Xorcom Astribank

Astribank is a versatile and powerful channel bank that was specifically designed for the Asterisk IP-PBX. Astribank supports all the common telephony lines and trunks: FXS, FXO, BRI, E1/T1 PRI, T1 CAS and E1 R2. The Astribank driver is a part of the standard Asterisk distribution.


Knowledge Base

Sip debugging with wireshark

Wireshark and Cloudshark are invaluable tools for debugging sip and iax issues on your Asterisk server.

Here we have a short Video that goes over the basics of getting a call captured and opened in Cloudshark

we also have a short tutorial for download here in PDF format

First we need to get the packets we want. This is far simpler than its thought. We use a simple command line tool called tcpdump, if its not installed install it now, You wont be able to live without it.

Here we have 2 commands, The first captures packets on interface eth0, -n means we won’t convert addresses, -w means we just capture raw packets and udp means its only the udp packets we want and finally port 5060 means its only the sip messaging we want. In the second we dont specify port 5060 so that we get the rtp stream as well.

/usr/sbin/tcpdump -n -i eth0 -w /tmp/wireshark.pcap -s2000 udp port 5060
 /usr/sbin/tcpdump -n -i eth0 -w /tmp/wireshark.pcap -s2000 udp
screen -S "udpDump" -dm tcpdump -n -i eth0 -C 9 -W 15 -w /var/log/asterisk/dumpsip.pcap -s2000 udp port 5060

The command above will write to file in the background and will rotate at 9 meg so suitable for cloudshark

Once you have started the capture and made a call as required you will get a file called for example /tmp/wireshark.pcap copy this to your workstation via ftp or sftp as you would copy any file.

Knowledge Base

Better SIP security

In Seven Steps

Original Text by J Todd March 28th, 2009

In case any of you were wondering why there has been a fairly notable upswing in the attacks happening on SIP endpoints, the answer is “script kiddies.”  In the last few months, a number of new tools have made it easy for knuckle-draggers to attack and defraud SIP endpoints, Asterisk-based systems included.  There are easily-available tools that scan networks looking for SIP hosts, and then scan hosts looking for valid extensions, and then scan valid extensions looking for passwords.You can take steps, NOW, to eliminate many of these problems.  I think the community is interested in coming up with an integrated Asterisk-based solution that is much wider in scope for dynamic protection (community-shared blacklists is the current thinking) but that doesn’t mean you should wait for some new tool to defend your systems.  You can IMMEDIATELY take fairly common-sense measures to protect your Asterisk server from the bulk of the scans and attacks that are on the increase. The methods and tools for protection already exists – just apply them, and you’ll be able to sleep more soundly at night.

Seven Easy Steps to Better SIP Security on Asterisk:


1) Don’t accept SIP authentication requests from all IP addresses.  Use the “permit=” and “deny=” lines in sip.conf to only allow a reasonable subset of IP addresess to reach each listed extension/user in your sip.conf file.  Even if you accept inbound calls from “anywhere” (via [default]) don’t let those users reach authenticated elements!


2) Set “alwaysauthreject=yes” in your sip.conf file.  This option has been around for a while (since 1.2?) but the default is “no”, which allows extension information leakage.  Setting this to “yes” will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks.


3) Use STRONG passwords for SIP entities.  This is probably the most important step you can take.  Don’t just concatenate two words together and suffix it with “1? – if you’ve seen how sophisticated the tools are that guess passwords, you’d understand that trivial obfuscation like that is a minor hinderance to a modern CPU.  Use symbols, numbers, and a mix of upper and lowercase letters at least 12 digits long.


4) Block your AMI manager ports.  Use “permit=” and “deny=” lines in manager.conf to reduce inbound connections to known hosts only.  Use strong passwords here, again at least 12 characters with a complex mix of symbols, numbers, and letters.


5) Allow only one or two calls at a time per SIP entity, where possible.  At the worst, limiting your exposure to toll fraud is a wise thing to do.  This also limits your exposure when legitimate password holders on your system lose control of their passphrase – writing it on the bottom of the SIP phone, for instance, which I’ve seen.


6) Make your SIP usernames different than your extensions.  While it is convenient to have extension “1234? map to SIP entry “1234? which is also SIP user “1234?, this is an easy target for attackers to guess SIP authentication names.  Use the MAC address of the device, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, try “md5 -s ThePassword5000?)


7) Ensure your [default] context is secure.  Don’t allow unauthenticated callers to reach any contexts that allow toll calls.  Permit only a limited number of active calls through your default context (use the “GROUP” function as a counter.)  Prohibit unauthenticated calls entirely (if you don’t want them) by setting “allowguest=no” in the [general] part of sip.conf.


These 7 basics will protect most people, but there are certainly other steps you can take that are more complex and reactive.  Here is a fail2ban recipe which might allow you to ban endpoints based on volume of requests.  There is discussion on the asterisk-user and asterisk-dev mailing lists of incorporating this type of functionality into Asterisk – let’s hear your ideas!


If you’d like to see an example of the tools that you’re up against, see this demo video of an automated attack tool that does scan, guess, and crack methods via a click-and-drool interface.
In summary: basic security measures will protect you against the vast majority of SIP-based brute-force attacks.  Most of the SIP attackers are fools with tools – they are opportunists who see an easy way to defraud people who have not considered the costs of insecure methods.  Asterisk has some methods to prevent the most obvious attacks from succeeding at the network level, but the most effective method of protection are the administrative issues of password robustness and username obscurity.


Knowledge Base Technical

Nagios plugin for reading the Asterisk Database

This is a simple plugin that is based on one by Jason Rivers We have changed it now to read the ASTDB (Asterisk internal Database and then based on ok and Critical keys it will report OK or Critical staus reports to Nagios.

This was written for reporting if an Elastix system is in Day or Night mode.

You can define the Database Family, Key, Critical value and OK value. This means you can cutomise it to what ever you need to report.


The Code is below, make you may need to change /usr/bin/nc for what ever you use for netcat.

any issues email us, but dont forget this is given for free not supported for free.

# Program : check_asterisk_ami
# :
# Author : Original code by Jason Rivers < >
# : Modified by for checking the asterisk Database
# :
# Purpose : Nagios plugin to return Information from an Asterisk host using AMI
# :
# Parameters : --help
# : --version
# :
# Returns : Standard Nagios status_* codes as defined in
# :
# Licence : GPL
# Notes : See --help for details
PROGNAME=`basename $0`
PROGPATH=`echo $0 | /bin/sed -e 's,[\/][^\/][^\/]*$,,'`
REVISION=`echo '$Revision: $' | sed -e 's/[^0-9.]//g'`
print_usage() {
echo "Usage: $PROGNAME [-H hostname] [-u username] [-p password] [-P port] [-k DBkey] [-c critical] [-o ok] [-f family]"
echo " -H Hostname"
echo " -u AMI Username"
echo " -p AMI Password"
echo " -P (optional) AMI PORT"
echo " -k Database key"
echo " -f Database family"
echo " -c Critical Key"
echo " -o OK KEY"
echo ""
echo "SupportedCommands:"
echo " Most DB familiys that toggle such as DayNight in elastix"
echo "Usage: $PROGNAME --help"
echo "Usage: $PROGNAME --version"
print_help() {
print_revision $PROGNAME $REVISION
echo ""
echo "Nagios Plugin to check Asterisk ASTDB using AMI"
echo ""
echo ""
echo "Asterisk Call Status Check. orignal version by © Jason Rivers 2011 changes to do ASTDB by"
echo ""
exit 0
# support
# If we have arguments, process them.
exitstatus=$STATE_WARNING #default
while test -n "$1"; do
case "$1" in
exit $STATE_OK
exit $STATE_OK
print_revision $PROGNAME $REVISION
exit $STATE_OK
print_revision $PROGNAME $REVISION
exit $STATE_OK
-u) AMIUSER=$2;
-p) AMIPASS=$2;
echo "Unknown argument: $1"
if [ "${AMIPORT}" = "" ]; then
if [ "${FAMIL}" = "" ]; then
echo="CRITICAL: Unknown KEY"
## Checking Astdb
CHANNELS=`/bin/echo -e "Action: login Username: ${AMIUSER} Secret: ${AMIPASS} Events: off Action: DBGet Family: ${FAMIL} Key: ${DBKEY} Action: Logoff " | /usr/bin/nc $REMOTEHOST ${AMIPORT} | awk '/Val/ {print $2}'|tr -d " "`
if [ "$CHANNELS" = "" ]; then
echo "UNKNOWN: Unable to get ASTDB status"
if [ "$CHANNELS" = "${OKNAME}" ]; then
MSG="OK: ${DBKEY} Asterisk Emergency message not active"
elif [ "$CHANNELS" = "" ]; then
MSG="WARNING: Asterisk Unknown status"
elif [ "$CHANNELS" = "$CRITICALNAME" ]; then
MSG="CRITICAL: ${DBKEY} Asterisk Emergency message active"
echo $MSG
exit $exitstatus

Design Installation Services Support Technical

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Knowledge Base

General Configuration Guide Skype for SIP and Asterisk


If you are new to SIP, Asterisk is a useful, open-source (GPL) platform with which to test and experiment with the Skype for SIP. This is a guide on how to install Skype for SIP on a system agnostic or “vanilla” Asterisk server.


To install Asterisk on your server, please see the Digium documentation here


This configuration guide is based on Debian Linux (Lenny 64bit). With a basic installation of Debian you can install Asterisk by issuing the following APT command at the command line:-

apt-get install asterisk



Configuration Files for Vanilla Asterisk


In configuring Skype for SIP on a vanilla Asterisk system we are primarily concerned with two configuration files:-


  1. sip.conf (located in the /etc/asterisk/ directory)
    The sip.conf file holds the registration details for the Skype for SIP channel
  2. extensions.conf (located in the /etc/asterisk/ directory)The extensions.conf holds the dial plan telling Asterisk what to do with incoming and outgoing calls.-


Let’s do a walkthrough of the configuration steps.


Configuring the sip.conf File


Step 1


The sip.conf file has two sections that need to be completed. The “General” section (denoted in the file with the [general] heading) and peer section denoted in the file with the [peers] heading.


In the General section we need to add a “register” line. This tells Asterisk to register with Skype at the Skype local point of presence.


Add the following, under the “[general]” section in the file, substituting your 9905xxxx number and password with your actual credentials for the Skype for SIP profile you wish to use. Your SIP Profile details can be found in the Skype Business Control Panel (BCP):-


register => 99051000xxxxxx: /99051000xxxxxx


Step 2

To ensure that we also receive the callerID from Skype clients we also should add:-


trustrpid = no

sendrpid = yes



Step 3

Next, we add a section for the peer, in the “[peers]” section of the sip.conf file. Again we substitute the 9905xxxxx number and password with the SIP Profile credentials from the Skype Business Control Panel (BCP):-



type = peer

username = 99051000xxxxxx

fromdomain =

fromuser = 99051000xxxxxx

realm =

host =

dtmfmode = rfc2833

secret = PaSsW0rD

nat = no ;This should be set to reflect your network NAT configuration

canreinvite = no

insecure = invite

qualify = yes

disallow = all

allow = alaw

allow = ulaw

;allow = g729 ; Uncomment this if you have G729 licences

amaflags = default

trustrpid = no

sendrpid = yes

context = skype_in


Please Note:

If your Asterisk PBX is behind a NAT device, you should set “nat = yes” in this section.


If your Asterisk PBX has a dedicated internet IP address, set this to “nat = no”.


Step 4

After setting these changes, reload the Asterisk’s SIP module by typing:-


asterisk -rx “reload”


…….at the command line.


Step 5

After the SIP Module has reloaded enter asterisk -rx “sip show peers” at the command line, which should return:


pbx*CLI> sip show peers

Name/username Host Dyn Nat ACL Port Status

99051000xxxxxx/99051000xx 5060 OK (52 ms)


Then enter asterisk -rx sip show registry” which should return:


pbx*CLI> sip show registry

Host Username Refresh State Reg.Time 99051000xxxx 105 Registered day, dd mmm yyyy hh:mm:ss


If you see output similar to the above, then you are registered to the Skype SIP gateway and ready to make and receive calls.


We now need to setup the extensions.conf so that we have a dialplan setup and Asterisk knows how to deal with incoming and outgoing calls.


Configuring the extensions.conf File


The extensions.conf file requires a “context” and an “extension” to be added for incoming Skype calls, plus an extension to be added to the context that users use for outgoing calls.


Incoming “context”


Add the following lines to the [context] section of extensions.conf, substituting 9905xxxxxxx with the 9905 number for the SIP Profile. Again you can find the details of your Skype SIP Profiles in the Skype BCP:-



exten => 99051xxxxxxxx,Noop(${CALLERID(name)} , ${CALLERID(num)})

exten => 99051xxxxxxxx,n,Dial(SIP/100,30,t,r)

exten => 99051xxxxxxxx,n,voicemail(100|u)


This is a simple “vanilla” context that shows us the callerID name and number, dials extension 100 for 30 seconds and finally, if unanswered, goes to voicemail. This sequence will need to be amended to suit your requirements. If you are planning on having many SIP Profiles or Online Numbers that all need to end up at the same destination, or the destination is decided by the Skype Business Account that the online number is registered against, a more complicated Dialplan can be used. For example:-



exten => 99051xxxxxxxx,1,Noop(${CALLERID(name)} , ${CALLERID(num)})

exten => 99051xxxxxxxx,n,Queue(sfs|r|||40)

exten => 99051xxxxxxxx,n,voicemail(100|u)



Outgoing “Context”


The outgoing context must be included in the context for your user’s phones. Usual security measures apply. Do not include this in a context for incoming calls.




exten => _90Z.,1,Set(CALLERID(num)= 99051xxxxxxxx)

exten => _90Z.,n,Dial(SIP/0044${EXTEN:2}@99051xxxxxxxx)


exten => _900.,1,Set(CALLERID(num)= 99051xxxxxxxx)

exten => _900.,n,Dial(SIP/${EXTEN:1}@99051xxxxxxxx)



In the sip.conf add the following to create user 100





callerid=”myskypetrunk” <100>

















in the extensiosn.conf add the following to the default context


exten => _XXX,1,Dial(SIP/${EXTEN},20)


Also create a context called international



include => default

include => skype_out