Domain: grandstream.com
Stories and comments across the archive that link to grandstream.com.
Comments · 23
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Re:Cell phones are usually tied to a person
Grandstream is good for "cheap" phones of acceptable quality. They just recently announced this:
http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/gxp2200
Although it might sound nice to have the whole interface be a touch screen, I think that the hard-keys for dedicated functions end up improving the usability of the device.
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Your best/cheapest option is an ATA with a router
Your best option is to get a SIP ATA (analog telephone adapter) that has a router built-in. I have personally used a Grandstream 486,and they work great. Vonage uses SIP and I have read (but never tested myself) that you can use any SIP compliant device with it. The difference between Vonage's ATAs and others' like Grandstream's is that Vonage's are locked down to only work with Vonage.
So you would go from your DSL/Cable modem to the ATA/Router then to your Wireless/LAN access point or switch. If you would prefer to still use your wireless router for everything you could set it up in a DMZ on the ATA and put the ATA on a different subnet than the wireless router. -
SIP is the *open* and *free* alternative
You don't know the alternative and call yourself a geek? Or, maybe an AC is no a geek!
The alternative is to use SIP phones. And then if you don't like one provider, you get another. For example,
http://les.net/
is one provider I've had experience with. But you can get lots more if you want,
http://www.sipcenter.com/sip.nsf/html/Service+Providers
With SIP you can use ANY provider and not waste money on substandard service. Heck, with SIP *you* can be your own provider with Asterisk PBX software.
There is probably more real phones available for SIP than the proprietary protocols like Skype,
http://www.grandstream.com/products.html
Very good phones from my own experience. Skype has been an obsolete VoIP solution for years now. Anyone seriously looking for a flexible VoIP solution, will only look at SIP. -
Make your own Linux-based PBX system
We did it ourselves and saved >$100/month for a small business. Just use Asterisk (free and open source), buy some inexpensive but full-featured phones like the Grandstream GXP-2000 (about $80 each), and get a termination provider like VoicePulse Connect for Asterisk ($11/month for four simultaneous channels, free incoming, and below $0.01/min for most outgoing). It took some work to get it all set up and working properly, but now is actually more reliable than the analog phones ever were. (We had phone company issues every few months... just awful.)
--
Educational microcontroller kits for the digital generation. -
Re:So, how do you tell your clueless neighbors?
One way is to ignore it, because it's not your problem.
Another way is to point out gently that it's a problem. Except then, you have made it your problem; and you can expect to be treated like a free 24/7/52 helpdesk forever from then on. Or treated as though it was your fault that it wasn't secure.
Yet another way is to set up your a router of your own, with broadly the same settings as theirs, but with a proxy configured to do something like this. But don't switch it on just yet. Then, while their network is idle, disable their router (remember the password .....) and enable yours. The only thing that could possibly be more phun than this would be listening in on their frantic phone calls to their ISP's support hotline. And, with the appropriate equipment, you could even hi-jack their phone wiring ..... but that's a little bit much to expect anyone to survive! -
Re:Better yet...
Someone's been reading this, haven't they?
:)
If / when I ever get any wireless kit, I will change the name of my neighbours' unprotected router (currently set to the make and model name; a quick Google search revealed the default password) to "pWn3d", have my router emulate theirs but with suitably distorted graphics, and see what happens. Jut a shame I can't listen in on their call to tech support ..... but I could, if I had what fone phreaks once referred to as a "Sky Blue Pink Box with Yellow Spots On". Oh, wait, such a thing already exists!
Now, that does sound like serious PHUN! -
Re:IAX?
A few months ago I tried to attach my Mother-in-law (who lives overseas) to my Asterisk server. I bought a SIP analogue telephone adapter and went to visit. I spent at least 2 hours a day for 8 days trying to get it to work. I fiddled with every setting on the ATA and (via an SSH tunnel) loads of settings on my router and I never got the ATA or any SIP softphones to work. Finally, I tried an IAX2 softphone and it worked first time.
So, when I returned to the UK I bought an IAX2 phone for the same price as the ATA and sent it to Estonia. My Mother-in-law plugged it in and it worked first time. Now she and my wife chat for hours without it costing me a penny.
Conclusion: SIP works fine on a LAN (where I now use the above ATA), but if you have any NAT anywhere in the link then the pain of making SIP work isn't worth it when you can get IAX2 kit for the same price and it "Just Works".
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Man in the Middle attack
The bad part? The Moviebeam player also requires a connection to a phone jack -- every fortnight the box dials a toll-free number in the middle of the night to tally how much you've spent on movies so far, for the benefit of your monthly statement.
Am I the only person who thinks this is going to be spectacularly easy to hack?
You will need one of these handy little gadgets plugged into your PC, a copy of Asterisk, and you're almost good to go. Just convince the Moviebeam player that your PC is the Moviebeam central office. It'll phone through and report your usage. But your PC isn't the Moviebeam central office, so no bill will be generated. You may also have to get your PC to call the real Moviebeam central office and report no usage.
Old-timers will have heard of various coloured boxes in connection with the phone system: Black Box {free incoming calls}, Blue Box {in-band signalling generator}, Red Box {payphone coin-insertion signal generator}, Beige Box {croc-clips to phone socket adaptor} and so on. More esoteric ones included the Jade {timer to avoid itemised bill threshhold}, Primrose {phone-line powered battery charger} and Violet {line holding circuit, defeats money-run-out on some subscriber-owned payphones} Boxes {all the good colours were already taken by the time they were invented}. But this setup truly is the fabled "sky blue pink box with yellow spots on"! -
Re:Not enough upload
I'm not sure how they can claim that their service is broadband if it doesn't have enough bandwidth for VoIP. G711 uLaw only requires 107Kbps including all overhead source (pdf file).
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Why draw the line at half-assed?
When you can get an Handytone http://www.grandstream.com/y-htseries.htm/ or IAXy http://www.digium.com/index.php?menu=product_deta
i l&category=hardware&product=S101I/ type device anyway? I mean, really, get a job, these things only cost $100.... Oh, yea, and feel free to start the ususal rant about proprietary systems and how much they suck......it is skype right? Or is it only lame when proprietary isn't free? -
Re:Cheaper/better FXO/FXS from Grandstream
This sounds like an early production version. My HT-488 arrived several months later than projected due to production problems with the early devices. Furthermore the firmware was withdrawn at http://www.grandstream.com/ for more upgrade work. I think they do a good job of standing behind their products. I also think they're pretty embarrassed by the BudgeTone line of phones, which are pretty bad, but the ATA devices are top-notch in my book.
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Cheaper/better FXO/FXS from Grandstream
There is a cheaper/better FXO/FXS from Grandstream, the Handytone 488. This is a new item and can be bought for under $90. It is extremely small (a little smaller than the SPA-3000) and handles all the popular codecs. Its configuration is a little easier to understand than the huge Sipura menus. It works right away without SIP registration (Sipura needs a setting in order to work without SIP registration) which allows you to test it by placing calls to IP numbers directly.
Sipura units seem to have much more provisioning support but Grandstream supports the same provisioning protocols. This can help with large deployments where you want to automatically assign extension numbers from a central server.
Again, this a new product that just went into production and might save you a few bucks over the Sipura in quantity. See http://voipsupply.com/ and http://www.grandstream.com/ for some more detail.
Kris -
My VoIP experiences are so far positiveI've seen a lot of posts about the various experiences, VoIP isn't ready for the end user, etc. I agree with the end user bit, but VoIP is certainly ready for and should probably be exploited by the geek community. Here's my setup and situation:
- Internet: Mediacom 3Mbps down, 256K up cable modem. Quite reliable, down probably for 10 minutes a month, maybe less. About $45 for that.
- VoIP Provider: BinFone Service through Binhost Technologies, a company I'm a part of. We're small but we know our shit, we're cheap, and we have geeks running the entire show. We are more into reselling VoIP but also do individual IAX and SIP accounts. Rates are $0.03/min for USA, $0.05/min for Australia (wife is Australian, we call there a lot). More info here.
- Phone: Grandstream Handytone 486 SIP phone adapter. A very cheap ($65, I believe) phone adapter, but has a web interface, good features, and does what I need it to. It is plugged into the network via CAT5 and into the phone patch block via standard POTS wire.
- IAX Server: I run my own IAX server (Asterisk) in-house. It talks to Binhost's server through the IAX protocol (Asterisk proprietary) which is very efficient. I have an X100P FXO PCI card in it that allows connection to the PSTN (my landline) and a NIC to talk to the network.
- Firewall: All of this sits behind the firewall, a simple Pentium 233 running Slackware 9.1 and using iptables and QoS scripts to regulate traffic. The QoS designates packets by the MAC address of the Grandstream as highest priority so my VoIP packets always get through quickly.
The phones in the entire house are connected to the phone patch block through the patch panel and a 66 block. The VoIP adapter is also connected to the phone patch block as well as the network. The Asterisk box is connected to the network and to the PSTN landline. So. When I pick up a phone (any of the three in the house), I simply dial a number. The signals from all the phones run through the Grandstream VoIP adapter to the Asterisk box. The Asterisk box figures out if it's a local call or long distance. If local, it uses the FXO card to send out the call on the PSTN. If long distance, it communicates via IAX to the Binhost server and places the call over the Internet. No intervention is required on my part as to where it goes, it just does it right.
If the Internet connection is down or otherwise inaccessible, it automatically falls back to the landline so calls can still be placed.
The end result is that I get much cheaper phone calls than I would if I used my long distance on the landline (7 cents US/12 cents Australia vs 3/5), yet I don't have to inconvenience myself with having to worry about which phone I have to use for a phone call.
Incoming calls are received by the Asterisk box. Assuming I haven't turned on call forwarding or do-not-disturb, it rings through the VoIP adapter to the phones in the house. If nobody answers, Asterisk picks up the line and gives a message and allows the user to pick either my or my wife's voice mail box and leave a message. Very handy.
Costs:
Monthly VoIP service: About $20 for the calls, $5 for the line.
Internet: $45/month
Asterisk: Free
Asterisk server: Free donation
FXO Card: $15 on eBay
VoIP Adapter: $65
Wiring: out of some old box
Firewall: Free donation as well
Landline costs: $17.95/month
So total? $80 in startup, $87.95 monthly for all my phone calls and Internet service. I call that a *deal*. -
Re:really missed the point
Already built. Grandstream makes the HT486. Plug a phone in one port, one port goes to your cable modem, other port is NATed to your local lan.
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Works fine at my house
My main phone line comes over a 6.1 mile 802.11b link. I use Asterisk PBX with the IAX protocol to bridge the calls.
And my Grandstream SIP phone works great attached to a Linksys WET-11 client bridge.
And my Ipaq runs IAXComm just fine over it's wireless card to use as a netphone.
Does the battery life suck... yes... does it work and show promise... YES!
Just because people have problems with these cheap (as in quality)(usually SIP or H.323 based) piece of crud phones doesn't mean the technology and possibilities are not still there. SIP is VERY prone to problems from NAT (which many wireless networks use of course).
Anyways... for my 2 cents though I say... just give it time. -
Re:Unlimited Long DistanceAsterisk, X100P "voice modem", NuFone for dirty-cheap calling and Vonage for North America wide calling.
NuFone is good for outgoing long distance calls. They charge in 15 second increments to many numbers (others are 30 or 60 seconds) and are pretty darned cheap compared to other providers.
I have great luck with Vonage for my local calling (North America, flat rate is like, $45 p/m and gets you all the dandy doodads). I also have Asterisk setup to receive faxes and Email them to me, so far no corrupted pages at all and the bandwidth usage is pretty reasonable.
I have this setup on my Asterisk box (Vonage attaches using an X100P card ($100 from Digium for the real-thing, clones have been spotted for cheap including $0.99 but YMMV), NuFone is native IAX).
Cordless phone is attached using a Grandstream Ata-286, so I can wonder around the house with a cordless headset whilst talking to who-ever using VoIP.
and don't forget to register your number on e164.org, for native voip
;)This is an incumbment free zone
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Working Solutions
Setup an asterisk pbx server, and signup with any number of VoIP providers who support G.711 codecs (like Voicepulse or their no bells service, Voicepulse Connect service). Plug your fax machine into a TDM400p card from digium.
Another option, pickup a Grandstream HandyTone 286 (from here for instance) or a Sipura SPA-2000 (from here for instance) (SIP devices, plug a regular phone, or fax, into it) instead of the asterisk box, but it gives you less flexibility. Both devices would work with the Voicepulse services, or most any other true SIP based VoIP service.
This works, been able to fax to people over Pulver's Free World Dialup service without any problems using both types of setup.
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Re:It has already started
You mean like:
SIP Phone
or snom
or Grandstream
or Pulver
and that's just naming a few. -
Re:It has already started
Actually there are several voip phones. There are even terminal adapters that will let you use your existing phones on a voip connection (cisco ATA-188 for example). Grandstream technologies [http://www.grandstream.com] offers several low cost options for the home user. We use the Cisco 7960's at work with asterisk on the backend doing voicemail and handling call routing etc. It's been an excellent experience for us.
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Just got mineOrdered them right away, to have something to play with. They arrived this morning.
They basically work as advertised; pick up the receiver, punch in the number and you're connected within a sec or two (considering that the call is routed from Germany to San Diego and back, not too bad). Voice quality is OK.
It seems SIPphone is just reselling preconfigured units from Grandstream; they have a full manual online, as well as a current firmware image for your phone to boot off. SIPphone apparently did not customize the firmware, or lock any of the settings, so you can do whatever you wish with the phone.
As I suspected (but the SIPphone FAQ or docs don't mention), the phone has a built-in web-server for configuration purposes, with an easy-to-guess default password; so if you're going to put this phone up on a public IP, make sure you have a decent firewall in front of it.
The SIPphone directory just works; no frills, but works.
As my employer has two offices at opposite ends of the country, we're probably going to get a couple more, and look into open source gateway solutions. We've wanted to do that for quite some time, but we couldn't find cheap phone to try this with, so this offering almost perfect for us.
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Re:It's SIP service, silly
I would think that this means a SIPPhone could call someone using Microsoft's Messenger on Windows XP. However, I was not able to confirm this with a breif perusal of the SIPPhone site, and they also state this only works with other SIPPhones.
Check out the product spec from the manufacturer.
The SIPPhone page states the make and model.
Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products (e.g., MS Messenger, Cisco IP phone and gateway, etc)
Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length
G.711 is the granddaddy of the voice codecs. It doesn't say it uses H.323, but I'm guessing it does, seeing as it interoperates with cisco and msn messenger voip.
You can probably even use a different directory service than SIPPhone.com's ; the phone has a web interface for configuring it. -
Re:If it can only call similar phones...You're right -- if it can only call similar phones, it's doomed. That's exactly the kind of truism comment I'd expect from here -- have you ever actually used VoIP?
Here's why this might be reasonably successful:
- What they are selling is a Directory Service + Grandstream phones which support the SIP protocol -- which is *the* standard for VoIP signalling -- oh, which is also supported by the Cisco ATA 186, Cisco 7960, MS IM, X-Lite, Asterisk, etc. -- i.e. basically anyone playing in the VoIP space who doesn't have a legacy H323 or proprietary protocol already deployed.
- They've already got an interconnect agreement with FWD which has circa 40,000 users signed up. (albeit not fee paying)
- The phones aren't locked to being used for this particular service -- nothing to stop you taking the phones and pointing them at FWD/your own Gatekeeper etc. (Refer: Michael Roberton's comments)
They've also had the smarts to set their SIP phone numbers as a "US area code" (don't know if they've actually been allocated it, who knows) -- no doubt PSTN access is in their plans at some stage like Vonage.
Doomed? I doubt it. While nothing here is revolutionary, the genius is in offering the total package (phone + directory service) for a one-off fee that even your grandmother could figure out how to use. All they need to add is PSTN access. If you'd like to learn some more about VoIP, I'd suggest FWD is a nice easy learning curve.
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Re:VoIP
I couldn't disagree more. In my research, I found that VoIP PBXs, even when putting in a new system from the ground-up, were not worth it. They all have voice quality problems (like echo), and the IP phones are much more expensive than wiring your building for Cat.3.
Voip phones are down to $75. That's a lot less than any proprietary mini-PBX phones that I'm aware of.
If you're having echo or other voice quality problems in this day and age, then you haven't configured things properly.
By using Voip phones you also save on admin hassle for moves - people just bring their phones with them to their new desk, and their extension follows... Even if they move to another building, or, if you choose to allow it, to their homes and hotels. Some of the proprietary non-voip systems do the former, but none of them do the latter.