Archive for January, 2009
Friday, January 30th, 2009
I bought the folloing kit ‘X5v Integrated VoIP Solution with bundled Global Village service – Model 5565′ after reading the review on adslguide.org
The first thing I did was check the firmware version and then I downloaded the latest firmware:-
X5v Firmware .exe
…..and flashed it.
Leaving VOIP on the back burner for a second, the ADSL router/firewall was a doddle to setup. I plugged my Wireless access point into it and I was working within seconds.
I can’t see anywhere to setup DHCP reservations and I need to get tracert working configured to work too. Web interfce is a little slow, but straightforward
Now onto VOIP.
Out of the box Zoom configure this with GlobalVillage, however they do NOT tie you to this. Click on Advaced VOIP settings and uncheck the automatic configuration button and you’re away and free to setup the router with whomever you want. I had it working with Sipgate within minutes and then reconfigured ot for VOIP user just as quickly. I’ve appended my configurations to this review, incase you want to setup this router with sipgate or VOIP User.
It would have been nice to have support for multiple SIP providers, eg use sipgate for incoming and say VOIPBuster or VOIP User for outgoing. But unfortunately it doesn’t!
The Zoom teleport feature is superb – I have a DECT base station plugged into the back, serving 4 phones with both normal PSTN and VOIP (By using the # prefix).
Only drawback with DECT is that once on the DECT phone you cannot make another call on another DECT handset.
In my house I have a DECT basestation connected to the router. VOIP is only available on the DECT Handsets. PSTN is also available on these handsets. I have 2 normal phones, connected to my incming line and not through the router. I can only make PSTN calls with these.
If I receive a PSTN call on the DECT phone, I CANNOT make a VOIP call.
If I receive a PSTN call on a one of my normal phones, I CAN make a VOIP call on a DECT handset.
If I receive a VOIP Call on the DECT Phone, I CAN make a simultaneous PSTN call on my normal phone, but I CANNOT make a simultaneous VOIP or PSTN call on another DECT Handset.
I hope thats clear!
Overall based on features/Performance I would give the Zoom x5v a 4 out of 5 and for the price I paid (£34 inc vat & P&P) a 5 out of 5.
Thursday, January 29th, 2009
Still searching for that “miracle” box that would allow me to integrate my SIP calls with a regular phone and my computers routing needs, I decided to give the D-Link 1402s a try. Did it hit a home run or should the search still go on? We’ll find out soon, but as a reminder, after trying the Zoom 5567 v3, I came to the conclusion that it wasn’t uPNP compliant in its 1.0 firmware state, which is a big no-no with my configuration, and also was making my cable connection drop almost 50% of the time.
With D-Link’s offering, this is what your $99 (SRP) will get you :
- a D-Link 1402s router with 4 ethernet LAN RJ45 ports, 1 ethernet WAN RJ45 port and 2 RJ11 phone ports
- a power supply
- ethernet cable
- a quick start guide
- an installation CD
Nothing special on the router configuration front, your usual IP settings, NAT and port forwarding setup. Same old news as with almost every other router. Do take note however that most 1402 units you will find at retail are locked for the Lingo VoIP service and therefore won’t let you use anything else.
There is a username and password to access the Advanced pages required to change the SIP settings, but, as far as I know, D-Link will not communicate them to you if you bought a Lingo box**. So make sure you order a 1402s and not a 1402s-l (L for Lingo I assume).
I must mention here that the unit I received acted very strangely during the first few hours I was trying to configure it, and it wouldn’t let me connect via a web browser. I spent quite a while with D-Link tech support, which I’m pleased to say is not outsourced to some barely english-speaking foreign country, but we couldn’t even ping the device. As weird as it may seem, I decided to try one more time two days later, before sending it back, and to my astonishment everything worked like a charm. Very weird but a happy ending for once.
There is no need to install the software that comes with the box, unless you run an outdated OS and need USB drivers and such. I installed the D-Link on a Windows XP network (2 PCs and 1 laptop), and everything has been working flawlessly.
I’m going to repeat myself here for those that already read my Zoom 5567 v3 review, but here it goes.
I will not describe the router setup as this is something that anybody who ever used a router will be familiar with. Furthermore, what will interest you is probably the VoIP features.
I tried the RJ45 phone ports with two different wireless phones, everything worked great. Sound quality from SIP to SIP or SIP to PSTN or PSTN to SIP was flawless. The VoIP setup pages allow you to change everything in your SIP setup, but it’s not as user-friendly as the Zoom unit in my opinion. That said, all the options you should need are there and they work, which is what matters.
I used the router with my SIPphone account and the call quality is great, no problems so far. I also got it to work with my outbound SIP to PSTN voipuser.org account and calls have been crystal clear so far.
I should mention that despite what I was told at the CES, it seems like this router is uPNP compliant as it works with my Play@tv streaming unit very well and without any messing around. I would like to see a way to turn that off in an upcoming firmware for those users that have no need for uPNP and would rather be more secure, but I’m nitpicking here.
So, have I found the perfect router for my setup? In a word, yes! Now if I could only figure out how to use the voipuser.org account, that would definitely be the cherry on top!
One suggestion to D-Link (those guys are definitely nice to deal with): allow the use of the second phone port as a relay for a regular landline so that one phone can be used for every type of call, and make that a simple parameter in the web configuration pages. This way, if the user wants to use two phones or one phone and a fax, or just one phone for everything, this jack-of-all-trades could do it all.
For my use, this router is a keeper and an overall score of 9 out of 10. If D-Link could add that extra feature, it would be a perfect 10.
** – The VoIP Wiki yielded the following info about unlocking a Lingo unit:-
Admin logon : PrimusAdmin
password : tu7w3z39op4n
This for the current (winter 2004) Lingo version, firmware L4.0S34 boot rom L4.0B03. CSR didn’t ask about my version, so this password should work on all. D-Link will probably change with a firmware update if it becomes too popular.
Wednesday, January 28th, 2009
Like many of the major PBX manufacturers, Cisco has firmly embraced SIP, it is currently supported on 4 of their IP phone models, here we review the their Flagship SIP phone the 7960G.
The current model is the 7960G, the key changes from the 7960 is that it is made for the Global market, with symbols instead of ‘English’ labels on the function key, with overlays for each region. The other importance change is that the Power over Ethernet is now standards (IEEE 802.3af) compliant, whereas on the 7960 is it was proprietary.
Anything from £70 to £350 depending if you buy new or used, shop around for the best deal.
From the factory, the phone is supplied with SCCP firmware, as are almost all used ones recovered from offices etc. Converting them from SCCP to SIP can be achieved, and can require a full set Cisco SIP firmware, depending on the SCCP version already loaded, occasionally things can go wrong and the phone can become locked in a loop.
Power over Ethernet (PoE)
The phone supports Power over Ethernet, but beware! the earlier non ‘G’ models don’t comply to international PoE standards (IEEE 802.3af), and connection of a standards based PoE injector or powered switch can damage the phone. You can also construct your own PoE injector. PoE is certainly the way to go at home, no need to worry about mains connections next to the phone, and having two leads trailing from the phone, just one patch lead carrying the power and data.
Like any hardphone, its key advantages over a softphone, is that your not dependant on a PC for telephony, so no issues with applications and handsets/headsets/mics & earpieces, or leaving the PC switched on, with the application running to receive calls. You can have one in every room that you currently have an analogue phone, with just a Cat 3 or better data cable. The audio quality is exceptional, no detectable difference between it and a traditional phone. You have easy access to traditional telephony features (hold, transfer, shortcodes, call log, CLI etc), new ones like xml features such as web based directories, information etc, you can store them on a central server for access by the phones from any location.
Touch & Feel
The build quality is excellent, solid, stylish, well thought out, a quality item, a vast improvement on the first generation Cisco phones.
Large LCD display, local directories, local call log (missed calls, made calls, received calls), call hold, call transfer (blind or attended), adjustable ringer, microphone mute, monitor speaker and handset audio, message waiting indicator. On-board Ethernet switch, to connect you PC via a single LAN port. Excellent handsfree mode, line selective intercom (auto answer), adjustable back-stand, xml services, direct message retrieval button, headset socket, 6 line buttons, user installable ring tones, you chose it, convert it and the phone will ring it.
Once running SIP firmware, they are easy to configure and can be configured from the phone itself, or via a configuration files and a TFTP server.
This is a tricky one, officially you need to pay for support from Cisco to obtain product info, support and firmware updates, it was designed to be deployed in large corporations not for single home users, but look around the Internet.
Very nice phone, if a little on the expensive side, but unlike some hardphones you won’t be disappointed.
Wednesday, January 28th, 2009
VoIP (voice over internet protocol) internet phone service provides a money saving alternative to the traditional land line phone, providing an annual savings of up to 80% on telephone expenses. All that is required for VoIP is a broadband internet connection and an internet phone service plan.
With so many VoIP service providers to choose from these days, the biggest problem most people have is deciding which internet phone company to get service with. Having high quality, reliable telephone service is extremely important, so you need a company that is established and reputable. Another very important issue is price and features – you want to get the most for your money, as long as the quality of service is excellent . Since broadband phone providers all use the same basic technology, selecting the best VoIP provider comes down to looking at some basic information about the internet phone company itself.
Factors to consider when choosing the best VoIP provider:
* The history and reputation of the company. Is the company financially sound? Have they been in the internet phone business for a long time, or are they new at this? Are they on the “up and up”, or are they entangled in lawsuits? The best VoIP providers are profitable, well managed, and highly experienced with the technology.
* The quality of support. Is the company committed to delivering the best possible service? Is their customer service and technical support staff always available, and quick to respond? The best VoIP providers will have 24X7 support and always respond to your issues very quickly.
* The quality of VoIP service. Is sound quality as good as a regular land line phone? Is the “uptime” of service 100% or near this? Does the VoIP provider offer call forwarding when your power goes out or internet connection goes down? Do they offer enhanced 911 (E911) service, which routes 911 calls to a local dispatch office and provides emergency operators with your location when calling? The best VoIP providers are always striving to provide top quality service, and are always proactive when it comes to delivering the latest technologies and service enhancements.
* What you get for your money. For most people, the main benefit of VoIP is getting unlimited local and long distance calling for a low, flat monthly fee. All internet phone service providers offer an “unlimited calling” plan, but the price usually varies from provider to provider. Vonage charges $24.99/month*, Opex costs $44.95 per month for unlimited home business service, Packet 8 charges $24.99/month, FonVantage charges $29.95 to $39.95 per month for business-class service, Lingo charges around $22/month, and comes with free international calling to some locations. VoIP.com charges $16.58 a month (with prepaid annual service) and cable companies charge an average of $39.99/month for the same unlimited calling service.
* With the exception of Vonage and Lingo, who both include free international calls to some destinations in their unlimited calling plan – most VoIP provider’s unlimited calling plan typically includes only the U.S. and Canada.
* VoIP Service Features. Standard features like caller ID, call transfer, call waiting, call forwarding, call blocking, voicemail to email, voicemail, three-way conference calling, etc… are usually included with every VoIP provider’s plan. The type of feature and number of features varies with each provider, but most of the best VoIP providers give you every feature you would ever want for free. Unlike land line phone service (where you have to pay for these “extras”), features are usually free with the best VoIP providers.
Our Picks for the “Best VoIP Provider for the Money”
For really cheap unlimited residential local and long distance calling within the U.S. and Canada, we highly recommend Voip.com internet phone service. VoIP.com and offers the best value for the money for those interested in basic broadband phone service. VoIP.com unlimited local and long distance is only $199 a year (paid annually), which works out to a measly $16.58 per month. Packet8 also has an annual unlimited plan for the same price. Both Voip.com and Packet8 have free softphone service, which makes it really convenient when you travel. With a softphone, you can literally take your broadband phone service with you on your laptop and never miss a call.
For those who want to make cheap international calls, Voip.com has the absolute lowest international long distance rates around.
For unlimited residential, and small business VoIP service, consider Packet8 due to it’s reasonable cost, company stability and longevity, and reputation for excellent “business class” phone service. Packet8 was the first VoIP provider to develop and offer video phone service to consumers, and Packet8 Virtual Office is one of the few VoIP service offerings specifically geared toward small businesses.
For a portable PC to PC or PC to Phone VoIP solution, we recommend voip.com or Net2Phone, depending on your individual needs. PC-based phone service, also known as soft phone or SIP phone service, requires a computer connected to the internet as well as speakers a microphone and a headset. It is possible to purchase a USB phone for use with PC based VoIP service, which makes it a little less inconvenient to use. The big advantage of using a soft phone VoIP service is the portability and extremely low cost. PC phones are not intended to replace land line phones. You should only use this type of internet phone service as a supplement to a traditional phone or cell phone.
VoIP.com offers some of the lowest rates for PC to Phone calls within the U.S., and international calls are super-cheap.
Before you decide on the best voip provider for you, we encourage you to first learn all you can about the providers and plans available to you. Each internet phone service provider offers slightly different plans, options and features, so it’s important to compare VoIP providers against what you need and expect out of your phone service. We only review the best VoIP providers, so you can be assured that the internet phone companies presented on this site all offer excellent service and value for the money.
Pair up your internet phone service with inexpensive high speed DSL internet service from AT&T
It is now possible to subscribe to DSL internet service through AT&T without also getting telephone service through them. DSL is a great cost-saving alternative to cable internet, and it works perfectly with the internet phone service provider of your choice.
Monday, January 26th, 2009
VoIP is the transmission of voice over packet-switched networks.
Traditional voice networks utilised circuit-switching technologies. Effectively, a pair of copper wires is used to connect one party to the other, completing an electrical “circuit”. With packet-switching, traditionally used by IP based networks such as the internet to transmit data, voice signals are broken down into tiny “packets” of digital sampled data, sent and then reassembled at the receiving end. The efficiency in voice over IP is in part down to this packet process. Multiple conversations (as many as 6,000) can be transmitted over one pair of copper wires.
Wider adoption of VoIP has been largely down to advances in IP networking technology, and processor speed increases. The faster a processor can sample voice data into small packets, and the faster those packets can be sent over an IP network, the better and more efficient VoIP becomes.
Cost savings with VoIP
Due to the explosion of broadband in recent years, more and more people have fast internet connections at their disposal. With VoIP, every call you make can be treated as a “local” call – the internet has no concept of international. When you access a web page in the United States from the United Kingdom, you don’t pay an “international” rate – you simply have a flat rate connection fee.
VoIP can be viewed as a similar utility to web access or email – if you’re on the internet with a fast enough connection, you can speak to another party without incurring any call charges whatsoever, as long as the other party is using a device compatible with yours.
Compatibility and peering of VoIP networks
To ensure that two parties can talk to each other using Voice over IP, they need to ensure they can reach each other and are using two compatible clients. A VoIP client is simply an audio device that performs the dialing, sampling and packet sending. VoIP clients come in both hardware and software platforms, the software devices using your PC to do the work and usually used with an audio headset.
In terms of reaching each other, both parties VoIP client need to be talking the same language – using the same protocol.
VoIP Technologies and Protocols
SIP – Session Initiation Protocol
SIP is now the most widely adopted protocol by the telco networks (with Skype as the only exception of note). VoIP User bases all its’ services on SIP as do the larger commercial providers such as Vonage and Sipgate
For more information on SIP see SIP, RTP and NAT
H.323 is now largely redundant, although still used as a backbone interconnect between traditional telco branch exhanges.
IAX/IAX2 – the Inter Asterisk eXchange Protocol
IAX2 has been widely adopted as the internal PBX protocol of choice as a result of the success of asterisk, an open-source PBX application.
Advantages and Disadvantages
While cost-saving is often looked at as the fundamental reason for choosing it as a telephony system, having voice broken down into packets for transmission comes with other advantages. Routing and re-routing of call data is easier, and quicker. Follow-me type clients are easily implemented – wherever you have an IP address (or internet connection) you can receive calls on your number.
On the downside, packet-switched networks have not yet reached the level of redundancy of the PSTN (Public Switched Telephone Network). Reliability of your telephony with VoIP wholly relies on the reliability of your internet connection.
There is a system in place, ENUM, for the translation of traffic destined for a PSTN telephone number to an alternative internet route. The organisation and heirarchy of ENUM as a public service remains a work in progress for the telecoms community.
What Level of Services are Provided at VoIP User
VoIP User was created as a community funded network of users who want to test and experiment with IP telephony. Our services are limited to what we are financially capable of within that remit.
Our servers and PSTN gateway are funded by the use of our non-geographic DID (DIrect Dial) inbound numbers. The revenue we generate on your use of those numbers goes straight into funding our servers and outbound PSTN routing. VoIP User itself is non-profit – we recycle all revenue into user services.
We run several servers – the one running this website, our SIP server (on a dedicated host in Paris, France) and an Asterisk server which we use for additional test facilities such as our echo test server.
Our entire network is based on the SIP protocol, and we have peering arrangements with other SIP based telcos such as Gossiptel, Plus Net and others on the way. Calls can be made from clients connected to our server, to users on those peered networks without a cost to the community.
Calls can also be made to the PSTN (regular landline phone lines) within certain restrictions and guidelines. Calls to the PSTN have to be paid for by us, and we do that using the revenue generated by the community in our outbound “pot”.
We operate primarily on a basis of trust. In order to keep abuse off our network, we have developed an algorithm which we run on our server, which gives each user a dynamic rating based on a scale of -10 to +10. This User Rating is calculated automatically on our server based on calls made, calls received and general contribution to the community including users writing reviews and helping other users on the forum.
Monday, January 26th, 2009
3 phone lines by default, with another 6 configurable lines
7 programmable keys
200 number phone directory, plus a 100 number caller list
Hands-free speaker phone and headset modes
g711 and g729 codecs
XML programming possible
The Aastra 9133i phone is a sturdy mid-range phone, costing between £80 and £100, plus VAT. It has 3 lines by default, which makes it ideal for both the office environment or and the home worker – or anyone else who needs more than one phone line. There are 7 programmable keys which can be programmed to provide further lines (making up to 9 in total) or as speed dial keys.
Unlike some phones, this phone does not have a lightweight plastic feel to it. The phone is very sturdy and stays put on the desk. Any worries I had that I would need to mount the phone on the wall proved to be groundless (wall-mounting is possible by inverting the base). The handset has a nice solid feel to it and sits comfortably in the hand. The three-line LCD is easy to read and can be angled for easy viewing, although the ability to dim the display would be nice.
In the past the phone was sold with PoE (power over ethernet) and required a PoE injector if you wanted to use it on a network that did not supply power. Fortunately (for me), the phone is now bundled with a power supply adapter (but still allows PoE as an alternative). Anyone buying this phone might want to check this point before purchase.
After using a simple, old-fashioned phone, the phone seemed to be oozing features, many of which are present on a mobile phone but nevertheless still took me by surprise. For example, the phone displays the name of the caller if they are in the directory. You can also list the last 100 callers, both answered and missed calls, and if you miss a call the phone tells you – no need to dial 1471 now! I also liked the ability to choose different ring tones for the different lines.
The sound quality is very good, although the default configuration has a considerable side tone (echo). However, once I had changed the settings, this disappeared.
Configuring the phone is relatively straight-forward, although you do need to download the administration guide from Aastra’s web site first. Although the documentation is available on-line, it would have been nice to have it included on a CD in the box. All that is provided is a paper installation guide. That said, I found it fairly easy to find the information I needed on the Internet. Because the phone is one of a range, much of the information relevant to the other Aastra phone also applies.
Configuration can be done either using a web interface, downloading a configuration file to the phone, or directly using the keys on the phone. I used a configuration file. To do this you need to set up a tftp server, which the phone contacts when it is booted up and downloads the configuration file. The phone can also use tftp to download the phone directory and firmware upgrades.
One thing the phone does not do well natively is work behind a NAT firewall; the NAT configuration options are just too limiting. After trying unsuccessfully to get the phone to talk directly to the SIP provider, I had to resort to setting up an asterisk server instead. Once done, the phone worked perfectly. The lack of NAT compatibility is, to my mind, a serious omission. Aastra seem to expect the phone to be on an internal network, connected to the outside world through a PBX. This may be a reasonable assumption for the corporate user, but not necessarily true for the home worker like myself.
My overall impression of this phone is a very favourable one. If you are looking for a sturdy multi-line phone then I would certainly recommend the Aastra 9133i.