Progress w/ trixbox

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X-Lite Screenshot

Originally uploaded by sogrady.

For those of you that are interested, there’ve been a few new developments in my personal Asterisk installation project. Couple of setbacks as well, but at least it’s moving forward. A quick summary:

  1. TrixBox Installation:
    Complete. Using the aforementioned and linked NerdVittles howto, I’ve not only completed an installation of TrixBox and FreePBX, I’ve set up an account (or “trunk”, in Asterisk-speak) for my new provider, TelaSIP.

  2. New VOIP Provider:
    While I considered integrating my existing Packet8 VOIP service into my Asterisk installation via hardware, I’ve continued to experience the very frustrating call drops where I can hear the caller but they can’t hear me. As discussed before, I gave Packet8 a chance to fix this and they were very responsive, but given that the problem was not solved I’m not inclined to give them a second chance.

    With Packet8 on the way out, I needed a new provider, and for the time being at least I’m going w/ TelaSip. I’ve heard good and bad things about them – often in the same conversations – but ultimately I’m choosing them because they seem to be very responsive to support questions over email. The fact that NerdVittles included them as a default in their howto didn’t hurt their chances either. We’ll see how they perform – I’m already having an issue (described below), but for now they’re the choice. The account is also about $10 per month cheaper than Packet8’s plan.

  3. Number Transfer:
    TelaSip issued me a new number at the time of account creation, but ultimately I’d like to preserve my old Packet8 number. The good news is that the number is indeed portable, so I should be bringing it with me to TelaSip. The bad news is that the porting process – according to TelaSip’s form – is expected to take a minimum of 30 days. Contrast that with the porting process for cell phones, which when I switched from Verizon to Cingular took about 45 minutes. Given how little I use this phone, however – most of my friends and family still use the cell phone despite the fact that it doesn’t work in my apartment – this isn’t a dealbreaker.

  4. Softphone Choice:
    Ultimately, I plan on connecting my Asterisk info back into a standard phone, probably by connecting my existing Uniden cordless to the system using something like this rather than a new digital ready phone. But for now, I’m using what’s called a softphone to connect to Asterisk. Think Skype. Unfortunately, neither Skype nor Gizmo can apparently be used as softphones to connect to my Asterisk instance, although Gizmo can call it because it speaks SIP. That would have been ideal, because then I could just run that and have both business and personal connectivity. But they don’t, from what I’ve been able to determine.

    I then considered Ekiga, the softphone shipping in Ubuntu by default, but it’s agonizingly slow on my machine – possibly because of its Evolution Data Server dependency (my GAIM instance was affected – I think – by something similar). Then I considered Tapioca, but I can’t find Edgy builds for the package and the Dapper builds conflict with some of my installed libraries. So for the time being, I’ve default over to the free but not open source X-Lite. I can’t say that I love the interface – the menus in paticular need a lot of UI love – but it works more or less adequately so far. Definitely open to suggestions here if any of you have them.

  5. What Works:
    TrixBox seems to be operating just fine. All of the typical Asterisk stuff works just fine – I can dial in and check my voicemail, I can create wakeup calls, I can dial 611 and get up-to-date airport weather information read to me in a slightly creepy robot voice, and so on. Pretty neat.

  6. What Sort of Works:
    I was about to write this up as not working, but as it turns out send voicemail to an email address functionality works just fine – Gmail just dropped it in my spam folder presumably because it came from a non-standard domain (determined from an nslookup of my Comcast IP). The catch? VLC can’t decode the WAV file, so there’s something wrong with the creation part. That should be solvable.

  7. What Doesn’t Work:
    Inbound and outbound calls. Yeah, the two things you expect a phone to be able to do. The outbound problem may be a service related one rather than something I’ve misconfigured, because I’m getting the standard telephone error message of “All circuits are busy, please try your call again” which indicates to me that I’m at least connecting to the regular phone network at some point. I have a request into TelaSip to see what they make of it.

    The inbound issue is more problematic, because I suspect the issue there is something that I’ve done wrong. Calls to the number TelaSip has issued me are picked off by their voicemail system, and seemingly never arrive at my Asterisk instance. Not quite sure where I start to fix this, but I’m sure Google will turn something up.

And there you have it – my current progress with TrixBox, Asterisk and VOIP. One of the things I plan on looking into as soon as I can get the basic inbound/outbound calling problems solved are the follow me features. Being able to have friends or family routed to specific numbers – say work during the day or cell while I’m travelling – would be quite nice. It’s easy to do with internal extensions, I’m not quite sure how to do it with external numbers yet. But I’m sure someone’s figured it out.

Any questions, suggestions, or corrections, drop a comment.


  1. You’ll learn to love the asterisk CLI. Run ‘asterisk -vvvvr’ and you’ll get a prompt with debugging info turned on, so you can see the progress of each call. ‘sip show registry’ will show if you’re correctly registered to TelaSip – I’m guessing not, since neither inbound nor outbound works. ‘sip reload’ will force an attempt to register so you can watch its progress. apt-get install sox to transcode voicemail.

  2. i’m always a big fan of the command line, so i bet you’re right. you’re also correct that i wasn’t registered properly. the Nerd Vittles howto left out a portion of the correct SIP gateway. since correcting that, inbound is now working.

    still no joy on outbound, tho.

  3. Since you’ve flooded off IRC, can I suggest using a pastebin?

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