Planet Asterisk

My office is closed entiry july, and according to the example above, you need to include both "open" and "closed". In that case, it would look something like this: include => inbound-closed|*|*|*|jul include => inbound-open|*|*|*|aug include => inbound-open|*|*|*|sep etc etc Isn't there some other way to do it? like: include => inbound-closed|*|*|*|jul include => inbound-open|*|*|*|!jul Would have been nice.
forsen (Erik Haider Forsen) at 2008-05-09 13:29

May 09, 2008 08:30 AM

The steady pace of computer technology marches on ...

According to a report on Engadget (here), Dell is set to launch a series of new Inspiron notebooks.

Having had an Inspiron 700m for what sees like ages now, feel good that Dell is coming off of its battery woes, but only time (and the gadget-buying public) will prove that right ...

And that may be soon, with the first one apparently coming right around Memorial Day.

Tags: Dell, Inspiron, Inspiron 700m, Memorial Day

Related Entries
• Dell picks Fonality VoIP Phone System - Jan 23, 2008
• Dell Gets into VoIP Game with OCS 2007 - Oct 16, 2007
• Memorial Day: Gadget Day - May 29, 2006
• Lenovo & ThinkPad Shaking Up Notebook World - Dec 29, 2005

May 08, 2008 09:26 PM

Matthew Fredrickson has posted details on which version of libpri to use with Asterisk 1.6

May 08, 2008 07:31 PM

He, I wold like to get info about load balancing of asterisk with UltraM. Could you please send me some info about experience on it...? Regards.
wturra (Waldino Turra) at 2008-05-08 17:21

May 08, 2008 05:21 PM

Om over at GigaOm writes that Jangl was looking to sell itself earlier this week and is "headed towards an ignominious end" and adds that Talkplus "is going nowhere fast". I wrote a detailed write-up on TalkPlus, interviewing TalkPlus CEO Jeff Black at ITEXPO and my biggest fascination was how the supposedly reverse engineered Skype. I wrote, "One final interesting thing we talked about at ITEXPO is that TalkPlus has built their own Skype gateway. In fact, when pressed further, Jeff mentioned they actually reverse engineered Skype's protocol. Although the Skype gateway isn't part of TalkPlus's launch today, Jeff explained that they have tested it in their labs and it's working very well."

The TalkPlus Java application was designed to allow you to view the presence of your Skype buddies, initiate a Skype call or even receive a Skype call. It even has SIP support with the ability to support other IM/VoIP clients such as Google Talk, AIM, MSN Messenger, etc.

It was cool technology, but perhaps a bit ahead of its time. One of the biggest features espoused by Jeff Black was "throwaway numbers". As I wrote previously, you can ditch a number if need be. Doctors or lawyers that are calling patients/clients can use their personal mobile phone and yet have their office CallerID number appear to the client/patient instead of their personal mobile number. In essence, this is a form of "CallerID spoofing", often a popular tactic used by hackers utilizing the Asterisk platform.

Does the fall of Jangl and if Om is correct, the fall of TalkPlus spell trouble for VoIP/Voice 2.0 applications? Maybe we're not ready for VoIP applications. Maybe we just want cheap minutes. Maybe "it's the cheap voice, stupid!" after all. Hmph... VoIP just got boring.

Tags: gigaom.com, Jangl, Jeff Black, Om Malik, TalkPlus, Voice 2.0, VoIP, VoIP 2.0

Related Entries
• Google buys Grand Central? - Jun 25, 2007
• SightSpeed Light Flash Video Conferencing & Chat - Jan 07, 2008
• FonCloud new Voice 2.0 app? - Sep 26, 2007
• PhoneGnome wonders - Where's the Voice 2.0 developers? - Feb 01, 2007
• Covad Vonage Deal Redux - May 08, 2008
• South Korea Blocks VoIP - here we go again... - May 08, 2008
• Covad to offer Vonage Broadband - May 08, 2008
• Packet8 Drops International Calling Phone Rates - May 07, 2008
• O-FONE Supports SIP on Symbian S60 mobile phone - May 07, 2008
• Telcos Join Forces to Battle Skype? - May 06, 2008

May 08, 2008 05:18 PM

Hi all, I bought a Cisco 7941g a few months ago. Uploaded SIP 8.3.1 firmware with windows tftp32 server, generated a SEPxxx.xml file and the phone worked very good!!!. I used it a long time with Xorcom Rapid appliance. MWI worked perfectly. A week ago I moved to trixbox 2.6 and problems started... Endpoint manager was really a disaster, I have to provision the 7941g manually, with my own generated SEPxxx.xml file. Anyway, MWI didn't work, nor using buggymwi=yes. I started to hate trixbox just a little... Well, I decided to downgrade 7941g to SIP 8.0.2 firmware, I heard this fw works good with MWI. Downgrade failed, and now I have a very nice and expensive brick!!!!!!!!!!!!! The phone ask for a DHCP address, when it receive one, it just freezes. No way on making the phone to ask for the new firmware. I've tried with tftp option 66 and option 150... I did factory default a lot of times, but the same happens.............. :::::(((((((((( Any ideas on how to make it back to life??
vpreatoni (Victor Preatoni) at 2008-05-08 05:19

May 08, 2008 05:19 AM

The Asterisk development team has released Asterisk version 1.4.20-rc2.

May 08, 2008 01:56 AM

AstriCon is the pioneer and longest-running event devoted to all things Asterisk, one of the most influential open source projects today. Attendees will learn about trends in Asterisk use, the growing Asterisk ecosystem, the newest applications and a wide range of technical topics from Asterisk developers, users and entrepreneurs.

“AstriCon is so vibrant and productive because it brings together the people who live and breathe telephony innovation,” said Mark Spencer, creator of Asterisk and Digium’s chief technology officer. “This is the one time in 2008 when the individuals who are most focused on development and use of Asterisk will come together, making it the show that both enthusiasts and those who are looking to get to know Asterisk must attend.”

In the past, many Asterisk users and Digium partners needed to select one Asterisk event to attend in the fall—AstriCon or Digium|Asterisk World, which has been held in conjunction with Pulvermedia’s VON Conference & Expo. However, due to recent changes at Pulvermedia, Digium will not hold Digium|Asterisk World this fall. As a result, the many ISVs, resellers and integrators, and developers who selected one over the other in the past, or needed to split attention and resources, will be able to focus on AstriCon.

AstriCon 2008 will be held at the Renaissance Glendale Hotel and Spa in Glendale, Ariz. Those wishing to submit speaking proposals may do so by June 1, 2008. Registration is now open at www.astricon.net.

May 07, 2008 08:25 PM

Rich received an email from O-FONE claiming they are the "First s60 SIP Symbian client". I'm not so sure about being the "first" since Truphone has a SIP client for Symbian phones, so do Jajah & Fring -- and there are others I've come across as well. Buzz2talk, from Indtelesoft had a SIP client for Symbian phones with PTT (push-to-talk) functionality way back in 2004. Alas, when I try and go to Indtelesoft's website, it appears to be defunct.

Well they may not be "first", but I'll let that marketing faux paux pass and I'll share the news with you Symbian mobile phone fans so you can have another choice to "get your SIP on". The S60 support means you can use this SIP client on several Nokia phone models such as the Nokia E70, Nokia N81, Nokia N96, and more. Several Samsung phones such as the Samsung SGH-G810 and the Samsung SGH-i560 should also work.

Simply download & install O-FONE's free software to your phone and, as long as you have WiFi or 3G cellular data access, you can use O-FONE's low cost call termination. According to their website, "Calling other people using O-FONE is FREE (with Global plan only)". I'm going to assume "with Global only" means you have to be a paying customer. This seems a bit restrictive to me. AFAIK competitors such as Jajah grant you free calling to fellow users just by being a registered user and without forcing you to be a paying (Global plan) customer.

Nevertheless, having a SIP-enabled client native to the Symbian S60 phone is quite useful, especially if you make a lot of international phone calls. Go check them out.

Tags: mobile phone, Nokia, O-FONE, o-fone.com, S60, Samsung, SIP, Symbian, VoIP

Related Entries
• Talkonaut 4.0 native Symbian S60 edition released - Feb 28, 2008
• D2 Technologies mCUE mobile convergence softphone launches - Jan 07, 2008
• fring coming to Windows Mobile phones - Jun 19, 2007
• TringMe Now Works with AOL/AIM - Apr 30, 2008
• Voxeo Adds VoIP Party Line App to Facebook - Apr 22, 2008
• Microsoft Response Point Announces SIP Trunking Service Providers - Apr 18, 2008
• PhoneFromHere.com & Digium Ink 5 Year Deal - Apr 09, 2008
• C2Call - New browser-based Java VoIP app - Apr 03, 2008
• JAJAH Mobile VoIP client for the iPhone - Apr 02, 2008
• Skype and other VoIP Apps on Mobile Phones - Apr 01, 2008

May 07, 2008 02:58 PM

AOL Math: 1.7 + .3 = 4   AOL has taken a page from Ma Bell in terms of creative mathematics. With each call, AOL first rounds UP the time of the call to the next minute and then rounds UP the total price to the next penny. Here’s the way their Terms of Service describe it:

“For point of clarity, the rounded up minutes are multiplied against the current rate effective at the end of the call (generally based on the location the call is placed) and then rounded up to whole cents (USD).” So a 70-second call in the U.S. (which should cost under 2¢ at 1.7¢ per minute using Plain Old Math) actually is billed to you at 4¢. Charitably speaking,

it’s creative to advertise the cost of a call in the U.S. as 1.7¢ per minute with all the rounding that is taking place. For short calls, it can be more than double that rate once you factor in AOL’s double rounding. In our example, the 70-second call is first rounded up to 2 minutes. And then the cost of the call is computed at 3.4¢ for the already rounded up call. Then the 3.4¢ computation is rounded up to 4¢. So you see 1.7 + .3 really does equal 4 in the bowels of AOL.

Source:  Click Here for the Full Nerd 

May 06, 2008 06:48 PM

I have probably spent 20 hours trying all variations on the above. There are too many holes in the CCM configuration because we can not see the images. Thus we can not tell what may be pertinent in the various CCM screens. If I follow what I believe the instructions tell me, I believe the RDNIS is not working as the rule seems to skip the 2 step in favor of going to 400 step (GoToIF command). Thus the voice prompts me to enter my extension and a password, which I believe is the 400 line. I am told that the RDNIS command has changed and is now CALLERID(rdnis) in version 1.4. Nothing I have tried will get my CCM 4.x to pass the RDNIS Redirecting Number (RGN) that this solution seems to require. This may be because I am simply not seeing something here that I need, but I must now give up integrating CCM 4.x to Asterisk 1.4 until such time that someone with far more knowledge of this steps in and figures it out. -- Steve
smhickel (Steve Hickel) at 2008-05-06 14:46

May 06, 2008 02:46 PM

I have probably spent 20 hours trying all variations on the above. There are too many holes in the CCM configuration because we can not see the images. Thus we can not tell what may be pertinent in the various CCM screens. If I follow what I believe the instructions tell me, I believe the RDNIS is not working as the rule seems to skip the 2 step in favor of going to 400 step (GoToIF command). Thus the voice prompts me to enter my extension and a password, which I believe is the 400 line. I am told that the RDNIS command has changed and is now CALLERID(rdnis) in version 1.4. Nothing I have tried will get my CCM 4.x to pass the RDNIS Redirecting Number (RGN) that this solution seems to require. This may be because I am simply not seeing something here that I need, but I must now give up integrating CCM 4.x to Asterisk 1.4 until such time that someone with far more knowledge of this steps in and figures it out. -- Steve
smhickel (Steve Hickel) at 2008-05-06 14:44

May 06, 2008 02:44 PM

I have probably spent 20 hours trying all variations on the above. There are too many holes in the CCM configuration because we can not see the images. Thus we can not tell what may be pertinent in the various CCM screens. If I follow what I believe the instructions tell me, I believe the RDNIS is not working as the rule seems to skip the 2 step in favor of going to 400 step (GoToIF command). Thus the voice prompts me to enter my extension and a password, which I believe is the 400 line. I am told that the RDNIS command has changed and is now CALLERID(rdnis) in version 1.4. Nothing I have tried will get my CCM 4.x to pass the RDNIS Redirecting Number (RGN) that this solution seems to require. This may be because I am simply not seeing something here that I need, but I must now give up integrating CCM 4.x to Asterisk 1.4 until such time that someone with far more knowledge of this steps in and figures it out. -- Steve
smhickel (Steve Hickel) at 2008-05-06 14:44

May 06, 2008 02:44 PM

Our service works Asterisk / Sipura etc and their price is much cheaper than HKBN 2b too. why not go to our website and email us from there. 1 HK DID with 1 outgoing line for only $99 HKD per month.
telehk (Sam Tam) at 2008-05-06 10:42

May 06, 2008 10:42 AM

I found a bug when trying to get Asterisk working with Ultramonkey. After fixing this bug I have Asterisk working with Ultramonkey doing load balancing and heartbeat without any problem. The bug: Asterisk real server health check does not work reliabily from Ultramonkey. ldirectord from ultramonkey sends SIP OPTIONS request for real server health ckeck. Many a times Asterisk sends "200 OK" response for this request on a wrong port. So, the real server is deactivated. Here are the details: - Ultramonkey could set up to use SIP OPTIONS request for Asterisk real server health check. When you do that the script /etc/ha.d/resource.d/ldirectord uses the same call-id for all the OPTIONS requests it sends. - In Asterisk, in chan_sip.c, when it receives a new SIP request it tries to see if there is an existing dialog setup for this request. If it doesn't find the exising dialog it will setup the new dialog. Since call-id, to, from and Cseq are same for every request sent from ldirectord it sometimes picks up the wrong earlier dialog and sends the response to this request on the wrong port. - ldirectord never receives response in the above case and marks the real server down. Solution: Modify ldirectord to generate new call-id for each request. Here is the modified code for ldirectord. After this change there is no problem in real server health check. Here is a quick modififications to ldirectord, check_sip subroutine. You can use any method to generate different call-id. I have used the following method. my $range = 100000000000; my $callid = int(rand($range)); my $request = "OPTIONS sip:" . $$v{login} . " SIP/2.0\r\n" . "Via: SIP/2.0/UDP $sip_s_addr_str:$sip_s_port;" . "rport;" . "branch=z9hG4bKhjhs8ass877\r\n" . "Max-Forwards: 70\r\n" . "To: \r\n" . "From: ;tag=1928301774\r\n" . "Call-ID: $callid\r\n" . "CSeq: 63104 OPTIONS\r\n" . "Contact: \r\n" . "Accept: application/sdp\r\n" . "Content-Length: 0\r\n\r\n"; If anybody wants full details of how to get Asterisk working with ultramonkey load balancing and heartbeat let me know.
madhuri (Madhuri Patwardhan) at 2008-05-06 06:39

May 06, 2008 06:39 AM

I found a bug when trying to get Asterisk working with Ultramonkey. After fixing this bug I have Asterisk working with Ultramonkey doing load balancing and heartbeat without any problem. The bug: Asterisk real server health check does not work reliabily from Ultramonkey. ldirectord from ultramonkey sends SIP OPTIONS request for real server health ckeck. Many a times Asterisk sends "200 OK" response for this request on a wrong port. So, the real server is deactivated. Here are the details: - Ultramonkey could set up to use SIP OPTIONS request for Asterisk real server health check. When you do that the script /etc/ha.d/resource.d/ldirectord uses the same call-id for all the OPTIONS requests it sends. - In Asterisk, in chan_sip.c, when it receives a new SIP request it tries to see if there is an existing dialog setup for this request. If it doesn't find the exising dialog it will setup the new dialog. Since call-id, to, from and Cseq are same for every request sent from ldirectord it sometimes picks up the wrong earlier dialog and sends the response to this request on the wrong port. - ldirectord never receives response in the above case and marks the real server down. Solution: Modify ldirectord to generate new call-id for each request. Here is the modified code for ldirectord. After this change there is no problem in real server health check. Here is a quick modififications to ldirectord, check_sip subroutine. You can use any method to generate different call-id. I have used the following method. my $range = 100000000000; my $callid = int(rand($range)); my $request = "OPTIONS sip:" . $$v{login} . " SIP/2.0\r\n" . "Via: SIP/2.0/UDP $sip_s_addr_str:$sip_s_port;" . "rport;" . "branch=z9hG4bKhjhs8ass877\r\n" . "Max-Forwards: 70\r\n" . "To: \r\n" . "From: ;tag=1928301774\r\n" . "Call-ID: $callid\r\n" . "CSeq: 63104 OPTIONS\r\n" . "Contact: \r\n" . "Accept: application/sdp\r\n" . "Content-Length: 0\r\n\r\n"; If anybody wants full details of how to get Asterisk working with ultramonkey load balancing and heartbeat let me know.
madhuri (Madhuri Patwardhan) at 2008-05-06 06:37

May 06, 2008 06:37 AM

The Asterisk development team has released Asterisk version 1.4.20-rc1.

May 06, 2008 01:20 AM

The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1.

May 06, 2008 01:19 AM

May 05, 2008 07:24 PM

Today we introduce a SIP termination provider for Asterisk with a familiar name and competitive prices. The setup with Asterisk will take you under five minutes and is well worth the effort...

May 05, 2008 07:00 AM

I wrote down a small tutorial on my blog in which I describe how to install a fritz card on trixbox. Here it is: [http://www.ivanoiu.com/avm-isdn-fritz-card-pci-on-trixbox-asterisk/]
ivanoiu (Andrei Ivanoiu) at 2008-05-04 16:10

May 04, 2008 04:10 PM

I too have compiled and make installed but have the not found issue. At the Asterisk CLI, if I tab out: module load app_ it finds app_flite.so and completes the line so it is looking in the correct directory. If I press enter I then get... [May 4 14:01:24] WARNING[17671]: loader.c:644 load_resource: Module 'app_flite.so' could not be loaded. Jason
jbassett (Jason Bassett) at 2008-05-04 12:57

May 04, 2008 12:57 PM

Digium Partner BRI/FRA/PRI/ANALOGUE Installations all over Ireland Plan install and support PABX systems of all sizes We supply all hardware ( HP , AASTRA ,) CONTACT : Brendan Redmond asterisk@nssl.ie
nssl () at 2008-05-03 20:51

May 03, 2008 08:51 PM

Hi, I have the same problem Jeremy has, where after going through the procedure Flite doesn't show as registered in Asterisk. I'm using 1.4.11. My printouts are exactly like his. Has anyone found a work around? Thanks!
rbdnz (Rob) at 2008-05-02 22:26

May 02, 2008 10:26 PM

May 02, 2008 06:34 PM

May 01, 2008 08:23 PM

I'm reticent to spread rumors about a VoIP company's demise - I certainly don't want to be be the Valleywag of VoIP blogs. However, Rich Tehrani, my boss, has been hearing rumors about ShoreTel and it doesn't sound good. It goes without saying that Rich Tehrani has his "ear to the ground" in the VoIP industry perhaps more than anyone - myself included. Rich heard things about TMC's main competitor long before the story broke out in the blogosphere, but both Rich and the TMC team took the "high road" and chose not to write damaging stories about our competitor.

With this in mind, Rich wrote in his blog, "I have never heard the kind of rumors about a vendor that I am hearing about Shoretel. Reports of unhappiness in the ranks of the workers and management problems persist in the industry."

Rich adds, "I normally hate to talk about rumors without doing more research than this but in my experience this level of negativity is unusual for any company and for it to come out of the blue and unsolicited from multiple sources means there could likely be fire causing the smoke."

Greg Galitzine also weighs in when he writes, "In my opinion these guys were once the truest darlings of the VoIP world. Maybe I was simply a sucker for a great desk set (and they had some nice hardware, I tell you) but in the wake of all the sour news in the VoIP world I have to wonder if the old adage “where there’s smoke, there’s fire” applies."

Greg adds, "Shoretel stock (SHOR) was down 10.10 percent on Wednesday, but appeared to be making a huge run in aftermarket trading, bouncing back about 8.5% at 6:50 pm ET."

ShoreTel is one of the early IP-PBX pioneers and they have some really great technology. I for one would be sad to hear if the rumors of their demise are indeed true. While it could be fun for me to speculate that pressure from low-cost IP-PBX solutions, especially Asterisk, the open source IP-PBX is affecting ShoreTel, I don't think this is the case. The last time I looked at ShoreTel, their technology was more scalable than any of the Asterisk-based solutions out there. Asterisk's sweet-spot is really under 80 seats (some exceptions excluded) where as ShoreTel's sweet spot is 80 seats and up. ShoreTel competes more with Cisco, Nortel, and Avaya.

So why is ShoreTel having problems? As Rich wrote in July 2007, "ShoreTel hit it big with an IPO which jumped 27.5% on it’s opening day. The IPO was derailed last week as a result of a lawsuit filed by Mitel. The company closed the day at $12.15, after the company priced 7.9 million shares at $9.50 a share." So perhaps Mitel's lawsuit against ShoreTel took the steam out of ShoreTel?

Perhaps all the recent lawsuits against various VoIP companies, including Vonage, which has lawsuits filed against them by Verizon, Sprint Nextel, Nortel, and even me over my cold pizza has stymied ShoreTel?

Tags: Asterisk, Cisco, Greg Galitzine, IP-PBX, Mitel, open source, Rich Tehrani, ShoreTel, VoIP

Related Entries
• Mitel sues ShoreTel - What does it mean? - Jun 28, 2007
• Asterisk 1.6.0 beta5 - Mar 06, 2008
• Digium launches Switchvox Free Edition - Oct 31, 2007
• Avaya #1 VoIP PBX Vendor? - Aug 21, 2007
• Predictive Dialing on Asterisk? - Jun 13, 2007
• Digium - The Showstopper! - May 18, 2007
• pbxnsip IP-PBX Review - Feb 21, 2007
• Digium podcast of Mark Spencer's new role - Jan 30, 2007
• Rev B of Astfin's BRI (ISDN) Asterisk Appliance Arrives - Apr 29, 2008
• Hulk Smash Asterisk 1.6! - Apr 16, 2008

Comments on this Entry:

(Afraid of the comments much? on May 6, 2008 7:37 PM) Is there a reason why you are not posting the comments? Isn't that what a blog is? People are outraged at your irresponsible entries as they couldn't be farther from the truth. I assume by now you are well aware of this fact as many, many people are responding and neither you nor Rich has been brave enough to post the responses - which contain the "truth" and not just some weird "rumors" that likely started at a company that is threatened by ShoreTel's success.

(Tom Keating on May 7, 2008 10:17 AM) >>neither you nor Rich has been brave enough to post the responses I have no idea what you're talking about. I post all comments, including yours. I do moderate comments to prevent spam and inappropriate content. I absolutely welcome comments to set the record straight. Also, your comment was the first comment to this post. As for Rich, he was traveling and he moderates his comments as well, so it might be a day or so for a comment to appear. As I wrote above, I like ShoreTel's technology. If you have information about them that points to them continuing to grow that you'd like to share, I'm all ears.

May 01, 2008 03:18 PM

Seemingly SIPAddHeader() does not work together with SendText() in Asterisk 1.4.18.1.
mdornsief (Maximillian Dornseif) at 2008-05-01 12:20

May 01, 2008 12:20 PM

I had typed like example but I had an error is: Permisson denied! Help me, please, as soon as possible! Thanks
aivanho (Dao Quang Minh) at 2008-05-01 08:43

May 01, 2008 08:43 AM

Well, it certainly didn't take long for someone to leverage AOL's Open Voice APIs featuring full SIP support. TringMe is apparently the 'first' out of the gate announcing support  to millions of AIM call-out users to make calls directly from TringMe. AIM users can use TringPhone - TringMe's fully web-based SIP phone to make calls directly from the browser.

According to TringMe, "with TringMe’s MobileVoIP solution, AOL/AIM users can use a mobile device (e.g. Symbian devices) to make VoIP calls from a mobile device. If they desire to use Gtalk to make VoIP calls over this service, that’s supported too. In general, AOL/AIM users can use any of TringMe’s supported originating devices to make calls (say Gtalk)."

Apparently it's pretty easy to setup. You just login to your TringMe account, enter your AIM or AOL screen-name (AOLScreenName@aol.com or AIMScreenName@aim.com) and SIP password in the TringPhone settings. Make sure to specify “AOL” in the Domain or Proxy setting as well. That’s it! Now you can use TringPhone for making calls through your AIM call-out account.

Via TringMe blog

Update: 11pm. Figured it was worth sharing some thoughts from around the VoIP blogosphere on the AOL Open Voice API news. Obviously, the main gist of the news is that you can now use any SIP-based device and register it with the AOL SIP registrar. TringMe was first to promote succesfully doing this. I thought about trying to register one of my spare Aastra phones or an X-Lite client, but Dan beat me to do it by getting X-Lite to register on AOL.

http://www.disruptivetelephony.com/2008/04/aol-launches-op.html
Dan York gives some good details on how the AOL Open Voice APIs allow you to use any SIP client, such as X-Lite. He also writes "Does accepting SIP connections at your SIP proxy constitute an "API"? Does providing SIP termination services to the PSTN constitute an "API"?" He has a valid point. I thought about that myself, but assumed there was an API in addition to the SIP support, which again TringMe was so quick to leverage.

Alec Saunders has a Squawk Box on the news and also emails fellow VoIP bloggers hinting the news is a non-event, while simultaneously discussing AOL's starts & stops in the VoIP space which directly affected Alec's company.
http://saunderslog.com/2008/04/30/squawk-box-april-30-trust/

Mr. Blog responds, "I agree that it is not an API, by any stretch. But I disagree that it is a non-event"
http://mrblog.org/2008/04/30/aol-open-voice-program-works-with-phonegnome/

One last thought on my end... I'm glad AOL is opening up their network to allow any SIP device to connect. I wish Skype would be so open. Heck, I wish Vonage, which is SIP-based would allow you to have open SIP credentials and use any SIP device. Whether this is enough to get people to switch from using SkypeOut minutes to AOL's PSTN termination remains to be seen. In theory, I can configure an Asterisk server to use AOL as an ITSP. But then AOL just comes another SIP termination service provider, which are a dime-a-dozen these days.

Tags: AIM, AOL, SIP, tringme, TringPhone, VoIP

Related Entries
• AOL Launches Open Voice APIs - Apr 29, 2008
• TringMe opens platform API to developers - Apr 16, 2008
• tringme TringPhone Flash VoIP App Launches - Dec 03, 2007
• AIM Call Out Update - Oct 29, 2007
• AIM Call Out Launches - Oct 26, 2007
• O-FONE Supports SIP on Symbian S60 mobile phone - May 07, 2008
• Voxeo Adds VoIP Party Line App to Facebook - Apr 22, 2008
• Microsoft Response Point Announces SIP Trunking Service Providers - Apr 18, 2008
• PhoneFromHere.com & Digium Ink 5 Year Deal - Apr 09, 2008
• C2Call - New browser-based Java VoIP app - Apr 03, 2008

Comments on this Entry:

(Jack on Apr 30, 2008 6:51 PM) Yes it defenitely did not take very long for tringme to provide support for AIM Call Out. I just made an International call & it works great. Thanks for that update tom. http://journals.aol.com/opensip/aol-open-voice-platform

(Randell Jesup on May 1, 2008 5:49 PM) This is far from an open SIP server - no registration, no direct IP calls, no inbound calls at all. It is, simply, an (open) outbound gateway, and users need to have AOL/AIM accounts with money. Useful, but not exciting.

April 30, 2008 10:39 PM

To include global variables from another file, remove any [globals] heading from that file. Then in the [globals] section of extensions.conf, add #include my-other-file-with-just-globals.conf
cmaj () at 2008-04-30 18:05

April 30, 2008 06:05 PM

In the world of business, partnerships can make or break one’s endeavors. More often than not, entities come together to complement each other’s strengths. In doing so, they are able to provide better and more services and grow their businesses.

(more…)

April 30, 2008 05:03 PM

While researching for my VoIP call screening post earlier today, I came across a patent for "call screening". Curious, I decided to check it out the filed patent and found it was filed by Nortel Networks in July 2004. Interestingly, the patent application agrees with me that hosted voicemail needs call screening and goes as far as to say service providers with hosted voicemail systems are at a competitive disadvantage when they don't have call screening. The patent app reads as follows:

The present invention relates to telephony communications, and in particular to allowing a user to screen calls by listening to a voicemail message being left in a hosted voicemail system from a telephone device.

Background of the Invention [0002] Telephone users with personal telephone answering devices can listen to callers leaving messages thereon, and during the call, decide to take the call. This highly desirable technique for screening calls is unavailable in hosted voicemail systems, because the voicemail system is a separate entity in the telephone network and is not directly associated with any individual's telephone device. In a hosted voicemail system, incoming calls that are not answered are forwarded to the voicemail system. Since many users, especially residential users, rely on the ability to screen calls, service providers with hosted voicemail systems are at a competitive disadvantage when trying to market hosted voicemail services to their subscribers.

Accordingly, there is a need to provide call screening for users subscribing to hosted voicemail services.

Yeah, no kidding 'call screening' is a desirable feature! What's even more interesting is the patent diagram makes no mention of VoIP, as seen here:

The text of the patent itself makes no mention of VoIP, so this is strictly traditional PSTN hosted voicemail call screening. Earlier today I griped about the lack of real-time call screening in VoIP services, such as Vonage & Packet8. It's not that hard, especially if using software. You should be able to easily setup a 3-way conference call via SIP to enable call screening. That is, one leg is the caller, the second leg is your phone, and the last leg is special call screening software running on your PC. You just send a SIP Invite to the PC, have the software auto-accept the SIP invite and connect legs #2 & #3. If the user accepts the call, simply connect/conference leg #1.

Now most people don't want to perform call screening on their PC. Most would want to do it via the phone, especially since the PC may not be on or nearby. Well, that's easy enough as well. First, you ring the user's phone, then after X number of rings, the phone stops ringing and the hosted voicemail system prompts the caller to leave a message. Simultaneously, the hosted voicemail system calls the phone again (via another SIP Invite) and this time tells the phone to play a special ringtone to indicate a caller is leaving a message. Hearing the special ringtone, the user can pick up the phone, be conferenced into the voicemail message being left (with mic muted) and if the user presses a touch-tone they can instantly pull the caller out and their mic is unmuted. Simple!

Damn, between the "special" ringtone to indicate the opportunity to screen the caller and the "instant" ability to pull a caller out of a hosted VoIP voicemail system, I should patent these! The special ringtone can even work with analog phones connected to ATAs simply by varying the length of the ring voltage, i.e. two super-quick rings.

Now while I didn't go crazy searching the patent database, I did look around and didn't see a patent for "VoIP call screening". Hmmm. Very interesting... Ok you patent trolls, here's an opportunity for you. File a patent for "VoIP call screening". In fact, file one for "hosted VoIP call screening" and another one for "customer premise VoIP call screening" that works on customer premise Voice over IP phone systems (IP-PBXs), such as Asterisk. Then when you rake in millions from patent extortion, just make sure you show me some lovin'. After all, I did give you the idea. If you don't show me some lovin' then may your guilty conscience eat you up. Oh wait, patent trolls with a guilty conscience? What in blazes am I thinking?

Update: (12:20pm)
I neglected to mention that there are some hosted voicemail providers offering call screening. They aren't necessarily VoIP or leveraging VoIP technology though. Some examples include CallWave, GrandCentral, and Ring Central. Figured they were still worth mentioning.

Tags: call screening, hosted VoIP call screening, Nortel Networks, patent, patent troll, VoIP

Related Entries
• VoIP Call Screening - Apr 30, 2008
• VoIP Pioneer Slams Sprint - Jan 25, 2008
• Another VoIP Provider Bites the Dust - Sep 26, 2007
• Vonage vs. Sprint - Court Decision Reached - Sep 25, 2007
• P2P or ENUM Saves Vonage? - May 11, 2007
• Vonage warns it could file for bankruptcy - Apr 18, 2007
• Vonage Patent Update - Apr 09, 2007
• Vonage NOT Saved by VoIP, Inc. - Apr 04, 2007
• Vonage inks deal with VoIP Inc bypassing Verizon patent claim - Apr 03, 2007
• Jangl & TalkPlus Spell VoIP Troubles Ahead? - May 08, 2008

Comments on this Entry:

(Ntwiga on May 2, 2008 12:29 PM) Would said patent be worthless since this web page effectively constitutes prior art?

(Colin Kelley on May 7, 2008 4:46 PM) In fact CallWave has holds a patent that relates to Call Screening on a hosted system including VOIP. 7,103,167

April 30, 2008 04:43 PM

Codeguys.rpc1.org is a popular website run by C0deKing and Kanalratte that offers crossflashing and overclocking of your DVD±RW drives using "hacked" firmware for your DVD burner. Often the hacked firmware enables missing features such as overclocking the speed of the DVD burn and more importantly, setting the 'book type' permanently to "DVD-ROM". (more on that later) RPC1.org also offer "autopatchers", which are are easy do-it-yourself tools with an easy-to-use GUI to patch DVD burner firmware.

I recently bought a Sony DRU-840A DVD burner and when I tried to burn a home movie I noticed Nero didn't list the booktype setting under the 'Options' button. (Here's a screenshot of the book type setting in Nero on my Vista PC with a different DVD burner:)


The Book Type setting, also knowns as "bitsetting" allows you to change DVD+R media's default book type of "DVD+R" & "DVD+RW" to "DVD-ROM" more more compatibility with home DVD players which are looking for this particular book type. The Book Type is four bits at the start of every DVD disc (in the physical format information section of the control data block), which indicates what the physical format of the disc is.Many DVD players will refuse to play burned DVDs without the proper book type. Changing the book type works on both single-layer (SL) and dual-layer (DL) DVD+R media but not on DVD-R or DVD-RW media. In other words, minus (-) is bad for compatibility and plus (+) is good.

So in any event, the reason I bought the Sony DRU-840a was because it was supposed to support bitsetting/book type. I had Nero 7 installed which should have recognized the drive as supporting this. So then I figured I may have had old firmware. I went to Sony's support page and couldn't find any new firmware for this drive. I knew there was "hacked" firmware out there and have gone to the RPC1.org website in the past to add "features" missing in my DVD burner. So I headed on over to codeguys.rpc1.org to see if I could find better firmware for my drive only to discover the site was down. I tried several days in a row and it was still down. The last Google cache is 5 days ago. Not good. Seems like a permanent outage to me. Wonder if they were shutdown for illegally distributing hacked firmware? Yeah well maybe if the original manufacturers didn't cripple the firmware users wouldn't resort to hacked firmware. Fortunately, after some creative Googling I found a RPC1 mirror here:
http://codeworks.cdfreaks.com/cgmirror/

I thought perhaps my Sony drive was a LiteOn OEM, since LiteOn is perhaps the largest OEM manufacturer of DVD drives that do not carry the LiteOn label and past Sony DVD burners I used were LiteOn. Generally speaking you can use "real" LiteOn firmwares with these so-called rebadged drives. But when I attempted to try the firmware loaders from the mirrored website, the utilities wouldn't recognize my Sony DVD burner.

I then figured out it was actually an OEM of the Samsung SH-202J DVD drive. I then tried the OmniPatcher utility which is supposed to support Samsung/Sony DVD burners, but it couldn't detect my drive. Back to square one.

I then said the hell with Nero 7, I'll just use ImgBurn, a popular freeware DVD burning software utility that has a very powerful book type setting utility that works with virtually any DVD burner. It even features an Advanced tab for configuring settings manually. I selected the 'Samsung' tab, and it gave me a warning "Unknown (Drive doesn't report it!)". I read you can just ignore this message as long as you are selecting the correct OEM company. I changed the drop-down box to 'DVD+R DL Media' and then changed the book type to "DVD-ROM". I then clicked 'Ok', and received a "Success!" message, as seen below.


As long as you get this "Success!" message you are good to go. In my experience with multiple burners, I find that you can try each of the tabs with the various models and try to change the book type. At worst you'll get a "Failed!" message. In any case, I burned a DVD and then verified the booktype was correctly set to DVD-ROM by clicking Drive Info in Nero.

Curious if the book type setting I set in ImgBurn would work in Nero, I then burned a second DVD and Nero correctly set the book type. So the bitsetting  change made by ImgBurn is global to all your DVD burning apps. Unfortunately, the setting isn't permanent. If you reboot your PC you have to go back into ImgBurn and change the book type setting again. Using "hacked" firmware would have saved me the trouble of doing this step.

Just when I was going to give up on Nero 7, I decided to see if any updates were available. I downloaded and installed the latest version from Nero 7 from Nero's website. I went back to the Options section in Nero and voila' the Book Type setting was there!

Here's proof:


Conclusion:
So you are having problems with the book type settings in your DVD burner, try these steps:

• Re-install Nero or other DVD burning software with the latest updates.
• Use ImgBurn instead to set the booktype. Here's a good tutorial with screenshots explaining how to do it.
• Go to the RPC1.org mirror and try crossflashing your DVD burner or installing "hacked" firmware. Be aware this will void your warranty and could break your burner!
• If none of these work, it's possible your drive doesn't support setting the book type. Time to go buy a new burner! Try Googling your DVD model + "book type" and see if any discussions are out there mentioning if it supports it.

Hope this info helps someone else since I wasted a couple hours trying to solve this book type setting problem.

Tags: autopatcher, book type, booktype, Codeguys.rpc1.org, crossflashing, DVD burner, DVD+R, DVD-ROM, DVD+RW, firmware, problem, rp1.org

Related Entries
• Apple to nuke unlocked iPhones - Sep 25, 2007
• Fix Scratched DVD problems - Dec 12, 2006
• Digium releases new firmware - Jul 28, 2005
• Vonage solves Clearwire blocking problem, asks feds for help - Apr 22, 2005
• Packet8 firmware download with new features - Jul 28, 2004

April 29, 2008 10:09 PM

What I can create new agents in my database, I need some table and some other configuration?, please help me
jmory (Juan Mory) at 2008-04-29 21:16

April 29, 2008 09:16 PM

What I can create new agents in my database, I need some table and some other configuration?, please help me
jmory (Juan Mory) at 2008-04-29 21:15

April 29, 2008 09:15 PM

You're cordially invited to take version 1.2 of our Leaner, Meaner Asterisk machine for a spin today with CentOS 5.1 (32-bit or 64-bit versions), Asterisk 1.4 or 1.6, FreePBX 2.4, Apache, MySQL, PHP, SendMail, Perl, Flite, and more. Free downloads for Linux and VMware are now available.

April 29, 2008 09:00 PM

Rev B  of Astfin's BRI (ISDN) Appliance has just arrived. The BR4 appliance (or BRI Appliance) is an open hardware BRI Asterisk Appliance running Astfin. I've written about this Asterisk-based appliance before. The Astfin.org blog author writes how he used the popular USB jtag (ICEbear) to connect to the new BRI Appliance rev B board. He then gives the blow-by-blow account of his experience, including running dumpreg & flashload. If you love to flash (memory that is - get your head out of the gutter) or if you're an Asterisk techie, it's a worthwhile read.

Some background - uCasterisk (you-see-Asterisk), which is a set of scripts, makefiles and patches to build Asterisk for uClinux and targeting Blackfin hardware. I should mention Digium's Asterisk Appliance is also based on Blackfin hardware. Also, uCasterisk was deprecated in favor of Astfin. The BRI Appliance certainly is more of an Asterisk fan's hobby rather than a direct competitor to Digium's Asterisk Appliance, which obviously is backed by Digium's technical support, customer service, warranty policy, etc.

Tags: Asterisk, Asterisk Appliance, Astfin, Blackfin, Digium, Linux, open source, uCasterisk, VoIP

Related Entries
• Astfin announces Asterisk Appliance based on Blackfin - Oct 30, 2007
• Digium reaches 1 million download milestone - Dec 19, 2007
• Digium - The Showstopper! - May 18, 2007
• Digium podcast of Mark Spencer's new role - Jan 30, 2007
• Hulk Smash Asterisk 1.6! - Apr 16, 2008
• Dell picks Fonality VoIP Phone System - Jan 23, 2008
• Digium Acquires Switchvox - Sep 27, 2007
• Digium Wins InfoWorld ''Best of Open Source Software'' Award - Sep 18, 2007
• Top 10 Reasons Why I love Asterisk - Aug 03, 2007
• Shoretel Rumors - May 01, 2008

April 29, 2008 04:55 PM

while installing asterisk their is a c file name called app_talkdetect.c and app_record.c, i want to change that file while any voice record is interrupted it should be recored in separate file insted of stoping. By default it was indicated as stop the function and process next thing in app_talkdetect.c i have to change it as insted of stoping i should record the voice.
ivrs (WIFI) at 2008-04-29 05:11

April 29, 2008 05:11 AM

I can call out over SIP with my E51 via Asterisk, but I can not receive calls on the phone. The phone rings, but freezes when I answer the call. The only thing that helps is removing the battery. I run version 100.4.20 RM-244 However, I have it working all well via sip.xs4all.nl. Both incoming and outgoing calls :-/
mdavids (Marco Davids) at 2008-04-28 18:06

April 28, 2008 06:06 PM

(more…)

April 28, 2008 03:30 PM

Thought I finally understood all this stuff then had unexpected behavior when I didn't pay attention to the little things... And for understanding how to use OUTBOUND_GROUP, check this: http://bugs.digium.com/print_bug_page.php?bug_id=2530 It's a parameter to tack onto the 5th argument of dial, not documented in the CLI help
skeagy (Scott Keagy) at 2008-04-25 22:43

April 25, 2008 10:43 PM

Maybe the existing documentation is fine and I just had to bash my head against it for a few hours, failing to achieve my immediate objective, before I became enlightened. Or maybe nobody has laid out a sufficiently clear explanation of how these things work. Here's my effort to explain it: 1) When a command GROUP()='groupname' is executed in the dialplan, it just puts a marker on the channel that will match when queried later 2) the GROUP_COUNT('groupname') function is just a query of all active channels (like the output of "show channels") that have the tag 'groupname' When I was struggling to understand this, I was thinking of GROUP like a variable that is scoped within a channel, and I couldn't understand how it would be reused in a different channel... like somehow GROUP was a variable name that held the incrementing value for that channel... but this is a completely wrong way to think about it. It is the channel itself that represents 1 increment in the value of GROUP_COUNT for a given groupname. A single channel might have multiple labels stuck on it to measure different things, but there is no notion of incrementing a counter within a single channel. Now OUTBOUND_GROUP is basically the same thing as GROUP... but instead of tagging the channel that starts the call (e.g. the ZAP channel for inbound calls on a PRI), it tags the channel or application, etc. that is bound to the destination side of this call... for example, a call into a PRI that goes to a meetme bridge would have a ZAP CHANNEL as the member associated with GROUP and the meetme bridge as the member associated with OUTBOUND_GROUP. Doing GROUP_COUNT tells you how many channels have that group label, and GROUP_LIST tells you how many different group labels are associated with a single channel. Hopefully I got this all right? It seems more clear to me now, and i guess I had to write this out to make it more clear for me... but certainly the existing documentation did not lend itself to me grasping this more quickly. Hope this saves someone a few hours of struggling. Or maybe it's just as obtuse as everything else out there already!
skeagy (Scott Keagy) at 2008-04-25 22:11

April 25, 2008 10:11 PM

Maybe the existing documentation is fine and I just had to bash my head against it for a few hours, failing to achieve my immediate objective, before I became enlightened. Or maybe nobody has laid out a sufficiently clear explanation of how these things work. Here's my effort to explain it: 1) When a command GROUP()='groupname' is executed in the dialplan, it just puts a marker on the channel that will match when queried later 2) the GROUP_COUNT('groupname') function is just a query of all active channels (like the output of "show channels") that have the tag 'groupname' When I was struggling to understand this, I was thinking of GROUP like a variable that is scoped within a channel, and I couldn't understand how it would be reused in a different channel... like somehow GROUP was a variable name that held the incrementing value for that channel... but this is a completely wrong way to think about it. It is the channel itself that represents 1 increment in the value of GROUP_COUNT for a given groupname. A single channel might have multiple labels stuck on it to measure different things, but there is no notion of incrementing a counter within a single channel. Now OUTBOUND_GROUP is basically the same thing as GROUP... but instead of tagging the channel that starts the call (e.g. the ZAP channel for inbound calls on a PRI), it tags the channel or application, etc. that is bound to the destination side of this call... for example, a call into a PRI that goes to a meetme bridge would have a ZAP CHANNEL as the member associated with GROUP and the meetme bridge as the member associated with OUTBOUND_GROUP. Doing GROUP_COUNT tells you how many channels have that group label, and GROUP_LIST tells you how many different group labels are associated with a single channel. Hopefully I got this all right? It seems more clear to me now, and i guess I had to write this out to make it more clear for me... but certainly the existing documentation did not lend itself to me grasping this more quickly. Hope this saves someone a few hours of struggling. Or maybe it's just as obtuse as everything else out there already!
skeagy (Scott Keagy) at 2008-04-25 22:11

April 25, 2008 10:11 PM

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April 25, 2008 06:31 PM

Zultys and Aastra Telecom have reached an OEM pact to delivery the next line of ZIP Phones. As part of the announcement, Zultys and Aastra announced full interoperability between the Aastra 5i Series and the Zultys MX (Media eXchange) line. I'm a huge fan of Aastra SIP-based VoIP phones and they are perhaps my favorite of any brand due to their XML capabilities and extensive customization capabilities. I have one sitting on my desk right now. The 57i CT in fact, which comes with a cool cordless handset that communicates wirelessly with the desktop IP phone.

"In Zultys, we are delighted to find a SIP PBX partner that shares our core underlying philosophy that open standards should afford customers an easy-to-use, cost-effective communications solution that offers the maximum flexibility," says John Drolet, Vice President of Sales at Aastra Telecom USA.

The interoperability testing involving the Aastra 5i Series, which consists of a line of SIP-based desktop and wireless telephones and expansion modules, including:
• 51i - Entry level single line set with 3 line/16 character display and 9 configurable speed dial keys using the number keypad
• 53i - Featured set with 3 line/16 character display and 4 programmable keys; compatible with up to three 536M expansion modules
• 55i - Advanced Featured set with 144 x 75 backlit LCD display, up to 26 programmable function capability; compatible with up to three 560M or 536M expansion modules
• 57i - Full Featured set with 144 x 128 backlit LCD display, up to 30 programmable function capability; compatible with up to three 560M or 536M expansion modules
• 57i CT - All the features of the Aastra 57i plus integrated cordless base providing VoIP mobility employing secure 2.4GHz Frequency Hopped Spread Spectrum technology
• 536M - Expansion Module with 36 programmable keys
• 560M - Expansion Module with LCD Screen and 60 programmable keys

Using any of the Aastra 5i series phones paired with the latest build of Zultys' MX software (version 3.2 or higher), gives some cool "out of the box" advanced functionality. In fact, Zultys told me, "MX software tightly integrates a suite of predefined critical features, including the so-called "Four P's" – that is, Paging, Parking, Pick-Up and Provisioning."

As a result, Zultys will soon make available the following additions to its portfolio of award-winning phones: the ZIP 53i, ZIP 55i, ZIP 57i, ZIP 57i CT as well as the ZIP 536m and ZIP 560m expansion modules. Under the terms of the OEM arrangement, those phones will be rebranded by Zultys and made available to its distributors and resellers.

"The addition of the Aastra 5i Series to our Zultys community creates a truly unique opportunity for organizations of all size migrating to pure IP or a converged IP environment. This further reinforces our commitment to offering customers unprecedented freedom of choice in finding a solution that is perfectly suited to meet their needs," explains Jonna Paquette, VP of Sales and Marketing at Zultys.

Tags: 57i, Aastra, Aastra 5i Series, Aastra Telecom, IP phone, Media eXchange, VoIP, ZIP 536m, ZIP 53i, ZIP 55i, ZIP 560m, ZIP 57i, ZIP 57i CT, Zultsys, Zultys MX

Related Entries
• More on the Aastra AastraLink Pro 160 appliance - Mar 18, 2008
• Aastra AastraLink Pro 160 - Mar 18, 2008
• Packet8 Drops International Calling Phone Rates - May 07, 2008
• O-FONE Supports SIP on Symbian S60 mobile phone - May 07, 2008
• Telcos Join Forces to Battle Skype? - May 06, 2008
• Shoretel Rumors - May 01, 2008
• TringMe Now Works with AOL/AIM - Apr 30, 2008
• Call Screening Patent - Apr 30, 2008
• VoIP Call Screening - Apr 30, 2008
• Skype 3.8 for Windows - Apr 29, 2008

April 24, 2008 02:29 PM

The Asterisk development team has released versions 1.2.28, 1.4.19.1, and 1.6.0-beta8.

April 24, 2008 01:52 AM

Asterisk Project Security has posted details of an amplification attack based on IAX2.

April 24, 2008 01:45 AM

April 23, 2008 05:37 PM

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April 23, 2008 03:12 PM

I just got a mail from Siemens: Das Gigaset S675IP akzeptiert (RFC-konform) nur SIP-Notify Messages die auch vorher via Subscribe-Message angefordert wurden. Die meisten übrigen SIP-Telefone am Markt akzeptieren jede (auch nicht-subscribte) Messages. Asterisk unterstützt erst ab Version 1.4 SIP MWI-Subscriptions. Prüfen Sie bitte, ob Sie eine Asterisk Version > 1.4 nutzen. In english it means that the S675IP only take SIP-Notify Messages wich are requested before. Asterisk > 1.4 uses the SIP MWI-Subscriptions. I hope this information helps. Richy
strikegun (Richy) at 2008-04-23 12:02

April 23, 2008 12:02 PM

Computing UK has an interesting article about Empire Cinemas using VoIP in their movie cinema chains. What better things in life are there than VoIP and movie theater popcorn? Not many... But maybe I'm biased...

Empire Cinemas is using the Voicenet Solutions platform using Cisco hardware, to VoIP-enable the main office and 17 nationwide branches. Importantly, Empire hopes to use the software to generate “substantial cost reductions”.

In fact, the article states, “The financial benefits speak for themselves and were certainly a factor that could not be ignored,” said Empire Cinemas’ head of IT Julian Timm. Julian adds, “We envisage we will save almost £100,000 over the first 12 months which includes substantial line rental, maintenance and call cost reductions."

Fantastic! So with these great cost reductions due to VoIP, will the price of my $10.50 movie ticket & my $6 bucket of popcorn go down? Hello? Anyone?

Well, it's a darn good thing for you that my homemade popcorn isn't nearly as good as your movie theater popcorn or I'd stay home and watch movies on PPV or Netflix DVDs instead.

Tags: cinema, Empire Cinemas, movie theater, Netflix, popcorn, VoIP

Related Entries
• Telcos Join Forces to Battle Skype? - May 06, 2008
• Shoretel Rumors - May 01, 2008
• TringMe Now Works with AOL/AIM - Apr 30, 2008
• Call Screening Patent - Apr 30, 2008
• VoIP Call Screening - Apr 30, 2008
• Skype 3.8 for Windows - Apr 29, 2008
• OnState, the Skype Call Center functionality Enabler, Used by B4 Consulting - Apr 30, 2008
• Rev B of Astfin's BRI (ISDN) Asterisk Appliance Arrives - Apr 29, 2008
• AOL Launches Open Voice APIs - Apr 29, 2008
• Cisco and Microsoft in Clash of the UC Titans - Apr 24, 2008