The rants and raves of a technogeek
Posts tagged sangoma
Call Analytics – Closed Alpha testing group
Mar 14th
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.
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Beyond the dialtone – PBX user experience revisited
Feb 12th
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?
- Image via Wikipedia
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.
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Digium TE205P vs. OpenVox D210P
Feb 2nd
If there is one thing I like doing is testing hardware, specifically, testing new hardware that is related to Asterisk. I was more than pleased when OpenVox had approached me, asking to review one of their products – specifically after I once announced that I really dislike cheap clone cards. So, I got OpenVox’s D210P card, which is a fairly similar clone to the TE205/TE210 of Digium, and I decided to take a it for a test drive.
So, first off, lets take a look at Digium’s TE205 card:
Digium TE205P Card
The card is based upon two specific chips, the Xilinx Spartan FPGA and an Inifineon based Quad E1/T1/J1 framer chip. Technically speaking, the entire brain of the outfit is located in the Xilinx FPGA (naturally), which on the TE205P now enables remote firmware upgrades and some additional features. Digium had been using Xilinx based boards for over 8 years now, and they’ve been doing the job more than well.
Now, let’s take a look at the OpenVox clone board:
OpenVox D210 Card
OpenVox utilizes the same Inifineon framer chip (well, it’s a clone after all), while utilizing the Lattice Mico8 FPGA chip. Now, from a technological point of view, I couldn’t really find much differences between the Mico8 and the Spartan, beside a minor differences here and there – but these are not important. So, I proceeded to testing the card with Asterisk. So, the nice thing about this clone is that it doesn’t require patches to the stock version of DAHDI, which in my book means that OpenVox are aiming at being a real-clone, not some would be patched version of a clone – so that’s good. Installation was fairly similar to that of the Digium TE205P card, so I couldn’t really find specifics in there to prefer one over the latter. So, I started testing the card in various situations: Normal telephony, 3G based transmission (64kbps bearer capability), dropped calls during high loads and checking CPU/Load spikes during high usage.
The Test Scenario and Comparison
All of the above tests were conducted according to the following scenario:
Testing Lab Server
In general, I’ve connected 3 different IP phones to the testing server: A Polycom 650, a SNOM 370 and a Grandstream GXP2000. All IP phones include the latest firmwares and updates and were all working flawlessly with another similar setup, so I assumed they were all bug and issue free for the testing lab. The main reason I’m using 64Bit CentOS is simply due to the fact that all my servers are 64Bit capable (mainly E5410 and E5405).
Test 1: Normal Telephony
Well, in general, the card does exactly what it should – provides a connection to an E1 circuit (we only have E1 circuits in Israel). I’ve conducted normal telephony functions from all the above mentioned phones. In general, I’ve conduct from each phone a total of 40 calls, and repeated the test once for the Digium TE205P card and once for the OpenVox D210P card. The results were fairly similar with a slight advantage for Digium. In general, the OpenVox card had slipped about 4% of the calls, mainly to an IRQ miss that occurred for some reason. With the Digium card, the IRQ misses were not exhibited, allowing for all 120 calls to traverse normally.
Conclusion: In a normal office telephony scenario, the D210P is a fair choice – however, not my preference for a Call Center or a service provider.
Test 2: 3G based transmission (64kbps bearer capability)
I’ve been playing around with IVVR and Asterisk, mainly using the Fontventa H264 packages for Asterisk (that’s why I used 1.4 branch). With this test, the D210P provided less then medium results, specifically when trying to stream large 3gpp based video streams, while the TE205P had showed no specific issue with the transmission. Main issues exhibited were related to choppy video streams, causing jumps in the stream. The Digium card was fully capable of stream the video without a hitch. Now, I won’t hold this again OpenVox, as this usage is fairly advanced and is required by a very small portion of the market, but I believe they still have some work to do there. As they are using the same framer as Digium, I would deduce that their firmware is either an older import from Digium (reverse engineer) or some other firmware related issue.
Conclusion: Not a pick for 3G transmission with Asterisk.
Test 3: Dropped calls during high loads
No matter what test I did, with OpenVox I’ve always received a dropped call ratio of around 3-4% – when at high loads that went up to around 7%. When I mean high loads, I mean generating 30 outbound calls from Asterisk to one circuit, then receiving them on the second port (yes, a back-loop). I’ve conducted 100 runs of this test, at various speeds. It would appear that when generating calls with a 100ms interval between one initiation to another on the circuit, the OpenVox will drop a call here and there – at sporadic intervals. This may be actually related to the IRQ misses exhibited in Test 1.
Conclusion: If you have high load anticipated – OpenVox is not the choice for you.
Test 4: CPU Load/Spikes
It is a well known fact that all card that are used with Asterisk introduce load spikes of a sporadic nature. In the past, the masters of low spikes were Sangoma, however, with the introduction of Digium’s VoiceBus, that balance had tipped and Digium took the upper hand. In order to evaluate the spikes, I’ve monitor the machines’ load while having 30 calls traverse from one port to the other. The calls were playing back a static file of 5 minutes, and after disconnecting the calls would generate and additional one and continue from there. Both cards exhibited slight spikes when multiple calls either originate or disconnect, however, the CPU spikes that the OpenVox card had exhibited were about 40% higher than the ones exhibited by Digium and there were more spikes than with Digium.
Conclusion: If your system isn’t as beefy as mine, and you need full capacity – OpenVox isn’t the choice for you
Overall Operational Conclusion
The OpenVox card promises to be a low-cost alternative to the Digium card, and it surely delivers. Over all, if you have an office PBX system or a low scale IVR environment, the OpenVox alternative can be evaluated, although it’s not my personal favorite. Sure, in many cases I can say: “OpenVox would do the job” – but hey, I would always rather go with the original and not the clone. I believe that OpenVox are far ahead of its clone competitors (Atcom, Yeastar, Varion, PhonicEQ, etc), simply because it does a better job at building and designing a better card – however, they still have some way to go in order to be completely in-lined with Digium and Sangoma.
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Asterisk updates, rants and raves
Apr 1st
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.
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.
Sangoma USBfxo: too little, too late…
Feb 11th
Sangoma recently introduced a new FXO product, the USBfxo. The USBfxo is a dual FXO port device, connected to your Asterisk server via a USB connection. Now, while I do admire the way Sangoma keeps trying to kick it up a notch with new products, but isn’t Sangoma a little late to jump on the USB train?
Sangoma USBfxo Device
Xorcom had been in this business for 4 years now and I see no reason why would the Sangoma product be any better than the Xorcom product. In addition, if Sangoma is targeting their product at the very low-end PBX systems, in my book, they actually missed the product line. In my view, if Sangoma wants to put a proper USB device on the market, it should have a minimum of 4 ports on it, 3 FXO and 1 FXS. You are probably wondering why I’m propsing such a weird combo, well, the reason is simple – Fax machines and they yet to be improved Asterisk FAX capabilities, and the fact that people still use FXS port of physical fax machines. I’m one of the biggest Asterisk and VoIP promoters I know, and even I use a physical fax machine at some points in time. True I used Hylafax and IAXmodem to receive most of my fax transmissions, but when it comes to sending faxes, nothing beats a physical machine.
So, as I started saying, Sorry Sangoma, too little, too late … better luck next time!
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