Over the last few days we have been discussing a vulnerability in Asterisk dialplans on the Asterisk Developer List. Olle has now posted a writeup of the problem and solution on his blog:

Friends,

Last week, Hans Petter Selansky alerted us of a potential security issue in all releases of Asterisk. In fact, it doesn't involve the code, but the most common way to construct dialplans. If you have something like this in your Asterisk, you need to update your dialplans:

[incoming-from-voip]
exten => _X., 1, dial(SIP/${EXTEN})

Many VoIP protocols support a large character set, that may cause harm in your dialplan
====================================================================

I've written an article about this on my blog, where my summary says:

"Because of a conflict between allowed characters in the called number or name in many VoIP protocols and the way Asterisk handles channel variables, there is a security risk hidden in many dialplans based on examples provided over the years by the Asterisk developers, trainers and community. The primary risk is that by using an ampersand in the dialstring, a user can access protected resources or misuse the pbx services. However, this character is not the only problem, as other characters may cause unexpected or problematic behavior."

There will be an Asterisk Security Advisory document coming out from Digium soon, as well as updated documentation and examples within the Asterisk source code tree. I strongly advise everyone to follow these and stay updated. (I have no access to the ASA system myself and can't generate an official security alert).

For more information about this issue and some code examples of what I personally currently think are good ways to prevent misuse of your services, please read my blog article at

http://www.voip-forum.com/?p=241&preview=true

Please help us to distribute this message!
=================================
We need help from all involved in the Asterisk eco-system. This is not something that the development team can solve by itself. We can add documents, READMEs and fix our own examples. But that won’t fix it. We need everyone involved to pump this information out in all the veins that runs through the Asterisk eco-system. In all languages needed, we shall say: "Audit your dialplans, fix this issue. And do it now."

Everyone that runs a web site with dialplan examples - audit your examples, fix them. Everyone that has published books on Asterisk - publish errata on your web site. Please help us - and do it now.

If you add web links, please add links both to http://www.asterisk.org where the official documents will soon be published, as well as to my blog (if you like, of course). But don't just refer to my blog entry alone.

I have updated my own servers and will now start auditing my customers' servers. After that I will have to update all my training materials so I don't repeat the bad examples. There's no magic bullet, no wonderful code patch, that can fix this, just hard work with all dialplans that accept calls over VoIP channels.

Let us all work together to fix this!

With Asterisk greetings!

/Olle

PS. If someone can update the entries on Queue() and Dial() in voip-info.org, that would be considered a good thing (TM).

Comments (Click to post)

Name:
Subject:
Website:
Message: