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Asterisk Stability & Security

Protect your investment


Introduction
 What if the server goes down ?
 What if someone hacks into your 8 e1
asterisk server and makes calls to
inmarsat ?
 Inmarsat : 5 euro / min.
In 24 hours, on 8 e1s  1728000 euro
Overview

 Asterisk Performance Update


 Asterisk Stability
 Asterisk Security
 Asterisk Monitoring
Asterisk Performance Update
 Updates since Astricon 2004:
- Smaller memory footprint
- Less file descriptors used
- Memory leaks found / removed
- Less RTP ports opened
- Codec optimizations (especially Speex)
- Hardware echo canceller
- FastAGI
- Realtime
- Remote MOH
- ds3000 / te411p
- Channel walk optimization
Astertest Testlab
Astertest Cables
Overview

 Asterisk Performance Update


 Asterisk Stability
 Asterisk server monitoring
 Asterisk Security
Asterisk Stability

 Hardware reliability

 Software stability
Asterisk Stability – Hardware Reliability

 What is the cost of having no PBX service


for your company ?

 What if you are an ISP and your


customers can’t dial out ?
Asterisk Stability – Hardware Reliability

 What if you experience:

- power outage ?
- a broken HD ?
- a broken Zaptel card ?
- a broken server ?
- no Internet connectivity ?
Asterisk Stability – Hardware Reliability

 Power outage:
 Traditional phones are self powered.

Solution: use a UPS to power the (PoE) phones,


the switches, PBX, modem, router,…

 If you have a low power PBX, the phone


system could run for hours on a small UPS.
 Don’t use Ethernet over power for mission
critical phone lines.
Asterisk Stability – Hardware Reliability

 A broken HD ?

 Use raid > 0


 SCSI has a bigger mean time to failure.
 Flashdisks, realtime, netboot, live CD’s.
Asterisk Stability – Hardware Reliability

 A broken Zaptel card or a broken server ?

 Make sure you have a replacement,


(maybe even hot standby) with all the
modules you need, jumpers already set,…
Asterisk Stability – Hardware Reliability

 No Internet connectivity ?

 Spare router / modem / switch ?


 Failover Internet connection ?
 Failover to / from PSTN ?
Label all cables!!
Asterisk Stability / Quality Updates
Software related since Astricon ‘04

 Real CVS-stable / CVS-head (Thanks Russell!)


 Major cleanups / code audits.
 New h323 channel coming (chan_ooh323)
 Packet Loss Concealment
 IAX2 / SIP jitter buffer (mantis 3854)
 A lot of libpri, chan_sip, chan_h323 changes for
better compatibility / stability.
 DUNDi (easier load balancing with round robin
DNS)
 OSP
 Kernel 2.6.11.x
Changes in hardware reliability

 New Zaptel hardware (te411p, te4xxp,


TDM, IAXy2, …).
 New drivers with a lot of bug fixes and
optimizations.
 End of life for x100p and Tormenta cards.
 Hardware echo cancellers -> lower CPU
load -> more calls it can handle before
asterisk turns unstable.
* reliability / stability recommendations

 Use decent but not exotic hardware


 Put Zaptel on a different PCI-bus than Nics and
video cards.
 Read tutorials on interrupts, APIC and other
common problems.
 Load test your setup
 Design a failover system
 Noload unused modules
 Use recent firmware Zaptel cards
* reliability / stability recommendations

 Use a stable Asterisk version.


 Take a common OS -> Linux.
 Test software upgrades in a test lab.
 Stay away from experimental Asterisk
modules -> h323, skinny.
 Don’t patch production Asterisk servers.
 Keep your old Asterisk binaries after an
upgrade for easy restore of known working
versions.
Overview

 Asterisk Performance Update


 Asterisk Stability
 Asterisk server monitoring
 Asterisk Security
Asterisk server monitoring

 NAGIOS
  http://karlsbakk.net/asterisk/

http://megaglobal.net/docs/asterisk/html/asteri
 Argus: http://argus.tcp4me.com/
 SNMP: http://www.faino.it/en/asterisk.html
Overview

 Asterisk Performance Update


 Asterisk Stability
 Asterisk server monitoring
 Asterisk Security
Asterisk Security

 Asterisk Configuration stupidity


 Asterisk hardening
 Privacy protection
Asterisk Configuration Stupidity

 Dial plan security


 SIP.conf
 IAX2.conf
 Manager.conf
 Billing problems
Dial plan security

 - Extension hopping
 - CallerID based protections
 - _.
 - Demo context
 - User access to the dial plan
 - Be careful with the default context
 - Limit simultaneous calls
Extension hopping
 User can reach ANY extension in the current
context:

[internal]
exten => intro,1,Background(question);
exten => 1,spanish,Goto(Spanish)
exten => 2,english,Goto(English)

exten => _XX.,1,Dial(ZAP/g1/${EXTEN});


CallerID based protection
exten =>
_X.,1,GotoIf($[“${CALLERIDNUM}”=“32134”?3);
exten => _X.,2,Hangup();
exten => _X.,3,Dial(${EXTEN});

 When not explicitly defined for each


user/channel in zapata.conf, sip.conf, iax.conf,
the user can choose his own CallerID!
Inappropriate use of _.
 _. Would match EVERYTHING!
(also fax, hang up, invalid, timeout,….)

Example:
exten => _.,1,Playback(blah);
exten => _.,2,Hangup;
 Causing a FAST LOOP.

(changed in CVS-head)
demo context

 Not a real security risk


 But… Someone might play with your
system and use up your bandwidth, make
prank calls to Digium, make Mark Spencer
very unhappy and cause him to introduce
you to a very big shotgun…
User access to the dialplan

 - AMP and other GUI’s might allow the


ISP’s user to change a dial plan in his own
context. E.g.: hosted PBX’s

 - Goto / GotoIf / dial(Local/…) -> context


hopping.
 - System -> could do anything
Default context

 Example:
[default]
Include outgoing;
Include internal;

OH OH OH, guest calls will go to the default


context!!!!!
Context usage:

 A call has two legs, the used context is the


context defined for that user/channel in the
config file for that protocol.
E.g:
- Zap to sip call:
context set in zapata.conf is used
- SIP to IAX2 call:
context in sip.conf is used
Context usage:

 In sip.conf, zapata.conf, iax2.conf…

A default context is defined, if there is no


specific context setting for this channel or
user, than the default context is used!
Limit simultaneous calls
 Sometimes you don’t want a user to make multiple
simultaneous calls.

 E.g.: prepay / calling cards

Solution: setgroup, checkgroup (don’t trust incominglimit.)


exten => s,1,SetGroup(${CALLERIDNUM})
exten => s,2,CheckGroup(1)

Only good if the CallerID cannot be spoofed !!!!


Consider using accountcode for this.
Sip.conf
 Default context
 Bindport, bindhost, bindip
 [username] vs username=
 Permit, deny, mask
 Insecure=yes, very, no
 User vs peer vs friend
 Allowguest
 Autocreatepeer
 Pedantic
 Ospauth
 Realm
 Md5secret
 User authentication logic
 Username= vs [username]
Bindport, bindhost,bindip

 If you only use sip for internal calls, don’t


put bindip=0.0.0.0 but limit it to the internal
IP.

 Changing the bindport to a non 5060 port


might save you from portscan sweeps for
this port.
Permit, deny, mask

 Disallow everything, then allow per user


the allowed hosts or ranges.
(Multiple are allowed.)
SIP.conf – insecure option
Insecure = …

 No: the default, always ask for authentication


 Yes: To match a peer based by IP address only
and not peer.
 Insecure=very ; allows registered hosts to call
without re-authenticating, by ip address
 Insecure=port; we don’t care if the portnumber is
different than when they registered
 Insecure=invite; every invite is accepted.
User vs Peer vs Friend in SIP
 USER: never registers only makes calls
 PEER: can register + can make calls.
[user1]
type=user
[user1]
type=peer

Is allowed and the same as type=friend if the other


parameters are identical!!!
Allowguest =…

 True: unauthenticated users will arrive in


the default context as defined in sip.conf
 False: unauthenticated users will get a
permission denied error message.
 OSP: to allow guest access for voip traffic
coming from an OSP server.
autocreatepeer

 The autocreatepeer option allows, if set to Yes,


any SIP UA to register with your Asterisk PBX as
a peer. This peer's settings will be based on
global options. The peer's name will be based
on the user part of the Contact: header field's
URL.

This is of course a very high security risk if you


haven't got control of access to your server.
 © Olle
Pedantic

 Defaults to pedantic=no
 If enabled, this might allow a denial of
service by sending a lot of invites, causing
a lot of (slow) DNS lookups.
Realm

 Realm=Asterisk; Realm for digest


authentication
; Defaults to “Asterisk"
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain name
How is authentication done?

 chan_sip.c: /* Whoever came up with the


authentication section of SIP can suck my
%*!#$ for not putting an example in the
spec of just what it is you're doing a hash
on. */
How is authentication done?

 Look at FROM header in SIP message for the username:

-> browse sip.conf for a type=user with that username


If found -> check the md5
If not found,

-> browse sip.conf for a type=peer with that username


-> browse sip.conf for an (registered) IP where the request is coming from
if insecure=very, no more checks are done
if insecure=port, if they are willing to authenticate, even if they are calling
from a different port than they registered with. (used for NAT not using the
same port number every time).
otherwise, check the md5 + allow/deny.

 If no peer found ? do we allow guest access (allowguest=true ?)


 Yes? OK, allow send it to the default context, if not reject.
Secret vs md5secret

 With SIP all passwords are md5 encrypted


when sending the packets, but are stored
in plaintext in sip.conf

 [user]
 Secret=blabla
Secret vs md5secret

 echo - n "<user>:<realm>:<secret>" | md5sum


 E.g.:
echo -n "user:asterisk:blabla" | md5sum
e1b588233e4bc8645cc0da24d8cb848d

[user]
md5secret=e1b588233e4bc8645cc0da24d8cb848d
Username= vs [username]

 [username] is for authentication a client


connecting to asterisk.

Username=… is to have your asterisk


server authenticate to another SIP server.
Iax.conf

 auth=plaintext,md5,rsa
 User authentication logic
 Default context
 [username] vs username=
 Permit, deny, mask
 Bindport, bindhost, bindip
 User vs peer vs friend
iax.conf - auth

 Plaintext: passes are sent in plaintext


 Md5: encrypt the password with md5
 RSA: use public key / private key – uses
AES.
User vs Peer vs friend
 USER: can only accept calls
 PEER: can only make calls
 FRIEND: can do both
[user1]
type=user
[user1]
type=peer

Is allowed!!!
How is authentication done?
 In iax2: (cvs-head!!)

Pseudocode:
Is username supplied ?
-> yes -> matched against iax.conf users starting bottom to top.
user found ?
-> yes : is IP in allowed / disallowed list ?
yes –> does password match ?
yes -> does requested context match a context=… line?

-> no -> is a password given ?


-> yes : Asterisk will look bottom to top for a user with this password,
-> if the context matches, or there is no context specified, and the
host is in the allowed lists (allow / deny) then the call is accepted.

-> no: Asterisk will look bottom to top for a user without password.
-> if the context matches, or there is no context specified, and the
host is in the allowed lists (allow / deny) then the call is accepted.
USER CONTEXT CALL
USERNAME ? YES YES IP ALLOWED? YES PW MATCH? YES YES
FOUND? OK? ACCEPTED!

NO NO NO NO

NO

CALL REFUSED!

PASS USER FOUND WITH THIS CONTEXT CALL


YES YES IP ALLOWED? YES YES
GIVEN ? PASSWORD ? OK? ACCEPTED!

NO NO NO
NO

CALL REFUSED!

USER FOUND WITH CONTEXT CALL


YES IP ALLOWED? YES YES
EMPTY PASSWORD ? OK? ACCEPTED!

NO NO
NO

CALL REFUSED!
 Add a last entry in iax.conf with no
password to force nosecret access into a
specific context.
 If you use realtime, don’t have any user
without a password and without
permit/deny.
Manager.conf
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[zoa]
secret = blabla
deny=0.0.0.0/0.0.0.
permit=221.17.246.77/255.255.255.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Manager.conf

 No encryption is used, even the password


is sent in plaintext.

 Don’t enable it on a public IP.


 Use http://www.stunnel.org/
 Watch out with management programs
with direct interface to the manager.
 Limit the privileges per user (especially the
system!!!).
Asterisk Security

 Asterisk Configuration stupidity


 Asterisk hardening
 Privacy protection
Asterisk Hardening
 Asterisk as non-root user
 Asterisk in CHROOT
 Asterisk in a JAIL
 Asterisk with limited read / write permissions
 ZAPTEL kernel modules
 Asterisk firewalling / shaping / NAT
 Tty9
 Linux hardening
 Remote logging
 Tripwire
 Limit running system processes
Asterisk as non root user
adduser --system --home /var/lib/asterisk --no-create-home Asterisk
chown -r asterisk:asterisk /var/lib/asterisk
chown -r asterisk:asterisk /var/log/asterisk
chown -r asterisk:asterisk /var/run/asterisk
chown -r asterisk:asterisk /var/spool/asterisk
chown -r asterisk:asterisk /dev/zap
chown -r root:asterisk /etc/asterisk

chmod -r u=rwX,g=rX,o= /var/lib/asterisk


chmod -r u=rwX,g=rX,o= /var/log/asterisk
chmod -r u=rwX,g=rX,o= /var/run/asterisk
chmod -r u=rwX,g=rX,o= /var/spool/asterisk
chmod -r u=rwX,g=rX,o= /dev/zap
chmod -r u=rwX,g=rX,o= /etc/asterisk
chown asterisk /dev/tty9
su asterisk -c /usr/sbin/safe_asterisk
or
Asterisk -U asterisk -G asterisk
Asterisk with limited read / write permissions

 Asterisk has no write permissions for its


config files and is running as non root ?

 In the unlikely event of someone breaking


in through Asterisk, your dial plan is still
vulnerable through the CLI or the
manager.
Asterisk in chroot

 Changes the root directory visible to


asterisk to e.g. /foo/bar
 Pretty useless if asterisk is running as root
and perl or gcc is available.
Asterisk in a jail

 Changes the root


directory visible to
Asterisk.
 Limits the commands
/ programs any user
in this jail can execute
to a list you specify.
 Expansion of chroot.
Zaptel kernel modules
 Zaptel is module only, cannot be put into the
kernel.

 Hackers like to hide in a module, they can


backdoor a module, compile it, load it in memory
and remove all traces on the disk.

 You could have the kernel check an md5 for the


Zaptel modules.
 I think Matt Frederickson compiled them in the
kernel before.
Firewalling / shaping / NAT

 Block everything except the ports you


really want. (5060, 4569, …)

 RTP ports are a big pita (see rtp.conf)

Sidenote: you might want to check your ISP


is not blocking anything in the range
defined in RTP.conf
Limit access to tty9

 safe_asterisk opens a console on tty9.

This does not require a password and will


provide a root shell to anyone passing by.
(by using !command on the CLI).
 Remove the offending line, or don’t use
safe_asterisk
Linux Hardening

 GRsec (2.6.x)
 Openwall (2.4.x)
 Remove all unneeded things.
Remote logging

 Remote syslog
 Put Asterisk log files (and other log files on
a remote server).
Tripwire

 Make hashes of all the important files on


the server and check them for changes
you didn’t do.
Limit server processes

 An Asterisk server should be only:

- OS + ASTERISK.
- No database
- No APACHE
- No PHP
(If you really need those, and don’t have enough
servers, don’t put them on a public IP and
firewall them!!!!)
Asterisk Security

 Asterisk Configuration stupidity


 Asterisk hardening
 Privacy protection
Asterisk privacy

 Encryption
 Monitoring
 CallerID spoofing
 CallingPRES
Call Encryption - SIP

 SRTP -> method to encrypt voice packets.


 TLS -> method to encrypt signaling
packets.

Both are not yet supported by asterisk.


Bounty on voip-info.org.
Call Encryption – IAX2

 30/12/2004 2:07
Modified Files: chan_iax2.c iax2-parser.c
iax2-parser.h iax2.h Log Message: Minor
IAX2 fixes, add incomplete-but-very-
basically-functional IAX2 encryption.

It would support any type of encryption you


like. -> Doesn’t work yet.
Call Encryption – General solution

 Send you packets through a VPN or


tunnel.

 Use only UDP tunnels to avoid delays.


Known to work:

IPSEC, VTUN, OPENVPN.


Call Encryption – Tunnel solution

Advantage, CPU expensive encryption


can happen on dedicated machine.
 Disadvantage: doesn’t work on
hardphones or ATA’s without adding an
extra server in front of them.
Monitoring

 ZapBarge
 ChanSpy
 Monitor