VOIP

Asterisk and the Mystery of the Asterisk.pid File

I've been fighting with Asterisk the last few days. I think now I have almost tamed the beast though. At first I was running into problems with the web interface. However, at the time I was running the packages that come with Ubuntu 8.04 server. I decided that it would just be better to just compile the latest stable version on my own.

This is fine except that the latest stable version's config files don't actually work. When I started Asterisk I would get an error about how it could not create /var/run/assterisk.pid. This, inspite of the fact that after looking at /etc/init.d/asterisk, it was telling asterisk to create the file in /var/run/asterisk where the asterisk user and group had full rights. Bizaar.

After leaving it for the weekend (My family went to go visit Grandma at her new house) I took a fresh crack at it this morning. It sfunny how when you leave something for a few days you get a whole fresh new look at things.

I decided that even with hacking the /etc/init.d/asterisk file to hard code the startup command to ensure it was passing the right parameters and getting no where, that the /etc/asterisk/asterisk.conf file must be the culprit. 

In the asterisk.conf file that the make command creates, it just housed one section called [global] with a number of directives for where Asterisk should find and create files. 

Well it turns out that this is horribly wrong, and people know about it. Lovely.

I changed the file to look like this instead:

Playing with Asterisk

I've decided to go even cheaper than Vonage. Don't get me wrong. I like Vonage. But I found this article on using Asterisk with the Broadvoice service, and I just couldn't resist anymore. I already had plans to setup an Asterisk server for the twice monthly podcast I run over at http://lordsoftyr.com. But having my own voip PBX system and being able to reduce my current $50 per month Vonage plan (Voice and Fax line) down to $20 per month is just too good to pass up.

Now all I have to do is get it setup properly! The Ubuntu repository binaries are a little old so I'm working on a script that will download and install all the bits that I need in a relatively automated fashion.  I'm running into a few compiling issues at the moment, but as soon as I work those out, I'll post my install script. Until then... check out that article on Asterisk from above. There are so many things you can do with it. I can't wait to start podcasting with my crew using it.

How I Use Open Source and Standards to Create Podcasts

Summary

Tools used: 2 Kubuntu 7.10 PC's, Gizmo VOIP client, Gizmo VOIP conference feature, Icemat Audio headset and usb card, Audacity audio editor, Podsafe Music from http://music.podshow.com,

Process:

  1. Use only SIP compliant VOIP clients (iChat, Gizmo, Ekiga) so that everyone can use the Gizmo VOIP conference call number.

  2. Use two Gizmo clients on your end. One to participate in the call and one just for recording. The PC that is recording with Gizmo should have its speakers and mic turned off. Otherwise strange echoing or feedback will occur.

  3. Everyone call in using the full phone number that you get from Gizmo (It should look like 1-222-xxx-xxxx). Using other methods doesn't seem as reliable.

  4. Record the call. Everyone will hear a message that says that the call is being recorded. Make sure that at the end of the call everyone is off mute and you record about 10 seconds of “silence”. This is for later on when you use the noise filter feature in Audacity.

  5. Hold your podcast session

Syndicate content