a PBX system, 11 hours per day, 22 days a week, yields a total of 242 hours per month. Thus, based upon a medium C1 instance, we’re talking about 49$ per month for running your PBX. Bandwidth for a PBX that size would cost around the 5$ per month, and the elastic IP is another 3$, thus, a total of 57$ per month, cost to the operator. The IP centrex model dictates a price per phone, thus, an office of 5 people, with a minimum charge of 39$ per phone per month, leaves us 137$ of profit per month. Let’s imagine that you’re using phones like SNOM 300 as your base phone, these are 80$ value at large quantities. Thus, the customer ROI is around the 4-5 months, which is normal in the services world.

Now, the PBX system can go online and offline on a daily basis, while being offline having its inbound calls directed to a company voicemail. This will save all the off-hours messages people may leave on the system and will also lower operational costs on a per day basis. If a company requires to have it’s PBX system up and running at all times, simply charge a bit more, no big deal.

In other words, the utilization of Amazon EC2 instances with proper AWS management and control can introduce a new business model – the Clouded PBX service. No more need to own large portions of data center realestate, no need to maintain large teams of IT and network personnel – just setup you AWS account, build your management system – and start earning cash. Look out Broadsoft, Asterisk and Amazon are coming – and a hail storm follows them.

As a part of my job, I manage and maintain customer platform – usually operating in the Calling Cards and VoIP services market. Over the course of time, I’ve learned to rely on some providers in this world, knowing that they work 99.999% of the time.

For example, i like working with DID numbers provided by Level3, GlobalCrossing and Voxbone. I have a fair dislike for DIDX and the like, simply due to the fact that their reliability, not the DIDX platform, but the providers themselves is questionable – at best.

So, why is this post called: “Battliing the GlobalCrossing CallerID blues”? simple, because the list that appeared before is now missing GlobalCrossing. Over the course of time, I’ve learned to live with the various quirks of GlobalCrossing, mainly, their inability to provide a proper e164 number as a part of the SIP headers. Usually, I would receive headers from global crossing that look like this:

FROM HEADER: <sip:3054230103@xxx.xxx.xxx.xxx>;tag=as54cf6928

Now, I new that in general, that didn’t post much of a problem, as long as it was consistent. However, starting today, some of the requests started looking like this:

FROM HEADER: <sip:13054230103@xxx.xxx.xxx.xxx>;tag=as1213141

However, to make things weird, one INVITE request would carry the non-valid e164 numbering, while the second INVITE may carry the correct format. In other words, there is no way to know exactly if the number is provided in full e164 or not. So, I tried doing some header mangling using Asterisk and other tools, however, nothing helped. Surely the format changed along the way, however, when I changed one side of the system, another side of the system broke – simply because it relied on something else – in other words, a fuck’n mess.

At this point, the problem is not yet resolved and i’m working with my DID provider to remedy the situation – after investigating it, the DID provider is currently bashing the heads at GlobalCrossing to fix the issue on their side. I will report back once I have more information.

If you suffered similar problems with other DID providers, I’d love to hear about it.

One of the biggest questions in the world of Asterisk is: “How many concurrent channels can be sustained with an Asterisk server?” – while many had tried answering the question, the definitive answer still alludes us. Even the title of this post says “3500 concurrent channels with Asterisk” doesn’t really say much about what really happend. In order to be able to understand what “concurrent channels” really means in the Asterisk world, let us take a look at some tests that were done in the past.

Asterisk as a Signalling Only Switch

This scenario is one of the most common scenarios in the testing world, and relies upon the basic principle of allowing media (RTP) to traverse from one end-point to the other, while Asterisk is out of the loop regarding anything relating to media processing (RTP). Examine the following diagram from one of the publicly available OpenSER manuals:

Direct Media Path between phones via a SIP Proxy

As you can see from the above, the media path is established between our 2 SIP endpoints.

This classic scenario had been tested in multiple cases, with varying codec negotiations, varying server hardware, varying endpoints, varying versions of Asterisk – no matter what the case was, the results were more or less the same. Transnexus had reported being able to sustain over 1,200 concurrent channels in this scenario, which makes perfect sense.

Why does it make sense? very simple, as Asterisk doesn’t manage or mangle RTP packets, Asterisk performs less work and the server also consumes less resources.

Asterisk as a Media Gateway

Another test that people had done numerous times is to utilize Asterisk a Media Gateway. People used it as a SIP to PSTN gateway, SIP to IAX2 gateway, even as a SIP to SIP transcoder gateway. In any case, the performance here varied immensly from one configuration to another, however, they all relied on a simple call routing mechanism of routing calls between endpoints and allowing Asterisk to handle media proxy tasks and/or handle codec translation tasks.

Depending on the tested codec, I’ve seen reports of sustain over 300 concurrent channels of media on a single server, while other claim for around the 140 concurrent channels mark – this again mostly relied on various hardware/software/network configurations – so there is nothing new in there.

These tests tell us nothing

While these tests are really nice in the theoretical plane of thinking, it doesn’t really help us in the design and implementation of an Asterisk system – no matter if it is an IVR system, a PBX system or a time entry phone system for that matter – it simply doesn’t provide that kind of information.

The Amazon EC2 performance test

In my previous post, Rock Solid Clouded Asterisk, I’ve discussed the various mathmatics involved in calculating the RoI factors of utilizing Cloud computing. One thing the article didn’t really tell us, did it really work?

Well, here are some of the test results that we managed to validate:

• Total number of Asterisk based Amazon EC2 instances used: 24
• Total number of concurrent channels sustained per instances (including media and logic): 80
• Average length of call: 45 seconds
• Total number of calls served: 2.84 Million dials
• Test length: approximately 36 hours

According to the above data, each server was required to dial an approximate 3300 dials every hour. So, let’s run the math again:

• 3300 Diales per hour
• 55 Dials per minute
• As each call is an average of 45 seconds, this means that each gateway generates 20 calls
  per second, and within 4 seconds fills the 80 channels limit per server.

According to the above numbers that we’ve measured, each of the Amazon EC2 instances used was utilized to about 50% of its CPU power, while consuming a load average of 2.4, which was mostly caused by I/O utilization for SIP and RTP handling.

Conclusion

When asking for the maximum performance of Asterisk, the question is incorrect. The correct question should be: “What is the maximum perfromance of Asterisk, utilizing X as the application layout?” – where X is the key factor for the performance. Asterisk application performance can vary immensly from one application to another, while both appear to be doing the exact same thing.

When asking your consultant or integrator for the top performance, be sure to include your business logic and application logic in the Asterisk server, so that they may be able to better answer your question. Asterisk as Asterisk is just a tools, asking for its performance is like asking how many stakes a butcher’s knife can cut – it’s a question of what kind’a steaks you intend on cutting.

Ok, it’s been almost 24 hours since I started my serious playing around with Amazon EC2, and I can honestly say that I’m tired – however, I’m very pleased with my results. Like any other experiment, this one started with a requirement. The requirement was to install and operate one of the dialer frameworks I’ve written in the past year on an EC2 based instance. In order to evaluate the process, let’s start with our baseline installation, meaning, what am I using in the real-world:

Hardware and Software Specification

The original machine answered to the following specification: Quad Core CPU, 2GB of RAM, 250GB of Hard Drive. My original machine was running CentOS 5.2 with an x86_64 kernel installed. In terms of software installed, we had Asterisk 1.4.22.1, MySQL 5.X, PHP, FreePBX, Apache and my dialer framework.

Introducing Amazon EC2 AMI

An EC2 AMI is basically an image of a computer, contained with a single installation manifest on the Amazon cloud computing system. AMI’s provide for the simplest manner to start using EC2, as these usually include a pre-defined server installation, that usually has some stuff already installed.

Choosing your AMI

Amazon provides a multitude of AMI’s to work with, unfortunately, most of these are either out-dated and the sheer number of these makes the choice somewhat overwhelming. I decided to start working with a working AMI image of Fedora Core 8, that already had the LAMP stack installed – the one I used was:

The YUM Repository issue

For some strange reason, the AMI images contained in the Amazon repository are all unable to perform any updates to their installed RPM packages. It took me a while to understand what’s wrong, but in general, the fedora project had simply removed the old releases from their repository, so I had to go in and manually modify the /etc/yum.repos.d/ configuration files. For you convenience, here is the repos list that I’m using at this point:

YUM Repositories for Amazon EC2 Fedora images <- click this to download the file

[development]
name=Fedora Core $releasever - Development Tree
baseurl=http://archives.fedoraproject.org/pub/archive/fedora/linux/core/development/$basearch/
enabled=1
gpgcheck=0

[extras-development]
name=Fedora Extras $releasever - Development Tree
baseurl=http://archives.fedoraproject.org/pub/archive/fedora/linux/extras/development/$basearch/
enabled=1
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-fedora-extras
gpgcheck=0

[extras]
name=Fedora Extras $releasever - $basearch
baseurl=http://archives.fedoraproject.org/pub/archive/fedora/linux/extras/$releasever/$basearch/
enabled=1
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-fedora-extras
gpgcheck=1

[updates-testing]
name=Fedora Core $releasever - $basearch - Test Updates
baseurl=http://archives.fedoraproject.org/pub/archive/fedora/linux/core/updates/testing/$releasever/$basearch/
enabled=1
gpgcheck=1
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-fedora-test

[updates-released]
name=Fedora Core $releasever - $basearch - Released Updates
baseurl=http://archives.fedoraproject.org/pub/archive/fedora/linux/core/updates/$releasever/$basearch/
enabled=1
gpgcheck=1
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-fedora

[base]
name=Fedora Core $releasever - $basearch - Base
baseurl=http://archives.fedoraproject.org/pub/archive/fedora/linux/core/$releasever/$basearch/os/
enabled=1
gpgcheck=1
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-fedora

If you are trying to install an instance, you may use the above for any Fedora based AMI on EC2.

Creating my own AMI

While going about and building a single server is fun, I needed a way to create my own installation AMI once I’ve completed my modifications. The Amazon EC2 resources website gives out a whole lot of information, which can be somewhat confusing for the first time reader. So, I searched for a better way to construct my own AMI image. I found the following web page, which was really really helpful: Azeez’s Notes.

Azeez’s blog gives a very concise and to the point, step by step, guide to building your own AMI image – it got me up and running in less than 10 minute – WAY TO GO AZEEZ.

So far, so good …

So far my installed instances are working just fine and I’m currently operating a cluster of a few of these dialer systems. I’m in the process of checking what kind of mileage I’ll get from the EC2 instances, in comparison to a real hardware server – which is really interesting.

My main objective here is to be able to build a dialer-on-demand framework, which will enable my customers to increase their capacity utilizing a simple web interface to initiate my AMI instances. I’ll report back with additional information once I have it.

Session Border Controllers (SBCs) are utilized as a means to providing both load balancing and security structures for VoIP networks. To be completely honest, 90% of my customers utilize SBC appliances, be it Acme Packet, Juniper, NexTone or others.

According to a report by Transnesus, a combination of OpenSER and Asterisk can be utilized as a Back-To-Back-User-Agent (B2BUA) structure – however, the general configuration and setup isn’t clear and straight forward. I’ve been thinking to myself: “Why hadn’t anyone written and Open Source SBC? could it be? usually there’s an Open Source alternative to any commercial product”.

Like any other search on the net, I’ve pointed my Firefox to Google, and typed the phrase “Open Source SBC”, aparently, such a thing exists from a company called Solegy – over at the web address: http://www.opensourcesip.org. So, I downloaded the source code, and after a 30 minute compilation phase (bearing in mind working on a Virtual server running under VMWARE Server) – the compilation completed.

Compiling was one thing, running it was a completely different thing – took me a while to realize where the binary is located and how the configuration works out – once I did that was a breeze. On my system, after compilation the binary was located according to the following:

[root@opensbc obj_linux_x86_r]# pwd
/root/OpenSBC-1.1.5-RC1-Bundle/opensbc/obj_linux_x86_r
[root@opensbc obj_linux_x86_r]# ./opensbc -x

Message from syslogd@ at Thu Nov 13 23:15:35 2008 ...
tvms OpenSBC[18900]: Starting service process "OpenSBC" v1.1.5-25

Per the information provided by Solegy, the OpenSBC project supports several modes of operations, ranging according to the following:

Full Mode - By default OpenSBC runs in full mode exposing its capability both as a
relay SIP proxy, Registrar and as a B2B User Agent. When OpenSBC receives an INVITE
or a REGISTER request it would follow the following procedure to make a decision how
to route a request:

● If the Request-URI resolves to a remote domain, the request will be relayed. If a
relay route is available, the request is sent to that route. If a relay route is not
available, then the URI is resolved via DNS.
● If the Startline-URI resolves as a local address and port, the To URI is checked
if it resolves to a local domain and port. If not, the request would be proxied
using Relay Routes or via DNS resolution. The Request URI would be rewritten to point
to the resolved route.
● INVITE: If both Request URI and To URI resolves to a local listener and port, the
B2BUA Route is used to route the INVITE.
● REGISTER: If both Request URI and To URI resolves to a local listener and port, the
local Registrar will process the registration. This would include Authorization of
the user.

B2BOnly Mode - This mode removes the relay capability but exposes the Registrar and
the B2BUA functionalities. This mode does not do the checks performed by Full Mode. It
will always process REGISTER and INVITE as local.

● INVITE: This mode always use B2BUA Route to route calls. If there is not corresponding
route found, a DNS resolutions is done against the Request URI or the To URI in case the
Request-URI resolves to a local address.
● REGISTER: Registrations are always handled by the local registrar.

Proxy Only Mode - This mode removes the B2BUA functionality but exposes Registrar and the
relay SIP Proxy functionalities
● Always uses Relay Routes for all messages including REGISTER. If a relay route is not
configured, Requests will be relayed using DNS resolution. If a registrations is resolved
as local, the registrar would handle the registration including authorization

B2BUpperReg Mode - This is almost the same as the B2BOnly mode but with the additional
capability of relaying registrations to upper registrars.

● INVITE: This mode always uses B2BUA Route.
● REGISTER: For registrations, it performs the Request URI and To URI checking and relay
for a remote domain or process the registration locally for local domains.
● Upper-Registration: This mode also has the capability to hijack-registrations towards
upstream registrars.

Per the above, I didn’t completely understand what I should use for normal IP phones operations, so, I guess I’m more or less on my own on this one. My general understanding says that I need to use the B2BupperReg mode, however, I can’t say I’m totally sure about it – I’ll be experimenting with OpenSBC and the virtual Asterisk servers i’ve written before over the couple of months.

Ever since the introduction of JaJah to the world, the world of free telephony services had been booming with various solutions and services. While each service concentrated on a different market niche, none of the services really is free of charge (at least not in full). For example, JaJah requires you to register and purchase additional air-time, RebTel operates under a similar approach, so does TruPhone and others alike.

Over the past 10 months, I’ve been working closely with a company called Parrot-Media, who operate the PokeTalk.com website.

PokeTalk Website

PokeTalk is a free international calls service, allowing users to make international calls FREE of charge. The service allows a user to make up to 50 free calls per month, of up to 10 calls per calls (that’s 500 of FREE minutes per month). Judging from normal ACD traffic around the world, a normal call duration is around 5-6 minutes, thus, the service is a great solution for making phone calls for FREE.

According to the PokeTalk economic model, the calls are being funded by the advertisement on the website. So, while you watch the ads on the website, the advertisers are paying for your call – it’s as simple as that.

As you may have guessed, the service utilizes the Asterisk Open Source PBX (after all, this is what I do). In addition, it utilizes a highly advanced, high-speed, highly-reliable Asterisk based dialer framework – enabling the system to initialize up to 140 calls per second, and sustain a total of 1200 concurrent phones calls across the entire platform. All in all, a fairly big and robust platform for a new service.

To enjoy the service, simple point your browser to http://www.poketalk.com, regsiter, and start making calls to your loved ones.

So, over the past 2 years, we’ve seen many Asterisk appliances – ranging from the Digium AADK on one side of the spectrum to the RoweTel IP04 appliance, it is clear that appliances are the hottest thing in the Asterisk sector to-date.

While most appliances take on a similar form, usually one of the following: an ADI BlackFin reference design or a PC-Engines Geode based appliance, one company decided to go completely the other way around – namely PIKA technologies. Now, don’t let the name PIKA fool you, these guys are no pokemons, they know their stuff.

While at AstriCon 2008, I’ve been approached by the people at PIKA technologies, asking me to test and evaluate their WARP Appliance product – a Linux based appliance framework, for building Linux based appliances, with a focus on telephony. They awarded me with an appliance to test, equipped with 5 FXS ports and 4 FXO ports, I was on my way to start my telephony appliance.

First thing’s first, computing power – while most appliances rely on fairly low scale computing power, the PIKA appliances boasts a massive PowerPC CPU, with a uCkernel based Linux 2.6.X kernel – while many other appliances still rely on 2.4.X. The on-board LCD display makes finding your appliance IP number a breeze, which is very important in my book. Opening the box is very easy, exposing its inner workings:

One thing nice about the LCD display is the fact that once you plug something in, the LCD will show the applicable LCD icon as connected – making for quick diagnostics for the lame sysadmin a breeze.

Now we’re getting to the slightly annoying part, the WARP appliance is not your average PBX appliance – in fact, it’s not a PBX at all – it’s a framework for building telephony appliances. So, if you want to get Asterisk + FreePBX to run on it, you’ll need to spend some time getting it up and running correctly. If you want to start working with the appliance, I suggest that you get to know cross-compiling and build tools, in order to utilize the buildtools provided with the appliance.

Over all, the appliance comes with a fairly stock Linux and Asterisk pre-compiled, with all the required modules to get a fully functional PBX running. For my initial tests, I’ve used the AsteriskGUI, which worked just fine with the system, making the configuration a breeze.

I’m currently evaluating their FreePBX build-tool, provided by Philippe from FreePBX, and we’ll see how that goes.