Free VOIP calls with my Android G1 and wifi, google voice and gizmo5

I am so happy, I canceled my T-mobile contract and now I use my G1 to make free VOIP calls using gizmo5 and google voice. I have a wireless connection while I’m at home and at work so thats like 90% of my day. At times when I don’t have a connection, the phone calls get forwarded to my prepaid t-mobile card which is good for 1 year. I paid $100 for it for 1000 minutes.

Things needed before you get started:
1. Sign up for a google voice account and set it up. You can purchase an invite from ioffer.com for like $2.
2. Gizmo5 account. you need to download the desktop client and register. Once you have your sip number (something like 747*******), write it down.
3. SIPdroid app for android
4. Unofficial GV app for android
5. Optional: prepaid sim card

Log in to your google voice account and under settings you can add a phone where you would like your GV number to forward to. Make sure you select type “gizmo” and add the 747 sip number. GV will ask to verify the number so you have to make sure you are signed into your gizmo.

On your android, open sipdroid and enter your account settings.
the server is proxy01.sipphone.com
port is 5060
protocol: UDP is better
You will get a green dot when connected.

Open the GV app (make sure its the unofficial one) and enter your account settings. select calling method as ‘call back’. call back number is your sip number. enter it and enter your pin.
* If you choose the ‘dial out’ method, you will need credits to place a call with gizmo5. the ‘call back’ method tells google voice to dial your number and then once connected with you, call the other number. (This is better because incoming calls are free)

Now you can make and receive calls free while your on Wifi or 3g. In case you want your offline calls forwarded to your prepaid phone, you can select call hunting feature in your gizmo5 account settings.

Enjoy!

27 thoughts on “Free VOIP calls with my Android G1 and wifi, google voice and gizmo5”

  1. The writing here are great. Thanks for having them. I love reading blogs about VoIP! It’s such an exciting technology. I don’t comment on many blogs but had to on yours. Thanks again – great site!

  2. Great but this offer only works for 3 mins, not more. However, there are offers which can get you unlimited Long Distance even Free phone calls to Countries such as India, Pakistan, Bangladesh and Canada too. There are many blogs where you could find this info.

  3. Do you ever experience any kind of call-quality issues? Every now and again I get a 1 second delay when using my ATA setup and wondered if it’s the same for a Mobile setup.

  4. Also, how does this effect the caller id? I assume that all incoming calls correctly display the information of the person calling, but do the outgoing calls display your Gizmo5 number or your Google Voice number?

  5. So what does the process look like when you make a call? What about receiving a call?

    I’m looking to see just how seamless the integration is, or if there are certain steps you have to take to answer/place calls.

    Thanks for all your help HydTech!

  6. I’ve had a Grandcentral(google voice), Gizmo voice and Sipura ATA setup for quite a while.

    The latency is pretty bad so I’d advise anyone to test out their setup before ditching their previous service.

    I test simply by calling my GV from a land line and picking up the VOIP lines.

  7. Hi, I’m trying this on the G1 running cyanogenmod 4.2.5

    I’ve setup sipdroid and I’m able to make calls from the sipdroid app.

    I have installed the GV (Unofficial) app, and set that up too, but I cannot make calls using that app.
    Here are my settings
    Calling Method: Call Back
    You Callback Number: my gizmo5 747 number
    voicemain PIN: my gv voicemail pin (I dont see an option to setup a pin @ gizmo5)

    When I try to make the phone call using the gv app, it force closes and of course, no joy.

    Can someone help?
    Thanks

  8. unfortunately, this hasn’t been working with cyanogenmod 4.2.5 for me either. evan charlton said he woud fix this problem but he hasnt released any update patches yet. later on i will be reverting to 4.2.4 and see if it works then.

  9. So is the integration seamless? Meaning when you receive a phone call, it just rings, and when you place a phone call, you just dial? No hoops to jump through?

    On my ATA home setup, incoming is perfectly normal, but outgoing calls take about 20 seconds before I’m connected and begin ringing the other party. Just wondering if it’s similar to that in this mobile setup.

  10. Impressive piece of information, let me elaborate more on VoIP. Voice over Internet Protocol has been around since many years. But due to lack of sufficient and affordable bandwidth it was not possible to carry carrier grade voice over Internet Protocol. But since the arrival of low cost internet bandwidth and new speech codecs such as G.729, G.723 which utilizes very low payload to carry carrier class voice it has recently been possible to leverage the true benefits of VoIP. G.723 codec utilizes only 6 Kbps (Kilo Bytes/sec) which is capable of maintaining a constant stream of data between peers and deliver carrier grade voice quality. Lets put this way if you have 8 Mbps internet connection, by using G.723 codec you can run upto 100 telephone lines with crystal clear and carrier grade voice quality. I am also a user of VoIP and have setup a small PBX at home. Since I have discovered VoIP I have never used traditional PSTN service.

    Dear readers, if you have not yet tried VoIP I suggest that you try VoIP technology and I bet you will never want to use the traditional PSTN phone service ever again. VoIP has far more superior features to offer which traditional PSTN sadly cannot offer.

    Also It has recently been possile to carry Video alongwith VoIP by using low payload video codecs. I cannot resist to tell you that by using T.38 passthrough and disabling VAD VoIP can carry FAX transmission, but beaware FAX T.38 passthrough will only work when using wide band protocols such as G.711, a-Law and u-Law.

    By using ATA (Analog Telephone Adapter) which converts VoIP signals into traditional PSTN you can also using Dial-up modems to connect to various dialup services. I wont go in to the details what VoIP can offer, to cut my story short VoIP is a must to have product for every business and individual.

    How VoIP Works

    When we make a VoIP call, a communication channel is established between caller and called party over IP (Internet Protocol) which runs on top of computer data networks. A telephony conversation that takes place over VoIP are converted into binary data packets streams in real time and transmitted over data network, when these data packets arrive at the destination these are again converted into standard telephony conversation. This whole process of voice conversion into data, transmission and data conversion into back voice conversation takes place within less than few milliseconds. That is how a VoIP is call is transmitted over data networks. I hope that now you understand basics of how a VoIP call takes place.

    What are speech codec’s and what role codec plays in VoIP?

    Speech codec play a vital role in VoIP and codec determines the quality and cost of the call. Let me explain you what exactly VoIP codec’s are and how they work. You may have heard about data compression, or probably you have heard about air compressor which compresses a volume of air in enclosed container, VoIP codec’s are no different than a air compressor. Speech codec’s compresses voice into data packets and decompresses it upon arrival at destination. Some VoIP codec’s can compress huge amount of voice while maintaining QoS which means use this type of codec will cost less because it will consume just a fraction of data network. Some codec’s are just not capable of encoding huge amount of voice they simply consume huge amount of data networks bandwidth hence the cost goes up.

    Following is a list of VoIP codec’s along with how much data network bandwidth they consume.

    * AMR Codec
    * BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
    * GIPS Family – 13.3 Kbps and up
    * GSM – 13 Kbps (full rate), 20ms frame size
    * iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
    * ITU G.711 – 64 Kbps, sample-based Also known as alaw/ulaw
    * ITU G.722 – 48/56/64 Kbps ADPCM 7Khz audio bandwidth
    * ITU G.722.1 – 24/32 Kbps 7Khz audio bandwidth (based on Polycom’s SIREN codec)
    * ITU G.722.1C – 32 Kbps, a Polycom extension, 14Khz audio bandwidth
    * ITU G.722.2 – 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
    * ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
    * ITU G.726 – 16/24/32/40 Kbps
    * ITU G.728 – 16 Kbps
    * ITU G.729 – 8 Kbps, 10ms frame size
    * Speex – 2.15 to 44.2 Kbps
    * LPC10 – 2.5 Kbps
    * DoD CELP – 4.8 Kbps

    Switch to VoIP Today and you will never want to use traditional PSTN ever again.

    Thanks

    -Imran

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