Akom's Tech Ruminations

Various tech outbursts - code and solutions to practical problems

Asterisk Getting Meetme to work in Asterisk 1.6 on Ubuntu Lucid

Posted by Admin • Tuesday, November 15. 2011 • Category: Asterisk
I'm assuming that you are attempting to use the Meetme() application and getting an error like this:
WARNING[10695]: app_meetme.c:1097 build_conf: Unable to open pseudo device
It's rather simple:
$ apt-get install dahdi dahdi-dkms dahdi-linux  #I think dahdi may be enough
$ echo dahdi_dummy >> /etc/modules
$ modprobe dadhi_dummy
$ ls -l /dev/dahdi/pseudo
crw-rw---- 1 root dialout 196, 255 2011-11-15 11:26 /dev/dahdi/pseudo
$ id asterisk
(Should include 'dialout' group)

Yay! Apparently in Asterisk 1.6 the timing source for the Meetme application is no longer ztdummy (fortunately I was unable to use m-a to compile zaptel) but rather relies on DAHDI, not only for timing but for conference mixing - see http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/

Asterisk Asterisk and obfuscated SIP port redirection - calls drop after 20 seconds

Posted by Admin • Tuesday, October 5. 2010 • Category: Asterisk
One of my asterisk setups got attacked recently by a brute force script kiddie. The attack is simple:
  1. Try sequential sip usernames from 1 to 10000, hoping that the admin chose a numeric username scheme
  2. If asterisk returns anything but a 401, then perhaps this is a real account, add it to the list
  3. If anything is on the list of "possibles", then brute force their passwords


Why this works:
  • Asterisk has no delay for failed logins, so a brute force attack is viable, hundreds of attempts per second are possible. Not sure what can be done about this
  • The admin didn't set alwaysauthreject = yes in sip.conf. It's off by default, why??? With this enabled, there is no way to tell if a username is valid from the reject response


To deal with this:

Continue reading "Asterisk and obfuscated SIP port redirection - calls drop after 20 seconds"

Asterisk My Not So Great Experience with Grandstream HT-502 ATA

Posted by Admin • Friday, March 13. 2009 • Category: Asterisk
I got 3 of these Grandstream HT-502's for a client's Asterisk setup, and I picked Grandstream basically because they were cheap (client on a budget) and I found no bad reviews. They were ordered February 2009.

First impressions: They are tiny, like two EZ-Passes stacked on top of each other. They are cute, and have all the features you'd expect or even desire in such a device: Dual WAN (routed or bridged, with port forwarding), two FXS ports, some status LEDs, remote configuration and pull provisioning, syslog support, etc.

Well that was the good news...

Continue reading "My Not So Great Experience with Grandstream HT-502 ATA"

Asterisk Data Warehousing and Metrics with Asterisk PBX

Posted by Admin • Sunday, February 1. 2009 • Category: Asterisk
An Asterisk installation I am doing for a client has the promise of two interesting developments in the near future:

  1. Writing a CDR/Voicemail status-tracking application that makes sure that all client voicemails are handled within one hour and keeps track of what was done
  2. A future custom metrics/data warehouse implementation that will use both the CDR and VoiceMail data

With that in mind (and the fact that I'm using Mysql master-master replication across offices) I looked into storing the voicemails in the database as well. This is fairly well-documented. I had this up and running in no time, and now for each voicemail I have a neat row in my MySQL table.

The problem is - how do I connect the row in CDR table to the matching voicemail row? There are no columns that have data to allow you to do match one to the other. Even timestamps, if you were so desperate, are not exactly the same! So this is how I did it.

Continue reading "Data Warehousing and Metrics with Asterisk PBX"

Asterisk Teaching your Asterisk phone system to control your music

Posted by Admin • Saturday, January 17. 2009 • Category: Asterisk
Ever since I built my Asterisk-based VOIP phone system I've been finding more and more interesting ways to put it to work. I mean, it's a business phone server, but it's got plenty of resources as it sits around waiting for a phone call. So I figured... why not be able to pick up the nearest phone, dial an extension, and be able to stop/start/skip my whole-house music? How about doing this from anywhere? OK this may seem odd to you, but you'd be surprised how often it saves you from having to get up and go find the remote or having to wake up a computer :-). OK you're not convinced that this is very useful. How about pausing music for a phone call? Automatically?

For those who don't know, Asterisk is a open source (and free) software PBX system. You know, the thing that IP phones connect to (you know, that little box you got from Vonage - it connects to something - well in my house, it connects to my server). It provides call routing and management, voicemail, IVR (menus), etc. In short, it's awesome. I mean, yeah, it's a little unintuitive at first, but it's not that complicated once you loosen up your old fixed programmatic thinking a little :-)

So, here is the setup.

Continue reading "Teaching your Asterisk phone system to control your music"