Here is a cheat sheet that I found out there in ZWAME , and here I play for myself or for someone to eventually give way.

Tested and working on E71. Very useful for doing some chamaditas to the fixed network from a wifi access.

 Profile name: anything eg Sapo Service profile: IETF Default access Pt.de: WiFi network from home eg. Public user name: +35130 xxxxxx@voip.sapo.pt (number created in the customer area frog) Use compression: No Registration: When needed Use security: No Proxy server address: proxy.voip.sapo.pt Realm: proxy.voip.sapo.pt Username: +35130 xxxxxxxx (number created in the customer area toad) Password: password defined area customers frog Permit referral.  Free: Yes Transport type: UDP Port: 5070 Ender.  server record: voip.sapo.pt Domain realm: voip.sapo.pt Username: +35130 xxxxxxxx (number created in the customer area toad) Password: (password defined in the customer area frog) Transport type: UDP Port: 5060 

Today I found myself looking at a mall near here (Forum Almada) to a keyboard Lifetech, which curiously I found out that Portuguese is a project which I always thought they were a white mark from Logitech, as I notice some similarities products, sold at a price much more appealing.

The keyboard in question is umNetPhone, and cost in the Jumbo Box about 20 EUR. Good price since it even had a mouse that seemed to be of dubious quality.

I looked for him because of the phone, which made me curious. Said to be a VoIP phone. ... ... and interpreted it as a softphone to connect to the PC, I do not think I was wrong. But the provision of telephone keypad, taking into account that I use my mouse to the right, and because it is VERY weird as keyboards and mice, led me to abandon the idea.

Probably this would be a * single * like many others I buy, and I end up not using, and is here at home. I'm really weird about the layout of the keyboard and mouse to ergonomics ... my current choice falls on an ordinary Logitech Internet Pro Keyboard (PS2), and a simple USB Optical Mouse, HP. Both values are very restrained and quite affordable.

Returning to the keyboard NETPHONE, was bringing a voucher Netcall extinct, which led me to think: this must really be old!

Netcall ended in December 2008, and I learned this at the time saw Carlos Andrade's blog ... a shock to all who had accounts there. Neither the nomadic numbering got away ...

Today curiously after trying to open the same page, he concludes that this is no longer opened. A Google search for Netcall led me to the blog Vitor Magalhaes, specifically the post "Why Netcall failed" .

The post in question, which I found very interesting, talks about the business in question, since it was linked to the project (founding partner) of which failed, and particularly on VoIP, marketing and business models - good and bad options can be taken on a project of this type.

I recommend reading, especially for those who are thinking "stick" in a startup is hard to be ... "entrepreneur" in this country, so all the good tips should be used.

After all projects are to survive, and become "major ones" with the imprint nationwide!

To summarize, what about the keyboard ... obviously did not buy!

I decided to accept the invitation I was made to integrate the Planet Geek national aggregator of blogs that I follow regularly.

So from now on my posts are also scattered around.

To all who will read me, and be given the first post, a very good day.

It is precisely in April a year since I decided to buy a NSLU2 and of course obviously the choice is not made in order to use it only to share the hard drive on the internal network, job perfectly for this small "device".

So far have been with the Unslung 6.8Beta (Released 12/04/2006) to serve some internal pages test with PHP, streaming to my Kiss DP1500 with Kissdx, some ftp and a few more thingy, however came out in 31/12/2007 6.10Beta version.

The upgrade is not advised to those who have version 6.8 because they have no practically nothing new, however it is strongly recommended to install the system go back / a new system.

So given that it starts with getting some garbage on the disk, and I'm going to change the disc, I in this beautiful weekend of rain, apply in Beta Unslung 6:10 vermin.

Already some weeks I have also been installing the system asterisk in NSLU, however confess that set everything "á la pata" is complicated when the weather is not much.

Thus it was reported early this morning in mailinkg list of NSLU2-asterisk optware the availability of the package asterisk 1.6.0-beta4.

So I'll add it and once again lose a little time to play with it. I wanted to see if it was this that could link my "PPCA domesticated" with the Voip VoIPTalk that use a phone connected to my router MEO in order to be able to create a dialplan "maneirinho" to call other VoIP networks.

After much searching I think I decided

Today it is vital have access to the technologies that save money and simplify life.

So on that basis will have some time I've been keeping an eye on IP phones.

Established, however, that do not want a brick around here, despite the many advantages that have a pure IP phone.

However a convergence solution would be best, ie: DECT + VoIP.

As a bonus even consulting the emails and do another "porradão" of cool stuff.

Manufacturer Link

As all that is good ... it's rare to be on sale in Portugal. Found it on sale in OneDirect for 145 EUR. Maybe August will arrive here in the house.

At that time it may be that now is even cheaper. For now I get the manual to better study the potential and see him.

Direct connection to gigaset.net which is nothing less nothing more than a network of Jabber, since the phone supports XMPP natively. Pity that in Portugal there are no good projects. It's all commercial ... expensive and worse.

By now I VoIPTalk the 2Wire router that came with mine. Serve for testing and for the expenses while awaiting any possible novelty that may arise.

It seems to me that the time to DECT telephone will have its days numbered, but this is just my thoughts. As it makes no sense to have infrastructure for data and voice on business, also at home this does not make sense. So it seems to me that the future will most certainly voip.

Highlights

• Answering machine with up to 30 minutes of recording time
• High Definition Sound Performance
• Expandable phone system with multiline functionality for up to 6 handsets and 6 SIP accounts from Different Providers
• Gigaset.net ¹: Worldwide free of charge calls Between Gigaset.net supporting VoIP phones.

Features

• Easy configuration of internet telephony (VoIP) with or without a PC
• Up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call with multiple handsets
• Screensaver to display internet information services
• Increased virus protection thanks to protected operating system
• Convenient handsfree talking
• CLIP ¹ / ¹ CNIP
• Large illuminated display with 65.000 colors and 128 x 160 pixels
• Intuitive icon-based menu
• SMS ¹ with up to 640 characters
• Up to 250 vCard entries to store first name, last name, 3 numbers, e-mail address and birthday of your contacts

¹ Dependent on network operator

Gigaset S675 IP - Technical Data

Update: Voip Frog with MEO is possible.

After some more testing, and after noticing that here on this page it says that:

Configuration data for mobile phones and other equipment.

Proxy Address: proxy.voip.sapo.pt
Proxy Port: 5070
Address register: voip.sapo.pt
Register port: 5060
SIP URL: No VOIP (eg +351302019999)
Username: No VOIP (eg +351302019999)
Password: set by client

I noticed that it had not been put in the username +351 (I think this would be the error).

However the correct connection profile for:

and after putting the user and pass

the result is this:

And ready ... so is the myth terminated. Make and receive calls with VOIP Sapo.

Myth: MEO does not make VOIP calls. Are blocked because the bogeyman (PT) just wants to make money and blah blah blah (insert here all the criticisms will detract PT circulating the forums on the net, especially with respect to m and multiply by two - who account adds one point) .

Facts: seems to be complicated to configure anything, it's a fact.

Reality: I finally managed to configure the line of work with me for VoIPTalk.

Unfortunately not get it to work with VoIP SAPO. The user has not recorded Rita Costa. I honestly did not realize .... It seems that giving a timeout on the 2Wire router. Anyway I use a softphone, the SJPhone to test with the asterisk (in those days I feel like playing) and also could not register. I get the idea that it would be a problem of SAPO. Does Rita by changing the password was not changed immediately?

Proof:

Here is how to make the setting in the 2Wire management console 2700HGV-2 Gateway.

He had already tried several times but had failed. This afternoon I managed to obtain a "signal" stable on the phone, but could not make calls.

After some more settings and the data above, I did a "Reset" to the profile (MDC>> Resets>> Resets the selected Voice Profile ") and then when I consulted on the homepage of the router gave the good news that was available.

Of course it is necessary to pass by Phone Wiring by the Set Up Phone Lines, Match Phones etc etc all on site. Interestingly also the splitter that came with the 2Wire where it says PHONE1 phone2 corresponds to the router and vice versa. "Bug" curious.

They are also settings where the provider explains how to configure a device.

I confess that I did not think it will not trivial given that documentation and requests from the 2Wire router are somewhat vague.

PS: Every time lack the patience for blogging these things, but that seems interesting and the result obtained here is breaking the myth.

In your phone's configuration menu there Should Be an option to define the SIP Server, SIP Registrar or SIP Domain value. Set the value to: voiptalk.org.

For SIP Authentication, set your SIP User ID or similar to your VoIPTalk ID (eg 84,411,076) and Set Password SIP or similar to your six-digit password VoIPTalk Provided in your activation email.

If You Have an Outbound Proxy setting, set this to: nat.voiptalk.org: 5065. Alternatively, if this setting Allows you to define the port in the Separate field, set the IP Address to: nat.voiptalk.org and the port number to: 5065.

Confirm the settings by reloading or rebooting your phone. Dial 902 to confirm your configuration.

For some time now that we have been promising, but the truth is that both I and Louis seem to suffer the same problem: lack of time.

Luis already bought your VoIP phone, and I do not either, but plan to use the VoIP functionality of my MediaGateway Meo to test with a simple phone.

The truth is that even a simple softphone used to it. The idea? Well ....

Know that this short list of features here makes me want to run the world.

Just thinking about a complete open source PABX interconnection with CTI without royalty is excellent. IVR, call centers, etc etc .... there are many ideas.

There is no time for such hardware megalomaniac ideas but Asterisk VoIP PBX is connected as well as possible.

For now the AsteriskNOW (Asterisk for Beginners as they call it) is already installed. Failure now to arrange a phone and get some experience.

Were it not work on projects that are in arrears, was so good ...

What is Asterisk

Asterisk is the world's leading open source telephony engine and tool kit. Offering Flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions ... for free.

Asterisk ® is released to open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk ® is the most popular open source software available, with the Asterisk Community Being the top influencer in VoIP

Asterisk as a switch (PBX)

Asterisk Can Be configured to the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS) and digital (T1/E1) connections.

Asterisk runs on a Wide Variety of Operating Systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and Provides all of the features you would expect from a PBX including many advanced features That Are Often Associated with High end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum Flexibility and Supports Voice over IP in many protocols, and Can interoperate with almost all standards-based telephony equipment using Relatively inexpensive hardware .

Asterisk as a gateway

It Can Also Be the built OUT The Heart of the media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk's modular architecture Allows Between convert it to a wide range of communications protocols and media codecs.

Asterisk as a feature / media server

Need an IVR? Asterisk's got you covered. How about a conference bridge? Yep. It's in there. What about an automated attendant? That Asterisk does too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok

Asterisk in the call center

Asterisk Has Been adopted by call centers around the world based on ITS flexibility. Call center and contact center developers Have built complete ACD systems based on Asterisk. Asterisk Has Also added new life to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and more.

Asterisk in the network

Internet Telephony Service Providers (ITSPs), competitive local exchange carriers (CLECs) and even first-tier incumbents Have Discovered the power of open source communications with Asterisk. Feature servers, hosted services clusters, voicemail systems, pre-paid calling solutions, all based on Asterisk Have helped Reduce costs and enabled flexibility.

Asterisk everywhere

Asterisk Has Become the basis for Thousands of communications solutions. If You Need to Communicate, Asterisk is your answer.