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Gizmo, created by SIPphone is one of my favorite VoIP clients and not just because they have one of the prettiest & slickest looking interfaces. Nope, Gizmo has some of the best features in any softphone client you will find and unlike
Skype which uses a proprietary protocol, Gizmo uses the SIP standard. Back in July, Gizmo announced
their "All Calls Free" program which allows you to call all of your Gizmo friends for FREE on their landline or mobile phones in 60 Countries. SIPphone said today that it is extending its All Calls Free calling plan to
business users world wide.
The only catch is that the person you are dialing must also be a Gizmo user with their landline and/or mobile phone registered within the Gizmo client. According to SIPphone, "Callers are encouraged to make free PC-to-PC calls whenever possible. The All Calls Free calling plan applies when both call participants are active Gizmo Project users making a few phone calls per week with Gizmo Project. Free calls may originate from anywhere in the world, but must be to a qualifying number in one of the 60 countries for which the plan is offered. Calls must be made from the caller's contact list to either the "home phone" or "mobile phone" number the call recipient included in his or her profile, and both parties must have shared each others profiles with one another." If the call doesn't qualify under the All Calls Free plan, then you simply pay the low rates offered by SIPphone.
The beauty of the SIPPhone client is that it supports dual SIP logons. Thus, you can register with Gizmo's SIP server, as well as any second SIP server you wish, such as an Asterisk IP-PBX, Switchvox, epygi, Mediatrix, etc. Now you can simply deploy Gizmo Project to all your employees (which integrates nicely with Asterisk and other PBXs) and get free calls between all your employees. With this dual SIP configuration, incoming calls from your PBX will be clearly identified, as they are when a call comes from Gizmo. In addition you can choose which outgoing line to use (i.e. Asterisk or Gizmo).
If I have some time I'll play around with the latest version of Gizmo Project and report back here.
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Written by VoIP & Gadgets Blog on September 19th, 2006 with no comments.
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Aswath noticed that it appearred AOL was cancelling their TotalTalk service. When you go to
www.totaltalk.com you are greeted with this message:
"We are very sorry to inform you AOL® Enhanced Services ("AOL ES") has decided to discontinue offering the TotalTalk™ service. For this reason, the TotalTalk service will be terminated on or about November 30, 2006."Andy basically says
TotalTalk was killed in favor of AOL's new PhoneLine service.Andy says that TotalTalk was basically AOL VoIP 1.0 which helped give birth to PhoneLine resulting in AOL VoIP 2.0. Some interesting thoughts by Andy you should definitely check out.
The difference between TotalTalk and PhoneLine is that TotalTalk was more of a Vonage-like solution to replace your home phone line (hardware ATA), and
PhoneLine is a 100% software-based (softphone) VoIP solution that supports both outbound and inbound (DID number) dialing. PhoneLine is an interesting product, but it has to battle the "me too" factor with the likes of
Google Talk, Skype, Gizmo, and others. I should point out that AOL TotalTalk was riding on the Level3 network and really didn't add much value or features that you can't get from any of a dozen VoIP providers. AOL was simply lost in the shuffle of the plethora of VoIP providers out there - many of which also ride on Level3 and with the same exact rates.
Ask any of your techie friends if you knew AOL did VoIP. I bet most didn't know. I bet many average people know
Vonage does VoIP but ask about AOL and they would probably say "AOL does VoIP?" Who would have thought that AOL with one of the most well-known brand names would get lost in the VoIP marketing wars?
If AOL wanted to compete with the likes of Vonage, perhaps they should have copied Vonage's marketing model by spending millions of dollars on TV commercials with a catchy tune. Or they could have entered a price war and offered $10/month unlimited VoIP service just to attract customers. Sure they'd probably eat millions of dollars in the first year, but they needed to do something drastic to stand about the "me too! me too!" VoIP service provider crowd.
Who am I kidding? In the end, even if AOL spent millions, the cable and phone companies
are going to win the war against the single play VoIP providers
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Written by VoIP & Gadgets Blog on August 31st, 2006 with no comments.
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Some great news from SIPPhone, developers of the
Gizmo Project announcing support for the open-source Asterisk PBX... The "Asterisk economy" continues to grow by leaps and bounds.
check it out:
SIPPhone today announced the delivery of Gizmo Project 2.0, highlighted by its support for Asterisk PBX software. Giving Gizmo Project users the ability to log into Asterisk means they can now be universally reachable via their Asterisk PBX or directly through the Gizmo Project network. The option to seamlessly receive calls from their office PBX while anywhere in the world leverages Gizmo Project's advanced NAT (“network address translation”), firewall and router traversal features and server infrastructure. More information about Gizmo Project 2.0 with Asterisk support may be found at www.gizmoproject.com/asterisk.
The benefits of using Asterisk to power an office PBX can be significant in terms of cost savings, efficiency and access to features previously only available to large businesses. Using Gizmo Project as an office softphone lets users easily place calls without the burden of special VoIP phones or the expense of traditional phone call charges. Incoming calls to an office PBX will reach mobile workers anywhere in the world. Gizmo Project also provides such high-end features as voicemail-to-email, free conference calling, call history, free access to millions of people on SIP-based networks, and built-in instant messaging (IM) capabilities.
“Whether a company is focused on a completely free Asterisk PBX installation or is running a premium version, Gizmo Project is now the ideal softphone for use with any size deployment,” said Michael Robertson, Chairman and CEO of SIPphone. “Our experience at routing millions of calls through almost any network setup means that mobile computer users can be reached anywhere as if they were physically in their office. Plus, companies save money with low domestic and International calling rates using Gizmo Project,” Robertson concluded.
Companies around the world deploying the Asterisk PBX software, premium PBXs developed by such companies as SwitchVox, Epigy, webFones, or any other SIP-based PBXs can now easily use Gizmo Project for making and receiving calls. Specific information about setting up Asterisk for use with Gizmo Project 2.0 may be found at www.gizmoproject.com/setupasterisk.
The free Gizmo Project software for Apple Macintosh, Microsoft Windows and Linux computers delivers crystal clear VoIP calls. Gizmo Project uses licensed, best-of-breed audio codecs such as GIPS and employs media relays around the world to route calls through the most efficient path. Gizmo Project also routes calls through one of several phone partners which provides for the lowest possible per minute International calling rates. PC-to-PC calls are always free.
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Comments on this Entry:
(michael on
May 23, 2006 1:00 PM)
I found it interesting that even though it is a generic SIP client implementation it is being marketed with Asterisk. Anyway, I am grateful for this, as now I can stick with Gizmo Project as my softphone client to use with my Asterisk server instead of using X-Lite. One thing I noticed is that it won't register with the server using a private IP address, it must be the public address of the server. Maybe not very many people have Asterisk behind a NAT, but I do so I noticed it.
(Tom Keating on
May 23, 2006 1:59 PM)
You can't use a private NAT'ed address? Really?
Have you tried port forwarding? i.e. port 5060
although, you shouldn't have to since Gizmo claims to be able to traverse NAT firewalls. Haven't tested it with Asterisk, but last time I used Gizmo with SIPPhone service, it traversed the firewall just fine.
(michael on
May 23, 2006 2:51 PM)
I guess I wasn't clear in my statement. My Asterisk server is behind NAT, on the same local subnet as my PC. In the Gizmo Project Secondary Account settings, I must put the public IP address of the Asterisk server, as it doesn't work with the private IP address of the Asterisk server.
(michael on
May 23, 2006 3:09 PM)
As a follow up, Gizmo Project appears to my Asterisk server as being at IP address 198.65.166.XXX, so the Secondary Account is being proxied by sipphone as well. That explains why the Secondary Account server must be a public IP.
Written by VoIP & Gadgets Blog on May 23rd, 2006 with no comments.
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