Planet Asterisk

Voip-Info: Asterisk UNISTIM channels / Re: i2002 NTDU91 Phones [ID: 57900]

Wed, 27 Aug 2008 23:06:40 +0000

You should try this for new firmwares : Search for const static unsigned char packet_recv_mac_addr[] = { 0xff, 0xff, 0xff, 0xff, 0x9a, 0x0d, 0x07, 0x31, 0x38 /*MacAddr */ }; and delete 0x31, 0x38, you should get this line after the change : { 0xff, 0xff, 0xff, 0xff, 0x9a, 0x0d, 0x07 /*MacAddr */ };
blu () at 2008-08-27 23:06 GMT

Voip-Info: / asterisk serious problem [ID: 57892]

Wed, 27 Aug 2008 18:05:26 +0000

hi everyone im a new user about asterisk so i have a real question, i have a server with centos5, kernel 2.6.18, samba, openldap as backend, when ive install asterisk and zaptel everythings looks fine but when i restart the server ive various problems, i cant login to the samba domain, cant add computers to the domain, the log says something like this Aug 12 11:52:35 sbs smbd[6027]: [2008/08/12 11:52:35, 0] passdb/pdb_get_set.c:pdb_get_group_sid(211) Aug 12 11:52:35 sbs smbd[6027]: pdb_get_group_sid: Failed to find Unix account for manager Aug 12 11:52:35 sbs smbd[6027]: [2008/08/12 11:52:35, 0] auth/auth_sam.c:check_sam_security(352) Aug 12 11:52:35 sbs smbd[6027]: check_sam_security: make_server_info_sam() failed with 'NT_STATUS_NO_SUCH_USER' Aug 12 11:52:35 sbs smbd[6028]: [2008/08/12 11:52:35, 0] passdb/pdb_get_set.c:pdb_get_group_sid(211) Aug 12 11:52:35 sbs smbd[6028]: pdb_get_group_sid: Failed to find Unix account for manager Aug 12 11:52:35 sbs smbd[6029]: [2008/08/12 11:52:35, 0] passdb/pdb_get_set.c:pdb_get_group_sid(211) Aug 12 11:52:35 sbs smbd[6029]: pdb_get_group_sid: Failed to find Unix account for manager Aug 12 11:52:35 sbs smbd[6029]: [2008/08/12 11:52:35, 0] auth/auth_sam.c:check_sam_security(352) Aug 12 11:52:35 sbs smbd[6029]: check_sam_security: make_server_info_sam() failed with 'NT_STATUS_NO_SUCH_USER'
saulrainier () at 2008-08-27 18:05 GMT

Tom Keating: Microsoft develops Vi-Fi (Vehicle Wi-Fi). VoIP while driving baby!

Wed, 27 Aug 2008 11:30:32 +0000

A new technology developed by Microsoft in alliance with the University of Massachusetts (UMASS), and the University of Washington has resulted in a new technology called Vi-Fi (Vehicle Wi-Fi). Vi-Fi solves the problem of hopping from one Wi-Fi wireless access point (WAP) to another when signal strength diminishes. According to the article, the goal of Vi-Fi could be to power VoIP calls, Internet, and radio in cars of the future.

Ironically, Andy wrote a controversial blog this week about VoIPing while flying (using Aircell) -- well now you can VoIP while driving too! Of course, I've been using VoIP over EVDO for some time now while driving. I was even able to register my Windows Mobile 6.1 device (EVDO as well) to an Asterisk server sitting in my office and make/receive office calls.

With this forthcoming technology, who needs XM or SIRIUS when you can stream radio for free over the internet with hundreds if not thousands to choose from? Time to short satellite radio stocks?

The system they are developing will provide users with on-the-go internet access more cheaply than existing cellular broadband systems. The challenge this alliance aims to solve can be explained by this excerpt of the article:

The key to Microsoft's new system is a steady signal. In Wi-Fi networks, as you travel overland, each signal station (base station) only broadcasts so far, so you have to hop between stations. During these transitions, known as "hard handoffs", the signal strength typically drops. At best this makes car Wi-Fi unpleasant, and at worst it makes it unworkable.

The key to to solving this challenge is the use of multiple base stations. The station can send and receive signals from more than on base station, and the strongest strength gets assigned as the "anchor" system, while the other systems are used as "auxiliary" systems. A complex algorithm is constantly calculating the probability of a packet not reaching the main station. If that probability reaches a certain threshold the auxiliary stations try to forward it to the main station.

They did some test trials at Microsoft's campus and found the system eliminated nearly all the problems of network hopping while driving. Just don't expect Wi-Fi / Vi-Fi coverage to be as ubiquitous as say EVDO, 3G, etc. while driving. EVDO, 3G and other cellular data transmissions can travel a lot further than Wi-Fi signals, which are only rated for 300feet. So you'll still need a ton of these specialized Wi-Fi stations with the special software to get decent coverage.

But considering most homes have Wi-Fi APs, it could be very cool if you could build a massive Vi-Fi network using consumer's existing Wi-Fi APs. Simple firmware upgrade? But then you get into the whole security issues, bandwidth theft, etc. Still, the work Microsoft and these universities are doing is fascinating. How soon will be before we have truly mobile and ubiquitous Wi-Fi Internet access? Who knows? But my EVDO card works just fine almost wherever I go. It's not as fast as Wi-Fi but it works well enough.

Tags: car, cars, driving, EVDO, Internet, Microsoft, UMASS, University of Massachusetts, University of Washington, Vi-Fi, VoIP, Wi-Fi, wireless

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Holy Toledo! Cheap Wireless T1 & T3 Internet! - Feb 01, 2008
Bye Bye Unlimited Flat-Rate Internet - Jan 18, 2008

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Tom Keating: Asterisk GUI 2.0 launches

Tue, 26 Aug 2008 16:08:42 +0000

Yesterday, afternoon I mentioned Asterisk GUI 2.0. Later that evening, Digium "officially" announced Asterisk GUI 2.0 and included some screenshots, which you can check out here:

here's a few screenshots to wet your whistle:

Tags: Asterisk GUI 2.0, digium, GUI, VoIP

Related Entries

Digium AsteriskNOW News and more... - Aug 25, 2008
Leslie Conway Joins Digium - Jul 11, 2008
Microsoft develops Vi-Fi (Vehicle Wi-Fi). VoIP while driving baby! - Aug 27, 2008
Digium Headquarters Tour - Aug 26, 2008
fring touchtone DTMF support - uh hello? - Aug 25, 2008
Meeting with Digium - Aug 21, 2008
VoSKY Exchange Pro VISIP-EX SIP PBX-to-Skype gateway - Aug 19, 2008
Digium AA50 1.2 Software Released - Aug 19, 2008
How much data does fring consume? - Aug 05, 2008
Well there goes VoIP's 5 nines (99.999) of reliability! - Aug 04, 2008

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Tom Keating: Digium Headquarters Tour

Tue, 26 Aug 2008 15:36:47 +0000

During my tour of Digium's headquarters in Huntsville, Alabama I snapped quite a few photos that I thought I'd share. I'll just share a few here, but if you want to see them all, go check out the online photo album I created specifically for Digium:
http://blog.tmcnet.com/blog/tom-keating/photos/digium/

Be sure to check my other photo albums as well:
http://blog.tmcnet.com/blog/tom-keating/photos/

The new Digium headquarters is very impressive. I couldn't help but be in awe of what Mark Spencer created. Obviously, there is money to be made in open source, and seeing the fruits of Mark's and his team's labor in this state-of-the-art building was proof of that. I can't help but reminisce back to my November 2001 column titled "In Search Of A Linux-Based PBX" where I listed several "young" Linux-based PBX solutions. One such solution I discovered was Asterisk. Check out what I wrote back in November 2001 to see how far Asterisk has come:

As my search continued, I stumbled upon a company called Asterisk. According to their Web site, Asterisk is an open source PBX and general telephony toolkit that runs on the Linux operating system. Asterisk provides a set of APIs that essentially make it a type of middleware between Internet and telephony channels like VoIP, voice over frame relay, etc., and telephony and Internet applications like voice mail, phone directories, call parking, and so forth. Asterisk supports a flexible and extensible channel API, allowing any number of real hardware or software interfaces. It is purported to support ISDN, PRI, T1, and POTS through an Adtran Atlas. The Quicknet Internet PhoneJack and Quicknet Internet LineJack are both supported. Plans to support Lucent-based Winmodems also in the works. This product is not a turnkey out-of-the-box Linux-based PBX, and its still under development, but it does have several telephony functions working, such as call bridging, call transfer, call parking, and rudimentary voice mail. Check them out.

There was no Digium telephony cards back then. It ran on Adtran hardware and some Quicknet PhoneJack cards. Who knew that Asterisk would explode and become the dominant Linux-based IP-PBX solution?

Well, there is no better proof on how far Asterisk and Digium have come than to see their brand new building. Check out these photos:

Front side of the building

Digium front lobby - Nice cafe to get your morning caffeine fix (coffee) before you head off to do some Asterisk coding.

This is the Asterisk logo in the center of Digium's headquarters on the 1st floor taken from the 2nd or 3rd floor above. Doesn't this remind you of the Star Trek Enterprise's warp core? (Seen below from a side view. Couldn't find a top-view image which would be more similar)

I count 6 fans and at least 4 heat sinks. Now that is some serious coolage! This might be the AA350 I snapped but not sure.

This is Rotary Dial Phone with an Ethernet Jack making it the world's only rotary IP phone! It was in one of the Digium offices I passed by.

Digium has many Collaboration Rooms. This is just one of them. Note the (almost) floor-to-ceiling whiteboard for plenty of collaboration space.

Just a shot of the hallway with the cabling system overhead. Hmm, I only see like 7 cables. What sort of magical networking is Digium doing that they only need 7 cables in a company of 150 people?

Digium has several Collaboration Rooms. Note the orange chairs to match the orange Asterisk logo.

Don't zoom in on the whiteboard. It contains top secret Digium information on how they plan on taking over the open source telephony world. Oh wait, they're already doing that.

Can't call yourself a software company without a foosball gaming table! Though I did notice Digium's vending machines only sold Dr. Pepper and not Diet Coke. Sacrilege I say!

Digium's training class room. Each student gets their own PC to install & configure Asterisk.

Digium and TMC having dinner
Left-to-right clockwise: Steve Sokol, John Todd, Russell Bryant, Dave Rodriguez (TMC), Greg Galitzine (TMC), Jane Brooks, Tom Keating (TMC), Bill Miller

There are several other pictures in the Digium photo album. Go check them out.

Tags: Alabama, Asterisk, Bill Miller, Dave Rodriguez, Digium, Greg Galitzine, Huntsville, IP-PBX, Jane Brooks, John Todd, linux, Mark Spencer, Russell Bryant, Steve Sokol, Tom Keating, voip

Related Entries

Digium podcast of Mark Spencer's new role - Jan 30, 2007
Digium - The Showstopper! - May 18, 2007
Leslie Conway Joins Digium - Jul 11, 2008
Awesome Adtran & Digium article - Jul 08, 2008
Interview With Asterisk Founder, Mark Spencer - Jun 26, 2008
Shoretel Rumors - May 01, 2008
Rev B of Astfin's BRI (ISDN) Asterisk Appliance Arrives - Apr 29, 2008
PhoneFromHere.com & Digium Ink 5 Year Deal - Apr 09, 2008
Digium reaches 1 million download milestone - Dec 19, 2007
Astfin announces Asterisk Appliance based on Blackfin - Oct 30, 2007

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Voip-Info: / Perl DB_File and Asterisk [ID: 57844]

Tue, 26 Aug 2008 07:19:59 +0000

Hi, I would like to use a perl script to manage my AstDB using DB_File module. If I write a family/key-value in my database from perl, reading it from asterisk CLI shows one missing digit in key and one missing digit in value, but reading in from perl shows all the digits. For example: my perl: tie %h, "DB_File", "/var/lib/asterisk/astdb", O_RDWR, 0640, $DB_BTREE or die "cannot open DB"; %h{'personalgreeting/46109'} = 'yes'; Asterisk CLI: aks1-dev*CLI> database show personalgreeting/4610 : ye Perl: foreach $key (keys(%h)){ print "$key -> $h{$key}\n"; } script output: personalgreeting/46109 -> yes Any idea? Thanks for your help vincent
vbex () at 2008-08-26 07:19 GMT

Tom Keating: fring touchtone DTMF support - uh hello?

Mon, 25 Aug 2008 21:53:15 +0000

I've been playing around with fring for awhile now. I like fring which I like to call "the Swiss army knife of mobile VoIP and IM", since it works with Skype, SIP, MSN Messenger, Yahoo! Messenger, Google Talk, ICQ, and AIM. Because it supports SIP, I'm able to configure fring to work on Asterisk-based systems to be able to make and receive phone calls.

Receiving calls to my '149' SIP extension (see screenshot) works great on my Windows Mobile 6.1 Verizon XV6700 (Apache) phone. in fact, VoIP using fring even works over a cellular data connection, unlike the Apple iPhone which blocks VoIP over cellular and I'm sure you heard about Nokia announcing they will no longer support VoIP. Om Malik wrote about this last week. As a gadget freak, people have been telling me I should dump my Windows Mobile for an iPhone. So let's see here, no VoIP on popular Nokia phones, no VoIP on the popular Apple iPhone (except over Wi-Fi). Further, you can't even run Skype on the iPhone. Tell me again why I should dump my Windows Mobile phone for an iPhone? Ok, I know, I know, it has a cool multi-touch interface and some other great apps. I'm getting an iPhone to test/play around with, but I already know I won't like it's VoIP capabilities.

Getting back to fring, my one complaint about it is that it doesn't support DMTF (touchtones) for traversing voicemail, ACDs/IVRs, etc. So it's useless if you want to call your credit card company, bank, etc. I'm shocked they don't support it yet. There are other VoIP apps for Windows Mobile that support DTMF, such as Talkonaut, so I'm not sure why this is such a technical hurdle for fring. It's the one thing that stops me from using fring more regularly. One other minor complaint is that it doesn't have a quick redial button. Other than that, fring works fantastic. I can make Skype VoIP calls, SIP-based calls, and of course the built-in IM support for the top IM providers. Take that iPhone!

Tags: Apache, DTMF, fring, iphone, mobile phone, nokia, Om Malik, sip, skype, touch tones, Verizon, VoIP, Windows Mobile 6.1, XV6700

Related Entries

Skype and other VoIP Apps on Mobile Phones - Apr 01, 2008
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fring coming to Windows Mobile phones - Jun 19, 2007
How much data does fring consume? - Aug 05, 2008
Packet8 MobileTalk on iPhone Adds VoIP Calling - Jul 11, 2008
Nokia N78 Comes to U.S. - Jun 24, 2008
sipgate adds SIP-based VoIP support to iPhone - Jun 04, 2008
O-FONE Supports SIP on Symbian S60 mobile phone - May 07, 2008
Talkonaut 4.0 native Symbian S60 edition released - Feb 28, 2008
Jajah native SIP client for EM-ONE phone - Oct 25, 2007

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Voip-Info: / voip calls lose audio [ID: 57829]

Mon, 25 Aug 2008 21:33:16 +0000

Hello. I have asterisk 1.4.21.2. With a pstn and a voip channel. the voip channel handles all calls outside of me local area. The problem I have is that on the voip channel, after a few minutes, sometimes less, I can not hear the outside party but they can hear me. It does not matter who originates the call and it is only on the voip channel. Any ideas?
jayboogee () at 2008-08-25 21:33 GMT

Voip-Info: / Re: FreeSwitch and Radius authorization [ID: 57826]

Mon, 25 Aug 2008 19:57:16 +0000

OpenSer as Radius auth. and FreeSwitch for b2b/media proxy/conference/voicemail/diag.message - it will work ? ----- In working scheme - openser+auth. module+b2b module and Asterisk for /conference/voicemail/diag.message ..
mmx_plus () at 2008-08-25 19:57 GMT

Voip-Info: / voip calls lose audio [ID: 57823]

Mon, 25 Aug 2008 19:37:53 +0000

Tom Keating: Digium Asterisk Appliance AA1080 or AA1060?

Mon, 25 Aug 2008 18:42:04 +0000

Bill Miller commented I may have seen some yet-unreleased products during my tour. During my tour of Digium's headquarters, I snapped some photos, so I thought perhaps one of my photos might contain some top secret Digium product, so I planned to give them a closer look.

Sure enough two of my photos caught my eye. I zoomed in on the photos and noticed a Digium Asterisk Appliance with a model number printed that says "AA1080" or "AA1060". I can't be sure which model number is correct. Although my Canon Powershot G9 is 12.1 mexapixels, I was too far away when I snapped the photo to get a good resolution shot of the model name. I put on my best CSI and CIA photo enhancement techniques (aka Photoshop) to try and enhance the image, but I still can't make out if it's AA1080 or AA1060.

Nevertheless, it would appear Digium has a yet-unreleased 1U appliance called the AA1080 or AA1060. Today, Digium has only announced the Digium AA60, AA300, and AA350 appliances. So the AA1080/AA1060 could be the next iteration. Check out the two photos for yourself...

Click for Full Size - zoom in by the Linux license plate.

Click for Full Size - zoom in by the Linux license plate.

Update: 3:35pm
Bill Miller, VP of Product Management at Digium writes to tell me, "There are no plans at this time to launch what you saw, but those units you saw are the voice system of Interop for the second time as we are technology sponsors with Polycom."

Ah well, it was fun while it lasted, but it appears no AA1080 or AA1060 will be coming to market -- or maybe that's just want Bill wants me to believe.

Tags: AA1060, AA1080, Digium Asterisk Appliance, voip

Related Entries

IEEE standardizes 801.11r fast Wi-Fi roaming - Aug 27, 2008
Microsoft develops Vi-Fi (Vehicle Wi-Fi). VoIP while driving baby! - Aug 27, 2008
Asterisk GUI 2.0 launches - Aug 26, 2008
Digium Headquarters Tour - Aug 26, 2008
fring touchtone DTMF support - uh hello? - Aug 25, 2008
Digium AsteriskNOW News and more... - Aug 25, 2008
Internet Finally Takes Off into the Clouds - Aug 20, 2008
VoSKY Exchange Pro VISIP-EX SIP PBX-to-Skype gateway - Aug 19, 2008
Digium AA50 1.2 Software Released - Aug 19, 2008
AT&T CallVantage One Step Closer to Oblivion - Aug 15, 2008

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Tom Keating: Digium AsteriskNOW News and more...

Mon, 25 Aug 2008 18:32:19 +0000

As I hinted last week, I would have some "interesting" news to tell after my visit to Digium's brand new headquarters along with colleagues Greg Galitzine and Dave Rodriguez. Before I get into the MAJOR news being made by Digium, let me cover what else I learned on my trip. It had been 3 years since I last visited Digium down in Huntville, Alabama.

After getting a grand tour of the new Digium building we sat down in a state-of-art boardroom complete with a high-end Polycom IP conferencing unit, theater lighting, and a projector with motorized retractable screen. The boardroom was named the Mark C. Smith Boardroom to honor Mark Smith, who was the founder and chief executive officer of ADTRAN and Mark Spencer's mentor. A picture honoring Mark Smith sits on a table in the room. We all admire how Mark Spencer was able to transform open source telephony - myself included - but no doubt Mark Smith and his company ADTRAN played a major role in making Asterisk the phenomenon it is today.

During our full-day meeting, Digium wanted to fill us in on the latest happenings going on and the future directions Digium was planning to take. One of the first takeaways I took was that their 1.5 year term CEO, Danny Windham believes strongly that Asterisk should be classified as an "engine". It's the "core guts" on which other software from third-parties can be added. For instance, Asterisk is not simply an IP-PBX, as of 1.6 it has an SS7 stack enabling Asterisk to be used in carrier and service provider applications. Another example they gave was how Asterisk can run just about any protocol, and were quick to point out that sipXecs, an open source alternative recently acquired by Nortel, can't even natively run Nortel's own digital phones while Asterisk works just fine with Nortel phones.

Another important takeaway was that Digium said flat out that claims Asterisk could not scale were false or at the very least were misleading since they were based on older versions of Asterisk. In fact, it seemed Digium took great personal umbrage that some of their competitors were making these claims. As evidence of scalability, they mentioned a particular service provider running thousands of ports and are currently writing a write paper case study on this customer. When they publish it, I'll be sure to include excerpts in my blog.

One last takeaway that is important to mention is that Digium considers themselves to be the "Benevolent Asterisk Project Sponsor and Maintainer". This is a key mantra that they want to reinforce. They have no plans to ever fork Asterisk into say an open source version and a premium closed source version. They know there are many companies that do add proprietary code on top of the Asterisk engine and don't give back to the community, but Digium has no plans on monetizing Asterisk that way. I should point out that corporations that purchase Asterisk Business Edition (ABE) do have the right under the dual licensing to add their own code without giving it back to the community, however, there are companies that have taken the Asterisk engine, didn't pay for ABE, added their own code, and didn't give the code to the community.

Some interesting statistics about Digium that they told me during the meeting that I thought I'd share. First, Digium currently has 150 employees and has had 26 consecutive growth quarters. The headquarters are now running on Switchvox, which Digium acquired in September of last year. This is the first year that Digium has been named to Gartner's renowned visionaries quadrant. One stat that I've always wonder about is what percentage of Asterisk code is developed by the open source community and what percentage by Digium. Well, the answer is 50% each, which was a little surprising. I would have expected the open source community to be higher, but this goes to show you how important Digium is to the future of Asterisk. They are obviously paying for developers and making an investment in improving Asterisk.

Some other interesting stats:

Over 1,000,000 downloads in 2007
Project 2008 downloads at over 1.54M
Over 4 million servers deployed with Asterisk
Over 56,000 active on forums
Over 17,700 on active Asterisk mailing lists
Over 7,248 on our Bug Tracker
Over 400 active contributors
Over 200 service providers worldwide using Asterisk

Further, the number of commits to Asterisk Trunk was 3,498 (mainly Asterisk 1.6 related) and the number of commits to Asterisk 1.4 was 1,401 (bug fixes). A staggering 132GB of bandwidth was consumed in 2008 YTD.

Here's an interesting photo I took from the 2nd or 3rd floor aimed at the Asterisk logo which I believe is the center of the 3 buildings. From what I'm told if you fly over Digium's building, the three buildings form the Asterisk logo. For some reason looking down at this circular area reminded me of the Star Trek Enterprise's warp core. Forget "Powered by Intel", this building is "Powered by Asterisk".

Digium recently hired Stephen Burcham, the Director of Quality who came from a company that designed controllers for anti-lock braking systems, which demand the most stringent quality control and reliability standards. Stephen has brought that same high standards mentality to Digium. Surprisingly, Digium's hardware production is done by two American companies, so it's Made in the U.S.A. Considering many electronics can be done more cheaply in China, I was happy to hear Digium was using American companies to manufacture their goods. In fact, Digium is using two local Alabama companies and they told me that they actually made in an investment in at least one of them to purchase a higher-end process manufacturing machine to meet Digium's stringent quality requirements.

One of the biggest challenges facing Digium has less to do with technology and more to do with market perception. Digium told us that they have to battle Cisco and their claims that Asterisk is not a business class solution. I found it fascinating that Cisco is still able to leverage their own name brand recognition to get (scare?) customer wins, but if anything that's great news for Digium and Asterisk. It isn't technical comparisons or feature-sets that Asterisk is losing customers to Cisco, it's name brand recognition. That'll improve with time and my guess is very quickly since Asterisk has a full head of steam.

Now for the interesting news. AsteriskNOW which previously ran on rPath will now be using the very popular CentOS (5.2) distribution. Now that it runs on CentOS it will support easy operating system updates (via .rpm files). Also new in AsteriskNOW will be a yum repository for easy updates of Asterisk itself. Just 'yum' the lastest version of Asterisk and it will upgrade your version of Asterisk without screwing up your customizations.

Also new is that the next release of AsteriskNOW (1.5) will bundle the very popular FreePBX front-end GUI. This is big news! FreePBX has become the defacto standard for web-based GUI administration of Asterisk, so now Asterisk fans can download AsteriskNOW, boot of the self-install CD and not only getting a working copy of AsteriskNOW, but FreePBX as well! This isn't to say Digium will stop developing their own GUI. In fact, when I asked that question, they said categorically they will continue to develop and improve their Asterisk GUI 2.0, which is currently installed on Asterisk Business Edition (ABE) and the Digium Asterisk Appliance 50 (AA50), as part of the 1.2 release I recently blogged about. Digium told me they expect the next release of AsteriskNOW to be available by Astricon.

Some other interesting news to talk about soon and I want to blog my grand tour of Digium's state-of-the-art facility, but I'll break that out into a separate blog post. Stay tuned...

Tags: adtran, AsteriskNOW, Cisco, Digium, Digium Asterisk Appliance 50, FreePBX, Mark C. Smith, Mark Spencer, nortel, VoIP

Related Entries

The IP-PBX Energy Wars... - Aug 04, 2008
Leslie Conway Joins Digium - Jul 11, 2008
Avaya #1 VoIP PBX Vendor? - Aug 21, 2007
Top 10 Reasons Why I love Asterisk - Aug 03, 2007
Digium podcast of Mark Spencer's new role - Jan 30, 2007
Digium Headquarters Tour - Aug 26, 2008
Digium AA50 1.2 Software Released - Aug 19, 2008
Cisco Manager Meets Jurassic Park - Jul 10, 2008
Awesome Adtran & Digium article - Jul 08, 2008
PhoneFromHere.com & Digium Ink 5 Year Deal - Apr 09, 2008

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Comments on this Entry:

(tony on Aug 25, 2008 6:32 PM) That's too bad...rPath now supports rpm uploads for their build software. Well, I guess they can now afford rebasing eh?

(Tony Rybczynsk on Aug 26, 2008 9:51 AM) Nortel didn't acquire sipXecs, we acquired Pingtel. Both Nortel and Pingtel architects see sipXecs as a highly scalable distributed SIP-optimized architecture. It's all about interoperability and applications, not about support for TDM phones (the current generation of Nortel IP phones are a firmware change from SIP). What's important of course is delivering value to customers. Our SCS 500 UC offer for SMBs is based on sipXecs and is distributed via IBM and Dell, reaching buyers who are more application than telecom oriented.

Voip-Info: / Request Pin?? [ID: 57815]

Mon, 25 Aug 2008 17:11:43 +0000

Hello to all, i will use asterisk 1.4 and i want any caller digit the pin stored into file .txt or extension.conf for continue or for continue the script. I have used Autenticate(PIN) but cannot work, anything can help me? Awaiting your reply. Best regards
l.mannucci () at 2008-08-25 17:11 GMT

Nerd Vittles: Free Asterisk Calls to Zillions of Phones with ENUM and Gizmo5’s Backdoor Dialing

Mon, 25 Aug 2008 07:00:00 +0000

What do Asterisk calls to Sprint, Verizon, and T-Mobile cellphones plus 40 million Asterisk systems around the world have in common? They're all free calls! And today we'll show you how...

Voip-Info: Asterisk High Availability Solutions / Re: Asterisk wih Ultramonkey load balancing; bug in real server health check [ID: 57771]

Sat, 23 Aug 2008 20:10:31 +0000

Hi Madhuri, I'm very much interested on how to make ultramonkey work with asterisk. Currently i have ultramonkey setup to load balance my web traffic (port 80 and 443) and it's working fine. I tried adding sip service, my ldirectord.cf looks like this: virtual=12.13.14.155:5060 real=12.13.14.130:5060 gate real=12.13.14.131:5060 gate service=sip scheduler=rr persistent=600 protocol=udp checktype=connect but does not seem to work when i try configure my phone to register on 12.13.14.155 any help would be really appreciated.
nhadie () at 2008-08-23 20:10 GMT

Voip-Info: X100P clone / Motorola modem (PCI ID 1057:3052) support [ID: 57740]

Fri, 22 Aug 2008 14:04:19 +0000

I thought I'd give Asterisk a go, but found out that my Motorola modem (PCI ID 1057:3052) was based on the Si3052 chipset which is currently unsupported by the Zaptel wcfxo module. However, it's actually quite similar to the chipset used in the x100p modems, so I hacked the Zaptel wcfco.c module and got it working reasonably well. I guess other people might be running into this problem, so here's a link to my patch (against wcfxo.c in Zaptel version 1.4.9.2): http://www.fileqube.com/shared/MLnfrhCy86349 I haven't tried it, but I hope this might help to get other modems working (such as the Intel 537EP).
michael999 () at 2008-08-22 14:04 GMT

Voip-Info: Asterisk Jabber / Re: Sending caller ID as a message? [ID: 57735]

Fri, 22 Aug 2008 10:20:14 +0000

Of course, that's easy in itself. My particular problem is more "can I send to one or two people depending on the extension being called, and send a supplementary message if, say, a message is left?".
pmrtvcom (Phil Reynolds) at 2008-08-22 10:20 GMT

Tom Keating: Meeting with Digium

Fri, 22 Aug 2008 04:10:02 +0000

I had a very interesting meeting with Digium today in Huntsville, AL. Just got back to my hotel and fly back tomorrow. Stay tuned for more on what I learned.

Sent from my Windows Mobile� phone.

Tags: alabama, digium, Huntville, meeting

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Interview With Asterisk Founder, Mark Spencer - Jun 26, 2008
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Rev B of Astfin's BRI (ISDN) Asterisk Appliance Arrives - Apr 29, 2008

TrackBacks | Comments | Tag with del.icio.us | VoIP & Gadgets Blog Home | Permalink: Meeting with Digium

Voip-Info: Asterisk setup success 2 / T400P board with T1 [ID: 57727]

Thu, 21 Aug 2008 23:54:41 +0000

I have a question for you. What about echo cancellation on this setup? Was it provided by the T400P's or by software? Was it good?
robsonfelix () at 2008-08-21 23:54 GMT

Voip-Info: / Making a new (AT&T) prompt set for FreeSwitch [ID: 57722]

Thu, 21 Aug 2008 18:54:10 +0000

Hi All - I'm a producer in LA and a VOIP enthusiast and am making a open source drop in replacement prompt set for asterisk using the official voice of AT&T (Pat Fleet). She has agreed to do it and release as a contribution open source. It's pretty cool because she has been the voice of the phone company for almost 30 years - even with what I have so far, it sounds very carrier grade! I recorded her last week doing the entire core and extra sets for Asterisk and am now making and trimming the prompt files. As long as I'm at it, I would also like to make a set that works with FreeSwitch in all bit rates, but I can't find any reference file that indicates what text goes with what file names. Can anyone help me? Thanks, Alex
lexlex () at 2008-08-21 18:54 GMT

Voip-Info: / Asterisk & ISDN settings help [ID: 57695]

Wed, 20 Aug 2008 23:12:18 +0000

hello, thanks for your help. i have an Asterisk server (trixbox 2.6 ce), and i have to configure to work with an ISDN E1. a heard that i must change the Zaptel.conf and Zapata.conf, but how?... my interface is a Sangoma 4xE1 PCI card. somebody could help me?, i appreciate if someone give me some manuals or guide or instructions... thank a lot.
SirGerard (Gerardo Chavez) at 2008-08-20 23:12 GMT

Tom Keating: Internet Finally Takes Off into the Clouds

Wed, 20 Aug 2008 22:42:59 +0000

American Airlines began offering Internet access on long-haul domestic flights today, making American the first U.S. airline to offer full in-flight broadband, according to a Reuters report.

The world's largest airline said its passengers on Boeing 767-200 aircraft can pay $12.95 for Internet access on nonstop flights between New York and San Francisco, New York and Los Angeles, and New York and Miami.

Airlines have been racing to get reliable Internet access on their flights in hopes of gaining a competitive edge in the troubled airline industry. (And making some money since the cost of fuel is sucking the life out of the industry.)

Other carriers like Southwest Airlines and JetBlue have made strides in that direction. (When you charge for everything -- why not?)

Delta said this month it soon will start rolling out broadband Wi-Fi access for its entire domestic mainline fleet of more than 330 planes.

Experts have said they expect in-flight Internet to become common in the industry.

American's mobile broadband service, which is called Gogo, is provided by AirCell. Each Gogo session includes full Internet access. Cell phone and VoIP are not available. (So you can't call the pilots directly ...)

Tags: aircell, american airlines, delta, gogo, internet, jetblue, reuters, southwest airlines, voip

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TrackBacks | Comments | Tag with del.icio.us | VoIP & Gadgets Blog Home | Permalink: Internet Finally Takes Off into the Clouds

Voip-Info: / Re: Automatic calling via Asterisk CLI [ID: 57693]

Wed, 20 Aug 2008 21:23:03 +0000

I think the following reference will provide you with an answer. I use a modified version with great success. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message
Verbbal (Tony Goodfellow) at 2008-08-20 21:23 GMT

Voip-Info: / Disconnect a call [ID: 57674]

Wed, 20 Aug 2008 09:52:59 +0000

Hi all, Is there any way in asterisk java to disconnect a call from manager eventlistener at a particular condition?
java_hunter () at 2008-08-20 09:52 GMT

Voip-Info: / Hardware specification for around 1000 concurrent SIP peers [ID: 57664]

Wed, 20 Aug 2008 03:18:22 +0000

Please help me to get the specification for the following: 1. What asterisk version we have to use (whether we can use the community version or we have to go for business edition) ? 2. How many concurrent SIP peers an asterisk server can handle considering the following features? 2.1 codec trans coding (g711, g729, gsm, etc.) 2.2 Conferencing 3. What is the hardware configuration required (shall we handle 1000 concurrent SIP peers in a single asterisk server)? 3.1 Please specify the number of servers. 3.2 Other equipments we have to use. 3.3 We may use Grandstream / Tenor for terminating the calls.
firoz1975 (Firoz) at 2008-08-20 03:18 GMT

Voip-Info: Asterisk config alarmreceiver.conf / Is this better than nothing? [ID: 57660]

Wed, 20 Aug 2008 00:57:26 +0000

Why is there just an example file? This file is included with the Asterisk distribution and it seems redundant to display it here. Better content would be to actually explain what this is used for similar to other .conf file pages. I propose that the file content be removed. Nothing is better than useless (for example, [http://www.voip-info.org/wiki/index.php?page=Asterisk+config+adsi.conf]).
ardentbiker () at 2008-08-20 00:57 GMT

Tom Keating: VoSKY Exchange Pro VISIP-EX SIP PBX-to-Skype gateway

Tue, 19 Aug 2008 17:53:20 +0000

VoSKY just announced the VoSKY Exchange Pro VISIP-EX, a SIP-to-Skype gateway product that bridges the Skype network with SIP-based IP-PBXs and SIP gateways. The VoSKY Exchange Pro VISIP-EX enables companies with SIP-enabled PBXs to add Skype trunks for inexpensive calling. This isn't the first SIP-to-Skype gateway, I've written about others here, here, here, and here, however, this is the first "Skype-certified", enterprise-grade PBX-to-Skype gateway for SIP networks.

The new VoSKY Exchange Pro VISIP-EX is based on the company's enterprise-grade Linux platform and connects an IP-PBX to Skype to dramatically reduce telecom costs up to 90% over traditional telco trunks according to VoSKY.

The VoSKY Exchange Pro VISIP-EX integrates with any existing IP-PBX through SIP trunk ports and adds up to 30 lines of Skype. Using aVoSKY Exchange gateway, employees can make and receive Skype calls from their regular office phone - no Skype software, computers, or headsets are required.

Further, VoSKY Exchange ensures that all inbound and outbound Skype calls are included in the company's Call Detail Records (CDR), which is important from a managerial perspective. VoSKY Exchange Pro VISIP-EX can take advantage of Skype's newly launched Unlimited World plans for ultra low cost international and long distance calls.

VoSKY explains, "Businesses can also enhance their online sales efforts by adding a Skype-powered click-to-call button on their e-commerce site, online support portal, or web banner, as well as expand market presence by adding a local number in any country where global SkypeIn is available." They added, "Furthermore, the VoSKY gateway lets telecommuters and traveling workers connect to the company PBX from wherever they are, allowing them to talk to the office for free or place low cost international calls. To add remote PBX access, companies only need to install a free Skype client on the employee's laptop or smart phone."

Tags: skype, VoIP, VoSKY, VoSKY Exchange, VoSKY Exchange Pro VISIP-EX

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TrackBacks | Comments | Tag with del.icio.us | VoIP & Gadgets Blog Home | Permalink: VoSKY Exchange Pro VISIP-EX SIP PBX-to-Skype gateway

Tom Keating: Digium AA50 1.2 Software Released

Tue, 19 Aug 2008 17:23:40 +0000

Yesterday, Digium released version 1.2 for the Digium Asterisk Appliance 50 (AA50). It's been awhile since I've heard any news about the AA50, so I'm glad to see Digium is still developing for the AA50 even with their new Switchvox-based AA60 appliance offering and their even newer AA300 & AA350 offerings. Actually, I discussed whether Digium's AA60, AA300, and AA350 would compete against their original AA50 Asterisk appliance, which might be worth a read. In a nutshell, I stated that they target different sized businesses. Still, there is some overlap, which is why I'm glad to see Digium is continue to develop for the AA50.

So anyway, what's new in 1.2? First this interesting tidbit for Polycom phone fans: "My Polycom 3XX IP phones display acted funny in the 1.1.x firmware releases. Can I have the original, 1.0.x, behavior back?" To which Digium responds, "Yes. In 1.2, on the Options menu, there's a setting called "Enable Idle Image Display." Turn this setting off, Apply Changes, and reboot your phone."

Also new in 1.2 is support for Internet Explorer 7 and numerous usability improvements in the GUI itself. A new System Status page shows "firmware revision, information about your trunks (Did my VoIP trunks register okay? Did they fail?), and information about your extensions (what's on your system, and where do they point?)."

One important new feature is on the Trunks page. The pre-configured VoIP service providers are now pulled dynamically from a Digium webservice, instead of being hardcoded into the firmware. Thus, the next time one of the pre-configured providers changes something in their required configuration, Digium can automatically address it without requiring users to download a new AA50 firmware.

Another important new feature is that the Outgoing Calling rules page now allows outbound call pattern matches, in addition to a trunk path, to also be sent to a local destination or out via a fail-over trunk if the primary trunk is unreachable.

All AA50 users that purchased their AA50 and registered their subscription before August 13, 2008 will receive an additional free year of subscription. So essentially, Digium is giving away the 1.2 upgrade for free to those who registered.

This is just the tip of the iceberg of what's new, so go check of the Digium blog post for more. Lastly, I'm actually headed down to Huntsville, Alabama to meet with the Digium crew at their headquarters to find out what Digium is up too lately. Hopefully, I'll have interesting news to report. Stay tuned!

Tags: AA50, Digium, Digium Asterisk Appliance 50, voip

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Leslie Conway Joins Digium - Jul 11, 2008
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Podcast Interview with Digium CEO Danny Windham - Apr 17, 2008
Hulk Smash Asterisk 1.6! - Apr 16, 2008
PhoneFromHere.com & Digium Ink 5 Year Deal - Apr 09, 2008
Asterisk USB Hub - Mar 31, 2008
Sangoma Technologies Offers Lifetime Warranty - Mar 18, 2008

TrackBacks | Comments | Tag with del.icio.us | VoIP & Gadgets Blog Home | Permalink: Digium AA50 1.2 Software Released

Comments on this Entry:

(Mark on Aug 24, 2008 9:28 AM) Can i retrieve the CDR from the AA50 version 1.2?

(Renald Pigeon on Aug 27, 2008 6:37 AM) AA50 Firmware 1.2

Voip-Info: / Block Inbound Numbers In Asterisk [ID: 57632]

Tue, 19 Aug 2008 10:01:05 +0000

Hi, Could someone please help me, I would like to block a certain number from contacting me, just a nuisance call.... I hae no idea what to do! Please can you tell me what to put and where! Thank you! Ch33ky
ch33ky_m0nk3y (Simon H) at 2008-08-19 10:01 GMT

Voip-Info: Asterisk Jabber / Sending caller ID as a message? [ID: 57594]

Mon, 18 Aug 2008 08:54:01 +0000

Is there a way I could send caller ID from Asterisk to the intended recipients of phone calls? Basically, I seek a message of the format: Call to ${EXTEN} from ${CALLERID(all)}. (the Jabber client can be set to show the time of the message anyway.)
pmrtvcom (Phil Reynolds) at 2008-08-18 08:54 GMT

Voip-Info: / Automatic calling via Asterisk CLI [ID: 57593]

Mon, 18 Aug 2008 08:23:18 +0000

Hey I would like to know if its possible to call via the asterisk CLI, and howto do it. The reason is I have some automation scripts i run on my system, compiling for instent. I want my asterisk to call my on my cell phone and play a pre-recorded messege that the compiling is complete, or something went wrong. Does any of you know if its possible? I use Perl to script. Best Regards Sune
sunebeermann () at 2008-08-18 08:23 GMT

Voip-Info: / *asttecs [ID: 57585]

Mon, 18 Aug 2008 06:56:50 +0000

* astTECS aim to revolutionize the telephony PBX market by enabling the open source PBX Asterisk to obtain a wide spread market acceptance through creation of a complete ecosystem to support the traditional PBX channels. * astTECS would focus on the supply chain aspects like reach, availability, packaging, promotion, and increased awareness. www.asttecs.com
asttecs () at 2008-08-18 06:56 GMT

Voip-Info: Asterisk cdr mysql / Importing from CSV records [ID: 57581]

Mon, 18 Aug 2008 04:57:47 +0000

This is a reposting of information in the CDR CSV page, but it might prove useful here. I've found that if you want to naively import the csv data in to MySQL using: echo "LOAD DATA LOCAL INFILE 'Master.csv' INTO TABLE cdr FIELDS TERMINATED BY ',' ENCLOSED BY '\"' LINES TERMINATED BY '\n'" | mysql asterisk; Then the fields won't line up if you use the cdr table provided above, so here's a slightly modified version: CREATE TABLE `cdr_csv` ( `accountcode` varchar(20) NOT NULL default '', `src` varchar(80) NOT NULL default '', `dst` varchar(80) NOT NULL default '', `dcontext` varchar(80) NOT NULL default '', `clid` varchar(80) NOT NULL default '', `channel` varchar(80) NOT NULL default '', `dstchannel` varchar(80) NOT NULL default '', `lastapp` varchar(80) NOT NULL default '', `lastdata` varchar(80) NOT NULL default '', `start` datetime NOT NULL default '0000-00-00 00:00:00', `answer` datetime NOT NULL default '0000-00-00 00:00:00', `end` datetime NOT NULL default '0000-00-00 00:00:00', `duration` int(11) NOT NULL default '0', `billsec` int(11) NOT NULL default '0', `disposition` varchar(45) NOT NULL default '', `amaflags` int(11) NOT NULL default '0' ); echo "LOAD DATA LOCAL INFILE 'Master.csv' INTO TABLE cdr_csv FIELDS TERMINATED BY ',' ENCLOSED BY '\"' LINES TERMINATED BY '\n'" | mysql asterisk;
pscrane (Paul Crane) at 2008-08-18 04:57 GMT

Voip-Info: Asterisk CDR csv mysql import / Another Hint [ID: 57577]

Mon, 18 Aug 2008 04:44:11 +0000

I've found that if you want to naively import the csv data in to MySQL using: echo "LOAD DATA LOCAL INFILE 'Master.csv' INTO TABLE cdr FIELDS TERMINATED BY ',' ENCLOSED BY '\"' LINES TERMINATED BY '\n'" | mysql asterisk; Then the fields won't line up if you use the cdr table provided in the mysql secton, so here's a slightly modified version: CREATE TABLE `cdr_csv` ( `accountcode` varchar(20) NOT NULL default '', `src` varchar(80) NOT NULL default '', `dst` varchar(80) NOT NULL default '', `dcontext` varchar(80) NOT NULL default '', `clid` varchar(80) NOT NULL default '', `channel` varchar(80) NOT NULL default '', `dstchannel` varchar(80) NOT NULL default '', `lastapp` varchar(80) NOT NULL default '', `lastdata` varchar(80) NOT NULL default '', `start` datetime NOT NULL default '0000-00-00 00:00:00', `answer` datetime NOT NULL default '0000-00-00 00:00:00', `end` datetime NOT NULL default '0000-00-00 00:00:00', `duration` int(11) NOT NULL default '0', `billsec` int(11) NOT NULL default '0', `disposition` varchar(45) NOT NULL default '', `amaflags` int(11) NOT NULL default '0' ); echo "LOAD DATA LOCAL INFILE 'Master.csv' INTO TABLE cdr_csv FIELDS TERMINATED BY ',' ENCLOSED BY '\"' LINES TERMINATED BY '\n'" | mysql asterisk;
pscrane (Paul Crane) at 2008-08-18 04:44 GMT

Voip-Info: / Kindness from strangers [ID: 57521]

Sun, 17 Aug 2008 02:32:38 +0000

I am a true noob in all things VOIP, which means that I know as little about FreeSwitch as I do about Asterisk. I am interested in setting up a virtual PBX for my law office, and in integrating the VOIP system with the other IT systems such as contact, workflow, document and project management. I thought I had no choice besides Asterisk, but everything I have read about FreeSwitch convinces me that it is a better building block. However, FreeSwitch is a much newer product, and so I have fewer Internet-based resources to draw upon. Is there anyone out the there who can consult with me (for a fee) on what I have in mind? Thanks.
vhastings () at 2008-08-17 02:32 GMT

Voip-Info: VOIP Jobs / SIP Engineers needed in bayarea, CA [ID: 57489]

Sat, 16 Aug 2008 01:41:03 +0000

We are in need of a mid to senior SIP engineer with the following skills for an ongoing project. -Extensive work experience in SIP based platforms. -Work experience around FreeSWITCH. -Good experience in application development, integration, and troubleshooting. -Familiarity with Asterisk, LDAP, Linux. - Familiarity with basic ISP issues(i.e. routing, DNS, email servers, and web servers) is preferrable. Job location would be Fremont, CA for a 6 months+ contract duration. We are also open to hiring suitable candidates fulltime. Multiple positions open - one junior position, 2 mid to senior positions. To apply, please send me a word copy of your resume along with your daytime contact number at your earliest. To know more about us, please visit us at www.etouch.net or www.etouchconsulting.com Thanks, Kamala Gopalan, eTouch Systems Corp, Tel: (510)795-4800 X108 Fax: (510)795-4803 Email: kgopalan@etouch.net
kgopalan () at 2008-08-16 01:41 GMT

Voip-Info: Asterisk tips 911 / However... [ID: 57485]

Fri, 15 Aug 2008 23:25:36 +0000

.... I have discovered that if a call comes in on the EMERGENCY_TRUNK, the far end is not forced off and can hold it open. Any advice on how to avoid this would be appreciated.
pmrtvcom (Phil Reynolds) at 2008-08-15 23:25 GMT

Voip-Info: Asterisk RealTime Extensions / Re: Only ONE context in a RealTime Extensions ??? [ID: 57458]

Fri, 15 Aug 2008 00:42:35 +0000

Hi, Anyone by chance get an answer to whether "new" contexts can be created in the Realtime extensions table? Todd
the_tfulton () at 2008-08-15 00:42 GMT