Skype support will be available on PSP
Jajah liberates VoIP users from PC
Yahoo Messenger for Vista Adds VoIP
British padding lampposts?
SippySkype SIP-to-Skype Gateway
Last month I wrote about my strong disagreements with a guest blog post on Skype Journal that was titled "A SIP/Skype Gateway Is NOT In The Forecast". I disagreed with the premise that a SIP/Skype gateway isn't in the forecast. Well, Skype Inc. may not care about offering a SIP-to-Skype gateway, but that doesn't mean VoIP enthusiasts are going to sit around and not build their own SIP-to-Skype gateways! I've blogged on a few occasions where I discussed the desire for Skype users to have SIP connectivity and various home-brewed solutions.
Well, we can add another SIP-to-Skype gateway solution to the mix. I just discovered SippySkype today, which is an open source & free SIP-to-Skype gateway.
Check out the features:
SippySkype is Java software that allows you to make and receive Skype calls from your SIP/VoIP adapter. Basically a Skype/SIP Bridge/Gateway/Proxy
Call Skype Users using speed dial or use Skype out.
SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks - 1.1 or higher
Skype to SIP authentication/denial mappings via incoming Skype User ID - 1.1 or higher
Support RFC2833 touchtone decoding (DTMF) - 2.0beta or higher
Could be used as an endpoint with Asterisk
Auto play pre-recorded file(s) to SIP callers - 2.0beta or higher
Incoming SIP Pin number authentication and dialing - 2.0beta or higher
Open Source - You can modify/fix it if you like.
It's free
System Requirements:
Skype Client
Working Java 1.6.0 or better runtime
mjsip/mjua 1.6 http://www.mjsip.org/ - Use those included with SippySkype as some bugs have been fixed.
Skype4Java 1.0 https://developer.skype.com/wiki/Java_API - Unmodified
SIP/VOIP adapter such as a spa-3102 to make and receive Skype calls or register with a provider or Asterisk.
Should work where Skype4Java works (windows/linux/osx). (I'm using it on Windows XP)
TomTom GO 930T & 730T
Where's My Cell Phone?
It's happened to be best of us - we misplaced our cell phone and after frantically looking for it we turn to our spouse/significant other and humbly ask, "Have you seen my cell phone?" or accusingly "What have you done with my cell phone? I had it right here!".
Now if your cell phone is on you can simply call it from your landline and hope you can track down the ringing (assuming you didn't leave it on vibrate). But what if you are like a number of people who have gone 100% wireless with no landline service? Well, you could use Skype if you have SkypeOut credits, but not everyone does.
So what to do? Well, head on over to www.wheresmycellphone.com and enter in your cell phone number and it will dial it for you for free. No doubt you'll hear your lost cell phone ringing in your pocket or the sofa seat cushions.
Now of course there is potential for abuse. You can enter in the cell number of a person you want to prank call and it will dial their number while protecting your anonymity. You could even write a script that hits this page multiple times to barrage your victim with countless calls. Perhaps even schedule the script to run at 3am. Ahh the fun to be had...
PBXtra 4.0 Released
Fonality today announced the release of PBXtra 4.0. New features in PBXtra include FindMe with Boomerang Mobile Integration, a feature that uses presence detection to automatically find employees on their mobile devices, allowing them to answer the call or bounce it to another extension. Other new features include tighter integration with mobile phones and web browsers, and enhanced support of branch offices.
One really cool feature that is part of the Boomerang Mobile feature is that you can dial *1 to record the mobile phone call and have it automatically stored on the PBXtra server. Another cool feature that Chris Lyman CEO of Fonality gave me a sneak preview a few weeks ago was FONcall, a new PBXtra plug-in for the Firefox web browser. It turns any phone number on any site into a link. When you click the link, PBXtra will automatically take an Aastra and Polycom off the hook and dial out to the number hands-free. I pointed out to Chris that there are similar plug-ins out there including Skype's browser plugin, and I added that Skype's plugin often brings your computer browser to its knees. Chris said he was well aware of that fact and they spent countless hours developing their plugin to make sure it wasn't a CPU hog. Also, no support for Internet Explorer yet - possibly never since Chris stated it was much harder to develop plugins for IE. (though I should point out that Skype's plugin works on IE)
Greg Galitzine has more on this news and was the first to post the story about this new release.
Talkonaut 4.0 native Symbian S60 edition released
The GTalk2VoIP dev team just released their VoIP+chat application for Symbian S60 phones that runs as a native (.sis file) application. I should stress that several Nokia smart phones come with a SIP stack, however they are limited to WiFi use only and won't work over your cellular network. You can thank the carriers for that one. Talkonaut on the other hand is not bound to WiFi only, allowing you to make calls over GPRS, EDGE, 3G or WiFi. Talkonaut has essentially developed their own VoIP SIP stack that allows Talkonaut users to make voice calls over most data connections your mobile phone might have, such as GPRS, EDGE, 3G and WiFi. The application also sports the ability to make free VoIP calls to Google Talk users, to SIP phones, to MSN, Yahoo, AIM and ICQ voice capable IM clients. I believe you can even chat with MSN Messenger, Yahoo, AIM, and ICQ users.
Here's a list of the new features in the 4th release according to the release:
- Talkonaut 4.0 was entirely rewritten in C/C++ and now runs on Symbian S60 3rd edition based Nokia smart-phones. Moving from Java to native platform allowed to reduce memory usage, improved speed and to add some functionalities that were not previously available for Jave applications, like VoIP, access to file local system (improved file transfer) and interaction with other applications (Web Browser for opening URLs right from chat window).
- Using a set of narrow-band Speex codecs and relying on Jingle Audio extension to XMPP protocol (same as implemented in Google Talk), Talkonaut now brings to the world a very powerful combination of IM chat, Presence and VoIP calls made over data connection.
- Talkonaut 4.0 allows to make free VoIP calls to Google Talk users, to SIP phones, to MSN, Yahoo, AIM and ICQ voice capable IM clients, as well as to other Talkonaut fellows.
- Talkonaut 4.0 allows to receive free calls from SIP phones (or any other VoIP networks), from Google Talk, MSN, Yahoo, AIM and ICQ users.
- Talkonaut 4.0 allows to make cheap VoIP calls to any mobile or landline phone number in the world.
- Talkonaut 4.0 allows to define any number of SIP accounts and use them to make free or cheap calls over third-party VoIP/SIP carriers. A flexible Dialing Plan feature is helpful for choosing routes to destinations between different carriers.
- Talkonaut 4.0 is fully equipped with all the features previously available in J2ME version of Talkonaut 3.0, inherits the same graphical user interface and menu structure.
Talkonaut 4.0 runs on the following sets of Symbian S60 based Nokia smart-phones:
o 3rd Edition: 3250, 5500, 5700, 6110, 6120, 6121, 6290, E50, E51, E60, E61, E61i, E62, E65, E70, E90, N71, N73, N75, N76, N77, N80, N81, N81 8GB, N82, N91, N92, N93, N93i, N95, N95 8GB
Talkonaut 4.0 can be downloaded over-the-air using mobile web browser from http://get.talkonaut.com/, or via PC from http://www.talkonaut.com/download.shtml
Junction Networks Adds Inbound Bridge to onSIP Hosted PBX
Junction Networks, provider of the hosted onSIP PBX service, has launched Inbound Bridge, an accessory service that ties in third-party providers of international and domestic DID numbers. Inbound Bridge saves money for Junction Networks’ business customers by allowing them to find the best per-minute price for inbound VoIP calling minutes in their chosen geographic regions. Essentially this bridging application lets hosting customers use third-party providers of DID numbers for best LD price in desired footprint.
According to Junction Networks, "Inbound Bridge solves a vexing problem for companies that want local numbers and 800 numbers in foreign countries or specific domestic regions—numbers typically rented from other VoIP network providers. Many of these providers simply deliver calls from one end point to the other; they don’t implement the key function of the SIP VoIP signaling protocol that allows callers to navigate phone menus, transfer calls, put callers on hold and trigger other events. Without this function (specified in the IETF’s RFC 3515), inbound callers are disconnected when they “press 1 for sales,” or try to transfer to extension numbers for specific people. Or at the least the DTMF touch tones are not recognized.
Junction Networks' Inbound Bridge performs the missing SIP function, maintaining the two-way SIP signaling needed for interactivity after a call is already established. It can therefore pass the entered touch-tone digits to the auto attendant and other PBX applications. It can integrate onSIP with any network vendor having an open SIP implementation (permitting communication with other SIP networks).
“This is a perfect opportunity if you’re a company that wants a toll-free service from Europe or Asia, or you have some other reason to use a third-party phone number provider,” said Rob Wolpov, president, Junction Networks. “But you do want to use our hosted solution, because it works very well, you have company extensions in one or more sites, and you like the price of that.” ($39.95 a month for core voice applications and an unlimited number of SIP extensions, with free calling between them.)
“You can get your toll-free origination from any provider that offers SIP delivery of inbound calls, and just have it go to us through the Inbound Bridge. We charge $1.95 a month per DID, and half a cent a minute to cover our cost of the Bridge. If you can get your toll-free for one or one and a half cents a minute, you wind up paying two cents a minute for toll-free, as opposed to paying almost four cents a minute to Junction Networks.
“That’s completely OK with us,” Wolpov stressed. “Go somewhere else to get your minutes and then come to us for the auto attendant, the ACD queue, the voice mail, all those sorts of applications.”
To date, Junction Networks has tested the Inbound Bridge with international DID provider Voxbone. Other third-party DID providers are to be tested and added in the coming months.
“We’re happy to supply onSIP customers with our core value—locally dialed phone presence in more than 40 countries around the world,” said Rodrigue Ullens, Voxbone CEO. “With our intercontinental voice-only backbone, Junction Networks’ Inbound Bridge and onSIP platform, these enterprises can get the best value in DID numbers and international VoIP transport, plus all the convenience and flexibility of hosted IP PBX.”
Nuvio sues Garmin over nuvifone
Hosted trixbox IP-PBX
SimplyWiFi SIP Phone
SimplyExchange Skype PBX Gateway
Skype Customer Service
Skype support will be available on PSP
Jajah liberates VoIP users from PC
Raketu does VoIP without PC, Internet
Vonage ends legal dispute with AT&T
Will Skypephone bring revolution in Mobile VoIP?
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