I’ve been looking for a softphone for Macintosh that uses IAX, but I haven’t been able to find a good one. The EyeBeam and X-lite uses sip and has stability problems so they are not an option either. So I started to write my own softphone. It’s nice to be programming in Cocoa again. But I’m becoming old, so I have to read up on programming in Cocoa again. I have a working softphone up and running already. Well, sort of. It receives calls and you can answer and talk. But I’ve only been writing on it for two days, and most of the time has been spent on installing a Subversion system on my server. As soon as I get a polished version, I’ll put up a beta program. I’m thinking of calling the program “Breaker-Breaker!” in the old CB-radio spirit. And yes, it will have the old style analog VU indicators for bandwidth and voice volume.

Update: I have released my IAX Softphone client for Mac OS X, You can read more about it here.

As soon as you mention the word VoIP, everyone responds with “Skype?” This immediately gets me into an discussion why people should not use Skype. The fact that your computer is used as a link to route traffic, much like BitTorrent. That it is based on a proprietary protocol, only known by Skype, so no other manufacturers can make compatible equipment without permisson (and probably a big fat wad of cash) to Skype Corporation.
All this means that there’s no connection between Skype and regular SIP phones that is emerging as the standard for VoIP calls. But all hope is not lost.

The Gizmo Project is a program that’s much like Skype, except it’s based on standard codecs and protocols. That means that anyone using Gizmo could call someone with a different provider of VoIP. You can also set up so your Asterisk server automatically receives calls from Gizmo and direct them to their extension. All for free.

All new users get a SIP phone number that can be used from other SIP phones.

So if you already have an Asterisk server, now it should be easy to convince people to try Gizmo out. They can call you for free to your regular phone, and you can call Gizmo users as well.

So I’m going to see if I manages to convert my friends to use Gizmo Project instead. There’s a client for Mac, Windows and Linux.

I helped my friend Micke to set up an Asterisk@Home PBX server today. But when setting up Free World Dialup FWD, we had some problems. First I forgot to enable IAX in the account setup. After that, dialing out worked, but we could not dial in to his FWD number. To connect we use an IAX trunk. We searched for an hour trying to find the problem, and after searching Google we finally found the answer. In the trunk settings for FWD, the incoming user context has to be named iaxfwd. Irritating to say the least. After changing the name everything worked fine.

Asterisk: The Future of Telephony online from O’Reilly Media. They have released it under the Creative Commons license. This means that you can download the book here for free. But it also means that you should buy the book here. Now you have the downloaded pdf version for quick searching, and the nice feel of a paper copy for browsing.

I got my first real Voip phone, the Grandstream GXP-2000 a week ago.

I have two ATA boxes to connect my old wireless phones, but I wanted to try out a real IP-phone. It’s a nice enough phone, feels a bit cheap (which it is), but I guess you can’t get everything for $100. Connecting it to my Asterisk server was a no-brainer. It has a web interface for setup, but you can do basic setup like ip-numbers from the phonepad. It’s great to have an message-waiting led on the phone. Then you just press the MSG button to listen to your voice-box messages. To enable this function, just put *97 in the field “Voice Mail UserID” in the Account page of the web setup. There’s support for up to 11 line indicators (with an additional key-module) but I think I’ll do fine with four. The only thing I haven’t got working is the Conference button, but it’s easy to just transfer the calls to a conference number. You also have 7 quick dial buttons. To configure them, just go to the web setup for the phone and enter the name and number. The sound quality is excellent. Much better than both the Linksys PAP2 and the i3 micro Vood VRG-121 box I have for my old analog phones. The speaker phone works, but isn’t fantastic. I normally just use the speaker phone while waiting for the call to go through, so for me that’s not a problem. There’s a new firmware I’ve just installed that’s supposed to make the speaker phone part work better, but I haven’t had time to test it fully. So would I recommend this phone? Yes, if you need a cheap voip phone that uses SIP I can really recommend this one.

More blog entries about Asterisk

Update:The code below does not work in Asterisk 2.0b4 or newer. I will update with instructions on how to get it to work in those versions. Thanks to Dan for pointing this out.

Update:I have changed the code for the remove part. It now requires a key press to accept the number to delete.

I have added a black list function to my Asterisk@Home server and thought that I’ll share it here if someone is interested. To add a number to the blacklist there is two ways of doing it. The easiest is to dial *32. Then you get a voice telling you the number of the last incoming call. To add it to the blacklist, just press 1. This is great for pesky telemarketers. If you want to add it manually, just press *30. Then you can enter the phone number directly on your telephone. To remove a number from the blacklist, press *31 and enter the phone number. If you don’t like the codes I use, just change them.

Here’s how you do it.
Continue reading →

I found this article that tells you how to get an account that gives you free calls to the US with your Asterisk server. And it works great! I called my old friend Martin and talked to him without any problems. And I realized that this is the first time I called him since he moved to the US. And that’s like 18 years ago. That’s embarrassing. I hope he forgives me. Anyway, it was nice to talk to him.

The service, called GoIAX.com is still in beta, and I wonder how long this will continue to work, but I put two other services as fallback if this stops working. One thing about the Nerd Vitties article, they have the codec setting to GSM, but if you have the bandwidth you can use ulaw instead with better sound quality.