Hello 2012!! The most >Geek< fun I've had in a while

ok ok, so this is really not at all keeping up to trying to post regularly. Got too busy, had a baby, etc.. excuses.

Now, I have a great reason to post!! I just finished a little pet project, tiny one.. but I’m quite excited about it as it’s something I’ve tried to do over 10 years ago, but the products were bad. Now, with newer hardware and software it’s a reality!

I had just succeeded to build my home SIP infrastructure. With a SIP client on my iPhone, I can connect to my personal PBX server, and make calls to regular phones via a SIP to PSTN router.

You are lost? I just built my own VOIP system, which I can make data calls over the air, from my iPhone, connect to my home based VOIP system, and call to a land line. What’s so big deal about that? Here in Singapore, the digital land lines are completely toll free! All outgoing LOCAL calls are free. With 3G and 10GB of quota a month, I can make a lot of phone calls this way!

I will share how I did it… and boy, it’s not easy. This setup is not for the faint hearted to even attempt to consider. Leave it to the real IT geeks *cough*

Here’s a glimpse on what you’ll need.

  1. Home broadband connection which has preferably unlimited usage (as we get here in Singapore)
    +the broadband connection must always be up (otherwise you cannot connect from outside of your home)
  2. Home wifi  network, if you want to make calls via your SIP client in the smartphone
  3. A Linksys SPA 3102 (retailing for about S$100 in Singapore) – this is a VOIP Router
  4. Asterisk – a free PBX Server – run this on an old PC, or a small atom machine, or a virtual machine
    (I run mine as a VMware Virtual Machine)
  5. A SIP client for your PC and/or smartphone
    PC – A good and free SIP client called X-Lite (http://www.counterpath.com/x-lite.html)
    iOS – many choices available, free and paid I use the free client from 3CX (http://www.3cx.com/VOIP/voip-phone.html)

Generally, each item is easy to get up and running, but to get them working together was not easy, especially when you don’t understand half the terms involved. This is one of the craziest situations where I was looking at pages of configuration items and they are mostly abbreviated.

In addition, there are few up to date guides on the net which tells you what to do. I found lots of guides for Asterisk, some very good, but are for older versions. I’m using version 1.8, and there are lots that are now redundant. After all I’ve done to get things to work, I must say it’s actually not a lot of settings we need to make.

Credits will be given when due. I will link to the sites which I found the most useful information.

 

13 thoughts on “Hello 2012!! The most >Geek< fun I've had in a while

    1. Jason Post author

      in terms of dial/busy tones they are already pre-configured, and it sounds like the US standard tones. There’s a whole lot of customization I can see that is possible, but I didn’t bother to change anything here. Off the top of my head what I recall is that I had to configure an IP, SIP dialing rules and the Asterisk server details.

      Reply
  1. Tan

    In that case, it would look pretty simple to configure the box. Will be trying it out soon. Read in other forums, there are people who had difficulties getting it to work with Digitial Voice.

    Reply
  2. Sean

    Hi,
    If you use those free SIP service provider, you no need the Asterix server and PC runnung the SIP client, all you needed to achieve this are.
    1. A 3G smart phone, iphone or android phone or windows phone with 3G data plan.
    2. A linksys SPA3102.
    3. Home telephone line.

    Reply
    1. Jason Post author

      thanks for sharing Sean. I’m not aware of the free SIP providers. Can you please share who they are, and how can we make use of the service and the SPA3102?

      Reply
  3. Sean

    Hi Jason,
    Thare are many.
    For example in Singapore pFingo.

    Let’s say we use pFingo.
    First U need register with pFingo at http://www.pfingo.com with two account.(say alpha and beta).
    At home on SPA3102 you use alpha for SIP registration.
    On your Phone ( 3G mobile), you can download the pFingo sip client and register it with beta account.

    Now you can call for free between alpha and beta each other.
    And with your SPA3102 connected to PSTN line, you actually can use your Mobile phone with 3G connection and call SPA3102 and with proper configuration on SPA3102, you can forward ( second dial) your call to any number via PSTN line.

    Reply
    1. Jason Post author

      Hi Sean, I just had a good moment to digest what you suggested, makes good sense. Which eliminates the need for asterisk for some use cases. Thanks for sharing!!

      Reply
  4. Jon

    Bought my spa3102 from sim lim a couple of months back. Been struggling with the echo ever since. But copled with an asterisk box, it’s worth it.

    Reply
    1. Jason Post author

      thanks for the comment Jon. I do notice echo from time to time as well. I haven’t done anything to fix it, but I’ve read around and it seems to be related to the volume amplification to be too high. You can try playing around with the parameters on the SPA3102, specifically the ones that adjusts the volume.

      Reply
  5. Jon

    Hi Jason, I finally got the time to sit on my issues with the SPA3102… I figured no matter how much setting I change on the gains, it’s never gonna work. I got a good break when I tried g711a as the primary codec (default is g711u) and it worked wonders and I’m now happily without echo (sometimes a barely noticable and faint echo).

    I think we should exchange configurations so we can get the best config from all and probably eliminate the echo once and for all.

    Reply

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