mpg123 -w
; convert the output wav file to format that Asterisk can recognize
sox -v 0.9 input.wav -c 1 -r 8000 output.wav
; -v: reduce volume
Showing posts with label asterisk. Show all posts
Showing posts with label asterisk. Show all posts
Friday, 20 December 2013
convert mp3 to gsm/ulaw for Asterisk
# Install mpg123 to convert mp3 to wav file
mpg123 -w
; convert the output wav file to format that Asterisk can recognize
sox -v 0.9 input.wav -c 1 -r 8000 output.wav
; -v: reduce volume
mpg123 -w
; convert the output wav file to format that Asterisk can recognize
sox -v 0.9 input.wav -c 1 -r 8000 output.wav
; -v: reduce volume
Monday, 16 December 2013
Install chan_mobile on ClearOS 5.2 SP2, Asterisk 1.8 and FreePBX 2.8
Source: http://samyantoun.50webs.com/asterisk/chan_mobile/
Install chan_mobile on ClearOS 5.2 SP2, Asterisk 1.8 and FreePBX 2.8
Install chan_mobile on ClearOS 5.2 SP2, Asterisk 1.8 and FreePBX 2.8
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Use an old Mobile Phone as a GSM Gateway in Asterisk
SOURCE: http://www.stocksy.co.uk/articles/Networks/use_an_old_mobile_phone_as_a_gsm_gateway_in_asterisk/
Like most people I carry a mobile phone, but mine is for emergencies only. Just a handful of people know the the number and that's how I like to keep it. Because I use Asterisk, I'm always reachable through my land line number which I route around between various destinations or voicemail depending on what suits me. I'm spoilt by this, so having my mobile phone ring unexpectedly at an inconvenient time is a bit intrusive.
But, increasingly, almost everyone I interact with wants my mobile number - employers, customers, banks, garages, insurance companies - if it doesn't start '07', they're not happy.
As usual I've decided to try a technical solution to a social problem. I began by using a 'personal use' 070 number which is designed for precisely the kind of single-number-reach setup I use. This 070 number was presented at my SIP provider, who would then route the calls to my Asterisk server across the internet. In the end, this proved to be unsatisfactory because many providers block the 070 range with the justification that it has been abused for premium-rate scams. For example, the number couldn't be dialled from T-Mobile or Orange. Shame.
Undeterred, I tried another approach. I now have an old spare mobile phone which never leaves the house and is permanently connected through bluetooth to Asterisk. This is a real mobile with a real mobile number. I simply feed any incoming calls to this mobile into a macro which handles the call in the same way as calls to my landline. If I want to take calls on my (real) mobile, I can. If I'm not available to take the call, the caller is passed to my Asterisk voicemail box. SMS text messages arrive as emails, my replies to which are sent by SMS. Having all my incoming calls and voicemail messages in one place is very convenient and it prevents me from missing calls when I am in the house and probably would not hear a mobile phone ringing.
Asterisk has included support for bluetooth connections to mobile phones and headsets for some time now. This is accomplished through chan_mobile. Not all phones are supported, so it's worth taking a look at voip-info.org's page which lists the confirmed compatible dongles and phones. I am getting good results with a D-Link DBT-120 dongle and a Nokia E72, 6306i, 6021 handsets, however only the 6021 works with SMS. It is worth noting that each bluetooth dongle can support only one mobile device - this is an annoying limitation of chan_mobile, but it's not as though USB dongles are very expensive.
Once the phone is paired in the normal way, it will send any incoming SMS messages to Asterisk over the bluetooth connection. Asterisk looks for an 'sms' extension in the context you specified in chan_mobile.conf. I suggest something like this in your dialplan:
Outgoing SMS messages are more work because it's necessary to parse the contents of the email message, the format of which will be a little less predicatable than an SMS. I elected to use python to do this because it already has a library to do this.
So, calling email2sms.py with the argument 'stocksy' uses the JS6021 mobile, and calling it with 'karen' uses the trollphone mobile.
You need to make sure that email for the domain you have chosen - in my case sms.stocksy.co.uk - is routed to the Asterisk box. This will normally be accomplished by creating an MX record or creating a transport for the domain on your mail server. Again, I'm not going to cover that part here, but I will cover how to pipe the incoming messages into the python script.
Assuming that you are using postfix, you'll need a new transport for each mobile you want to use. In my case:
If things aren't quite working, start by checking your mail log:
Good luck.
Like most people I carry a mobile phone, but mine is for emergencies only. Just a handful of people know the the number and that's how I like to keep it. Because I use Asterisk, I'm always reachable through my land line number which I route around between various destinations or voicemail depending on what suits me. I'm spoilt by this, so having my mobile phone ring unexpectedly at an inconvenient time is a bit intrusive.
But, increasingly, almost everyone I interact with wants my mobile number - employers, customers, banks, garages, insurance companies - if it doesn't start '07', they're not happy.
As usual I've decided to try a technical solution to a social problem. I began by using a 'personal use' 070 number which is designed for precisely the kind of single-number-reach setup I use. This 070 number was presented at my SIP provider, who would then route the calls to my Asterisk server across the internet. In the end, this proved to be unsatisfactory because many providers block the 070 range with the justification that it has been abused for premium-rate scams. For example, the number couldn't be dialled from T-Mobile or Orange. Shame.
Undeterred, I tried another approach. I now have an old spare mobile phone which never leaves the house and is permanently connected through bluetooth to Asterisk. This is a real mobile with a real mobile number. I simply feed any incoming calls to this mobile into a macro which handles the call in the same way as calls to my landline. If I want to take calls on my (real) mobile, I can. If I'm not available to take the call, the caller is passed to my Asterisk voicemail box. SMS text messages arrive as emails, my replies to which are sent by SMS. Having all my incoming calls and voicemail messages in one place is very convenient and it prevents me from missing calls when I am in the house and probably would not hear a mobile phone ringing.
Asterisk has included support for bluetooth connections to mobile phones and headsets for some time now. This is accomplished through chan_mobile. Not all phones are supported, so it's worth taking a look at voip-info.org's page which lists the confirmed compatible dongles and phones. I am getting good results with a D-Link DBT-120 dongle and a Nokia E72, 6306i, 6021 handsets, however only the 6021 works with SMS. It is worth noting that each bluetooth dongle can support only one mobile device - this is an annoying limitation of chan_mobile, but it's not as though USB dongles are very expensive.
How it's Done
chan_mobile is an addon, so it needs to be enabled before Asterisk is compiled. On Debian, it's pretty simple, just add a few packages:# apt-get install bluez-utils bluez-hcidump libbluetooth-devthen, go to your Asterisk source directory and use make menuselect to enable chan_mobile. It's in Add-ons -> chan_mobile:
# cd /usr/src/asterisk-1.8.11.0 # ./configure && make menuselectWhilst it compiled and installed OK, I had to make a modification to the chan_mobile source before it would recognise my phone:
# vi /usr/src/asterisk-1.8.11.0/addons/chan_mobile.c Find this: addr.rc_channel = (uint8_t) 1; Replace with: addr.rc_channel = (uint8_t) 0;Build Asterisk and (re)install:
# make && make installIn order to use a bluetooth-connected phone as a GSM gateway, it's necessary to pair the phone with the Asterisk server. In Debian, this can be accomplished painlessly through the CLI. First, make your phone discoverable and then scan for it:
# hcitool scan Scanning ... EC:1B:6B:64:C2:88 TrollphoneMake a note of the MAC address. In order to pair, a helper is required to handle the PIN. Run the helper in the background and begin the pairing process:
# bluetooth-agent 7472 & # rfcomm connect hci0 EC:1B:6B:64:C2:88Once the pairing has succeeded, make sure your phone is configured to automatically accept connections for this paring in future. You can verify that the paring is working at any time by running:
# hcitool con Connections: < ACL EC:1B:6B:64:C2:88 handle 41 state 1 lm MASTER AUTH ENCRYPTNow, Asterisk needs to be configured to use the paired phone. We need to know which rfcomm channel offers the voice service. The easiest way is to use chan_mobile:
# rasterisk *CLI> module load chan_mobile.soDon't worry about any errors loading the module, it'll do for now:
*CLI> mobile search EC:1B:6B:64:C2:88 Trollphone Yes Phone 2In this case it is rfcomm channel 2. In addition, we need to know the MAC address of the bluetooth dongle installed in the Asterisk server. Exit the Asterisk CLI and use hcitool:
# hcitool dev Devices: hci0 00:81:C5:33:25:A4At last we have all the information needed. Edit or create the chan_mobile configuration file:
# vi /etc/asterisk/chan_mobile.conf [Adapter] address = 00:81:C5:33:25:A4 id = pabx [Trollphone] address = EC:1B:6B:64:C2:88 port = 2 context = from-trollphone adapter = pabxYou will need something in the dialplan to handle this, at minimum something like:
# vi /etc/asterisk/extensions.conf [from-trollphone] exten => s,1,Dial(SIP/100) [my-phones] exten => *12,1,Dial(MOBILE/Trollphone/150)When the mobile rings, you should get a call on SIP extension 100. Dialling *12 will cause the phone to dial 150, which in my case gives me Orange customer services. I'm sure you get the idea.
What about SMS?
Trickier, but there is a solution. None of the phones I had spare were supported by chan_mobile's SMS capabilities. According to the chan_mobile wiki page, only three phones are known to support SMS: the Nokia models E51, 6021 and 6230i. Of the three, the 6021 seems to be the most widely available - I was able to get three of them from eBay for just a few pounds.Once the phone is paired in the normal way, it will send any incoming SMS messages to Asterisk over the bluetooth connection. Asterisk looks for an 'sms' extension in the context you specified in chan_mobile.conf. I suggest something like this in your dialplan:
[from-trollphone]
exten => sms,1,Verbose(Incoming SMS from ${SMSSRC} ${SMSTXT})
exten => sms,n,System(echo "To: stocksy@stocksy.co.uk" > /tmp/smsmail)
exten => sms,n,System(echo "Subject: SMS from ${SMSSRC}" >> /tmp/smsmail)
exten => sms,n,System(echo "${SMSTXT}" >> /tmp/smsmail)
exten => sms,n,System(sendmail -t -f ${SMSSRC}@sms.stocksy.co.uk < /tmp/smsmail)
exten => sms,n,Hangup()
At first, incoming messages were all arriving with a blank ${SMSSRC}, the easy solution was to
apply a patch and re-compile:
# cd /usr/src/asterisk-1.8* # wget --no-check-certificate https://issues.asterisk.org/jira/secure/attachment/42026/sms-sender-fix.diff # patch -p0 < sms-sender-fix.diff # ./configure && make && make installNow, incoming messages are delivered to me as emails claiming to be from +MOBILENUMBER@sms.stocksy.co.uk. Obviously, this requires the Asterisk system to have a working MTA, the setup of which I won't cover here. If you don't have an MTA at present, take a look at postfix.
Outgoing SMS messages are more work because it's necessary to parse the contents of the email message, the format of which will be a little less predicatable than an SMS. I elected to use python to do this because it already has a library to do this.
#!/usr/bin/env python
# (:? YOUR SCRIPT IS BAD AND YOU SHOULD FEEL BAD! (:?
# I'M NOT A DEVELOPER AND THIS IS PROBABLY VERY, VERY BAD, but it does work.
# email2sms.py James Stocks
# based upon emailspeak.py by sysadminman - http://sysadminman.net
# v0.0 2012-04-28
# Import libs we need
import sys, time, email, email.Message, email.Errors, email.Utils, smtplib, os, socket, random, re
from datetime import date
from email.Iterators import typed_subpart_iterator
from time import sleep
# Asterisk Manager connection details
HOST = '127.0.0.1'
PORT = 5038
# Asterisk Manager username and password
USER = 'your-ast-man-user'
SECRET = 'dysmsdvsa'
# Generate a random number as a string. We'll use this for file names later on
callnum = str(random.randint(1, 100000000))
# Taken from here, with thanks -
# http://ginstrom.com/scribbles/2007/11/19/parsing-multilingual-
# email-with-python/
def get_charset(message, default="ascii"):
"""Get the message charset"""
if message.get_content_charset():
return message.get_content_charset()
if message.get_charset():
return message.get_charset()
return default
# Taken from here, with thanks -
# http://ginstrom.com/scribbles/2007/11/19/parsing-multilingual-
# email-with-python/
def get_body(message):
"""Get the body of the email message"""
if message.is_multipart():
#get the plain text version only
text_parts = [part
for part in typed_subpart_iterator(message,
'text',
'plain')]
body = []
for part in text_parts:
charset = get_charset(part, get_charset(message))
body.append(unicode(part.get_payload(decode=True),
charset,
"replace"))
return u"\n".join(body).strip()
else: # if it is not multipart, the payload will be a string
# representing the message body
body = unicode(message.get_payload(decode=True),
get_charset(message),
"replace")
return body.strip()
# Read the e-mail message that has been piped to us by Postfix
raw_msg = sys.stdin.read()
emailmsg = email.message_from_string(raw_msg)
# Extract database Fields from mail
msgfrom = emailmsg['From']
msgto = emailmsg['To']
msgsubj = emailmsg['Subject']
msgbody = get_body(emailmsg)
# Find the part of the 'To' field that is the phone number
phonenum = re.match( r'\+?([0-9]+)', msgto, re.M)
# Whose mobile is this?
mobile = sys.argv[1]
# Write a log file in /tmp with a record of the e-mails
currtime = date.today().strftime("%B %d, %Y")
logfile = open('/tmp/email2sms.log', 'a')
logfile.write(currtime + "\n")
logfile.write("Call Number: " + callnum + "\n")
logfile.write("From: " + msgfrom + "\n")
logfile.write("To: " + msgto + "\n")
logfile.write("Subject: " + msgsubj + "\n")
logfile.write("Body: " + msgbody + "\n\n")
logfile.close()
# Send the call details to the Asterisk manager interface
s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.connect((HOST, PORT))
sleep(1)
s.send('Action: login\r\n')
s.send('Username: ' + USER + '\r\n')
s.send('Secret: ' + SECRET + '\r\n\r\n')
sleep(1)
s.send('Action: originate\r\n')
# Dummy channel - I don't actually want any phones to ring
s.send('Channel: LOCAL/1@sms-dummy\r\n')
s.send('Context: mobiles\r\n')
s.send('Exten: ' + mobile + '\r\n')
s.send('WaitTime: 30\r\n')
# This is a bogus value, but the field is required
s.send('CallerId: 5555\r\n')
# Do not wait for response from dummy channel
s.send('Async: true\r\n')
s.send('Priority: 1\r\n')
# The variables ${SMSTO} and ${SMSBODY} are used in the dialplan
s.send('Variable: SMSTO=' + phonenum.group(1) + ',SMSBODY=\"' + msgbody + '\"\r\n\r\n')
sleep(1)
s.send('Action: Logoff\r\n\r\n')
#Omitting this causes "ast_careful_fwrite: fwrite() returned error: Broken pipe"
sleep(3)
s.close()
Copy the above script to /usr/sbin/email2sms.py and make executable:
# chmod +x /usr/sbin/email2sms.pyThe script uses the Asterisk Manager Interface, so it will need an AMI user. Append this to
manager.conf:
# vi /etc/asterisk/manager.conf [your-ast-man-user] secret=dysmsdvsa read=call,user,originate write=call,user,originateand also make sure it is enabled in the general section:
# vi /etc/asterisk/manager.conf [general] enabled = yes webenabled = yes port = 5038You'll note that I'm using the context 'mobiles'. You'll need to make sure that the extensions you'll be using exist in this context in extensions.conf:
# vi /etc/asterisk/extensions.conf
exten => stocksy,1,MobileSendSMS(JS6021,${SMSTO},${SMSBODY})
exten => karen,1,MobileSendSMS(trollphone,${SMSTO},${SMSBODY})
Secondly, there is a dummy extension which the 'call' needs to connect to. A NoOp isn't
quite sufficient, I could only get it to work if the extension answered and then did something,
in this case answer and wait 10 seconds:
# vi /etc/asterisk/extensions.conf [sms-dummy] exten => 1,1,Answer() exten => 1,n,Wait(10) exten => 1,n,HangupReload Asterisk to pick up the changes.
So, calling email2sms.py with the argument 'stocksy' uses the JS6021 mobile, and calling it with 'karen' uses the trollphone mobile.
You need to make sure that email for the domain you have chosen - in my case sms.stocksy.co.uk - is routed to the Asterisk box. This will normally be accomplished by creating an MX record or creating a transport for the domain on your mail server. Again, I'm not going to cover that part here, but I will cover how to pipe the incoming messages into the python script.
Assuming that you are using postfix, you'll need a new transport for each mobile you want to use. In my case:
# vi /etc/postfix/master.cf sms-stocksy unix - n n - - pipe flags=FR user=stocksy argv=/usr/sbin/email2sms.py stocksy sms-karen unix - n n - - pipe flags=FR user=stocksy argv=/usr/sbin/email2sms.py karenpostfix needs to know that it must use these transports for SMS domains:
# vi /etc/postfix/transport ; postmap /etc/postfix/transport sms.stocksy.co.uk sms-stocksy sms.herdomain.co.uk sms-karenIf postfix doesn't already have a transport_maps setting, create one. Obviously this could break any existing postfix setup you might have, but if so I'm expecting you to know what you're doing:
# postconf -e transport_maps=hash:/etc/postfix/transportRestart postfix and that should be all that's necessary.
# /etc/init.d/postfix restartYou need to satisfy yourself that you are not allowing the entire world to relay through your SMS gateway! Understand and make use of postfix's security features! Don't wait until you've racked up a collosal SMS bill! Loud noises!
If things aren't quite working, start by checking your mail log:
# tail -f /var/log/mail.logYou can do a packet trace to see what's happening on the Asterisk Manager Interface:
# tcpdump -A -i lo port 5038Try talking to the AMI directly:
$ nc localhost 5038 Action: login Username: your-ast-man-user Secret: dysmsdvsa Action: originate Channel: LOCAL/1@sms-dummy Context: mobiles Exten: stocksy WaitTime: 30 CallerId: 5555 Async: true Priority: 1 Variable: SMSTO=5555555555,SMSBODY="foo" Action: LogoffWatch out for whitespace in the AMI - exten 'stocksy' != 'stocksy '.
Good luck.
Tuesday, 10 December 2013
Stop SIP Flood Attack
Source:http://kb.smartvox.co.uk/asterisk/friendlyscanner-gets-aggressive/
The port scanning probes used SIP OPTIONS while the really unpleasant, full-on, bandwidth-eating manifestation uses SIP REGISTER requests. The fact that in both cases the User Agent is declared to be “friendly-scanner” does not mean that this is a single application operating in two different modes. It probably indicates that some of the source code for these loathsome applications was derived from a common ancestor or that one is a re-worked version of the other. I believe sipvicious and a python program called svwar.py may have the dubious honour of being in some way the original seed for what has now turned into an irritating and potentially costly problem for VoIP users around the world. But should we blame gun manufacturers when there is an armed robbery or illegal shooting? Perhaps not. Anyway, I digress.
Making sense of what was happening was difficult because there did not seem to be any unauthorised calls and the internal office extensions were all working fine as were the analogue trunks that they use for inbound calls and as a backup in case of Internet problems. I knew the phone account passwords were all sufficiently strong and complex to not be hacked and other security settings such as “alwaysauthreject=yes” were configured for maximum resistance to attack, yet the firewall logs showed that there was a continuous heavy use of bandwidth from the Asterisk PBX to some address on the Internet. The fact that the outbound traffic was using approximately four times the bandwidth of the inbound made me think it must be some malware running on the server. This misconception caused me a considerable delay before I finally thought to run tcpdump and see exactly what was going on. By the way, the Asterisk CLI command “sip set debug on” will also show what is happening, but you may then find it difficult to turn off the sip debug because of the rate at which information is being written to the screen. Running “tcpdump udp -nn” at the Linux command prompt is safer because it just writes one line to the screen per request and Control-C is all that is required to stop it.
The outbound UDP packets outnumbered the inbound because Asterisk was sending a rejection for each registration attempt and then re-sending it when it didn’t get an ACK response – it would send as many as 5 responses to every one inbound REGISTER request. With inbound requests running at nearly 100 per second this was causing it a bit of a headache.
As soon as I blocked the inbound requests, the remote users were suddenly able to make calls again and everything went quiet. Job done I thought. Well, not quite.
So a quick solution at this point was to add a rule to iptables on the Asterisk box and let the requests come through the Draytek. That took the pressure off the Draytek, but no matter what ICMP response I sent back (including no response), the requests just kept flooding in. If anything, they were getting faster and using more bandwidth (see graph below).
I am now convinced that the unending nature of the attack is not deliberate, but is as a result of a bug in the code which puts it into an infinite loop. Clearly the people who produced these friendly-scanner hacking tools aren’t as clever as they would no doubt like to believe.
Copy the script from the box below and paste it into a file. I called my file spoof_sip_ok. (If you use a text editor like vi or nano while connected through SSH with Putty, then a right-click of the mouse will paste text previously copied into the clipboard). The original script is no longer available at Joshua’s own blog site, so I have reproduced the whole thing here with a couple of minor adjustments:
The original article was at “http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood”, but the whole domain seems to no longer be active (as of March 2013).
Before you run the above version of the script, make sure Asterisk is stopped. I found it would immediately restart when I used ”amportal stop” and instead used the CLI command “stop now”. You can check that it has stopped by typing this command at the Linux command prompt: netstat -lunp
If Asterisk is running, it will be using port 5060 and the output of the netstat command would look like this:
To run the Ruby script, just type ./spoof_sip_ok at the Linux prompt.
Not so friendly after all
In my October 2010 articles about Asterisk IP-PBX security (linked here), I described how port scanning probes from the so-called “friendly-scanner” could be seen several times a day on a typical SIP server exposed to the Internet. Since then, I – or at least one of my clients – had the displeasure of experiencing the full fury of this remarkably unfriendly scanner which, when provoked, seems to change from a gentle prod every few hours to a full scale Denial of Service attack at a rate of more than 80 SIP REGISTER requests per second, utterly relentless and lasting for days or even weeks.The port scanning probes used SIP OPTIONS while the really unpleasant, full-on, bandwidth-eating manifestation uses SIP REGISTER requests. The fact that in both cases the User Agent is declared to be “friendly-scanner” does not mean that this is a single application operating in two different modes. It probably indicates that some of the source code for these loathsome applications was derived from a common ancestor or that one is a re-worked version of the other. I believe sipvicious and a python program called svwar.py may have the dubious honour of being in some way the original seed for what has now turned into an irritating and potentially costly problem for VoIP users around the world. But should we blame gun manufacturers when there is an armed robbery or illegal shooting? Perhaps not. Anyway, I digress.
Symptoms of an attack
My client called me to say that their remote users were experiencing serious problems with their connection to the Asterisk phone system at the office. My own phone refused to register using exactly the same credentials as had worked a few days earlier. I tried to use remote access to check the server, but the connection was dreadfully slow to the point where it was unusable.Making sense of what was happening was difficult because there did not seem to be any unauthorised calls and the internal office extensions were all working fine as were the analogue trunks that they use for inbound calls and as a backup in case of Internet problems. I knew the phone account passwords were all sufficiently strong and complex to not be hacked and other security settings such as “alwaysauthreject=yes” were configured for maximum resistance to attack, yet the firewall logs showed that there was a continuous heavy use of bandwidth from the Asterisk PBX to some address on the Internet. The fact that the outbound traffic was using approximately four times the bandwidth of the inbound made me think it must be some malware running on the server. This misconception caused me a considerable delay before I finally thought to run tcpdump and see exactly what was going on. By the way, the Asterisk CLI command “sip set debug on” will also show what is happening, but you may then find it difficult to turn off the sip debug because of the rate at which information is being written to the screen. Running “tcpdump udp -nn” at the Linux command prompt is safer because it just writes one line to the screen per request and Control-C is all that is required to stop it.
The outbound UDP packets outnumbered the inbound because Asterisk was sending a rejection for each registration attempt and then re-sending it when it didn’t get an ACK response – it would send as many as 5 responses to every one inbound REGISTER request. With inbound requests running at nearly 100 per second this was causing it a bit of a headache.
Blocking the attack
As soon as I realised what was happening, it was relatively simple to add a firewall rule to block all inbound data from the single IP address that was sending the requests. Yes, it was all from one IP address and No, contacting the owners of that address was as much use as a chocolate teapot.As soon as I blocked the inbound requests, the remote users were suddenly able to make calls again and everything went quiet. Job done I thought. Well, not quite.
It just carried on and on
I assumed the attack would stop pretty quickly once the inbound packets were blocked at the firewall and no responses were coming back from Asterisk. Not so. I checked the traffic monitor on their firewall the next day and the day after. The inbound stream of requests was still there. After a week, the firewall began to creak and get sick because it was running out of memory. Well it is only a modest little Draytek router, but it is designed for a small business environment and should have been able to cope with the simple task of blocking a continuous stream of unwanted SIP requests coming from one IP address on the Internet. In my opinion, someone at Draytek should be shot for designing a router/firewall that adds an entry to the NAT sessions table before it inspects the firewall rules to see if the source address is blocked!So a quick solution at this point was to add a rule to iptables on the Asterisk box and let the requests come through the Draytek. That took the pressure off the Draytek, but no matter what ICMP response I sent back (including no response), the requests just kept flooding in. If anything, they were getting faster and using more bandwidth (see graph below).
I am now convinced that the unending nature of the attack is not deliberate, but is as a result of a bug in the code which puts it into an infinite loop. Clearly the people who produced these friendly-scanner hacking tools aren’t as clever as they would no doubt like to believe.
How to stop it completely
The answer was in a blog by Joshua Stein, to whom I am indebted. His solution involved redirecting the requests to a new port so it would not be necessary to shut down the Asterisk PBX application. However, my client didn’t need 24×7 operation of their phone system so instead I just modified Joshua’s Ruby script to use port 5060, stopped Asterisk, ran the Ruby script and that was it. The requests just stopped. Then I restarted Asterisk. This graph from the firewall shows the bandwidth being used over a week and the abrupt end of the problem when I finally ran Joshua’s script.
Graph of Internet bandwidth
Using Joshua’s Ruby script on a CentOS server
Install Ruby using yum: yum install rubyCopy the script from the box below and paste it into a file. I called my file spoof_sip_ok. (If you use a text editor like vi or nano while connected through SSH with Putty, then a right-click of the mouse will paste text previously copied into the clipboard). The original script is no longer available at Joshua’s own blog site, so I have reproduced the whole thing here with a couple of minor adjustments:
#!/usr/bin/env ruby
require "socket"
s = UDPSocket.new
s.bind("0.0.0.0", 5060)
while true
packet = s.recvfrom(1024)
via = packet[0].match(/Via: (.+);rport/)[1]
from = packet[0].match(/From: (.+)/)[1]
to = packet[0].match(/To: (.+)/)[1]
call_id = packet[0].match(/Call-ID: (.+)/)[1]
cseq = packet[0].match(/CSeq: (\d+) REGISTER/)[1]
remote_ip = packet[1][3]
remote_port = packet[1][1].to_i
puts packet.inspect
if packet[0].match(/^REGISTER /)
ret = "SIP/2.0 200 OK\r\n" +
"Via: #{via};received=#{remote_ip}\r\n" +
"From: #{from}\r\n" +
"To: #{to}\r\n" +
"Call-ID: #{call_id}\r\n" +
"CSeq: #{cseq.to_i + 1} REGISTER\r\n" +
"\r\n"
puts "sending to #{remote_ip}:#{remote_port}:\n#{ret}"
s.send(ret, 0, remote_ip, remote_port)
end
end
You can insert your own server’s IP address in the s.bind parameters if you want, but 0.0.0.0 should just bind to all interfaces. Also remember to make the file executable, for example using the Linux command chmod 755 spoof_sip_ok
The original article was at “http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood”, but the whole domain seems to no longer be active (as of March 2013).
Before you run the above version of the script, make sure Asterisk is stopped. I found it would immediately restart when I used ”amportal stop” and instead used the CLI command “stop now”. You can check that it has stopped by typing this command at the Linux command prompt: netstat -lunp
If Asterisk is running, it will be using port 5060 and the output of the netstat command would look like this:
user@asterisk:~ $ netstat -lunp Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp 0 0 0.0.0.0:5060 0.0.0.0:* 16301/asterisk
To run the Ruby script, just type ./spoof_sip_ok at the Linux prompt.
Friday, 22 November 2013
Asterisk queues.conf
Source: http://www.syednetworks.com/asterisk-queues-conf
In Asterisk queues.con file is where we define queues and the way that how to handle the callers.
• Reasonable queuing support within Asterisk
• Queues can have static or dynamic members
• Members can be channels, or Agents
• Automatic distribution of calls based on queue strategy
Here is queues.conf options are available:
In Asterisk queues.con file is where we define queues and the way that how to handle the callers.
• Reasonable queuing support within Asterisk
• Queues can have static or dynamic members
• Members can be channels, or Agents
• Automatic distribution of calls based on queue strategy
Here is queues.conf options are available:
persistentmembers = yes, so that when you restart Asterisk the agents will be automatically readded into their recorded queues. By default the option is set to yes.
Each queue is a separate section. For example, a snake in the [support]. The following parameters are available:
musiconhold In musiconhold.conf file defined category of music queue. We choose the top-scale Category default:
musiconhold = default
announce
you can specify an announcement, which to be played to the agents, when they answer the incoming call. Usually this option is used to inform the agents, which queue exactly, they will answer. This is made for agents which are set in more than one queue.
; announce = Technical-support
If you use this option, make sure that file technical-support.gsm should be availabe in the sound directory the default path of sound directory is /var/lib/asterisk/sounds.
Strategy
Determines how and what’s the order of the agents when a call comes to this queue:
ringall
All available agent should ring until one responds. (Default)
roundrobin
ring the each available agent one by one.
leastrecent
Call the agent who is longest idle in available agents
fewestcalls
ring the agent with fewest completed calls from this queue.
Random
Select the agent randomly from this queue.
rrmemory
Round-robin with memory. Starts the series in which, after the last call to the series.
In Asterisk 1.6 will probably be the roundrobin procedure is forfeited and rrmemory in roundrobin renamed.
It should be noted that agents, for which a lower penalty (penalty) has been established agent who always preferred a higher penalty.
Strategy = ringall
The setting depends on your circumstances: ringall may be annoying, while others cause additional waiting time for the caller, if agents not connected to the phone.
Level
Sets the time (in seconds), in which calls should be answered. Only for statistical analysis interesting ( “How many calls were made within the service time of x seconds answered?”).
Service = 60
Context
Here can be a context where you want to send the caller, if the caller presses the key (the one you want the caller to be pressed) the extension will be dialed from the mentioned queue.
context = support-context – so we could adjust it
Timeout
Determines how long (in seconds) a phone ringing, until we are not as busy (timeout).
timeout = 15
retry
Determines how long (in seconds) to wait before all agents phoned again.
retry = 5
weight
Weight of queue – when compared to other queues, higher weights get preference
weight = 0
wrapuptime
how long before sending agent another call
We let our agents a little breather to a sip of water to drink:
wrapuptime = 10
maxlen
Maximum number of the callers in the queue (default: 0 for unlimited).
maxlen = 0
announce-frequency
Certainly, at what intervals (in seconds) the callers their position in the queue and / or the estimated waiting time will be announced (0 for nothing i.e. announce-frequency = 0 means off).
announce-frequency = 90
announce-holdtime
Whether the estimated waiting time in the announcements for the position would be called. Possible values are yes (yes), no (no) or once (only once).
announce-holdtime = Yes
Round-announce-seconds
Rounding steps of seconds. With the setting of 0, only minutes, not seconds announced. Other possible values are: 0, 1, 5, 10, 15, 20 and 30 [72] (At 30, for example, would an estimated waiting time from 2:34 to 2:30 rounded.)
announce-round-seconds = 0
Language blocks
The following parameters are, what language blocks for the announcements of the position and waiting periods. I.d.R changes are not necessary:
queue-youarenext = queue-youarenext; “You are now first in line.”
queue-thereare = queue-thereare; “There are” …
queue-callswaiting = queue-callswaiting … “Calls waiting.”
queue-holdtime = queue-holdtime; “The current est. holdtime is” …
queue-minutes = queue-minutes … “Minutes”
queue-seconds = queue-seconds … “Seconds”
queue-thankyou = queue-thankyou, “Thank you for your patience.”
queue-lessthan = queue-less-than … “less than” …
queue-reporthold = queue-reporthold; “hold time” …
periodic-announce = queue-periodic-announce, “All reps busy, wait for next”
If these parameters are not specified (out), these default values.
periodic-announce-frequency
Certainly, at what intervals (in seconds) the callers the periodic announcement (periodic-announce, “There are all employees in an interview, please wait.”) To be announced.
periodic-announce-frequency = 60
Monitor format
The indication of this parameter turns the recording of interviews (as with the monitor () Application) and determines the recording format. (For auskommentiertem monitor-format recordings are not made.) Enter here gsm, wav (large files) or wav49.
If you want records the call in the gsm format uncomment this line, please note the call will be recorded when agent answeres the call:
; monitor-format = gsm
By default, the files (one-in-and out) to $ UNIQUEID) (named. You can change if necessary, by using the dial plan before the queue () call Set (MONITOR_FILENAME = filename). We take the example but no change.
Join Monitor
Combine the two by a call recording files generated …- …- in and out to a file. Values: yes or no
Monitor-join = Yes
joinempty
Determines whether callers in a queue without agents indicated.
Yes
Callers can be placed in a loop without agents or by non-agents available for classification purposes.
No.
Callers can not in a loop without agents indicated.
strict
Callers can not in a loop without agents indicated (but in a snake with only non-available agents).
Non-available should not be confused with the conversation (busy). An agent is not available when he is assigned to the snake, but not actually registered in the system (see also “member”). Caution: Static defined members (see “member”) are always available!
If a caller not to be a snake, the queue () application quits, and the Dial plan will continue.
joinempty = no, we want our callers are not unnecessarily wait
leavewhenempty
It will determines whether as soon as all the agents out the call will not go to this queue. Possible values as joinempty. After leaving the queue, the dial plan.
leavewhenempty = strict; callers are not unnecessarily wait
eventwhencalled
If a (yes | no), whether etc. the following events for the Manager interface is used: Called Agent, Agent dump, Agent Connect, Agent Complete.
eventwhencalled = Yes
eventmemberstatus
If a (yes | no), whether members queue status events in the produce manager interface (can be many).
eventmemberstatus = No
reportholdtime
Sets (yes | no), whether to the agent the waiting time of the caller will be announced before it entertained the caller. (Taste issue)
reportholdtime = No
memberdelay
Where, how long (in seconds) the caller hears silence before he connected with an agent.
memberdelay = 1
timeoutrestart
Determines whether the response time of an agent in a busy or overloaded pipe signal is reset. Can be useful for agents who have the right to reject a call.
timeoutrestart = Yes
autopause
Autopause will pause a queue member if they fail to answer a call.
ringinuse
Determines whether such agent phoned to be known of which is that they are in conversation. The channel so far the only driver, the status transmitted is SIP.
ringinuse = No
Member
It is possible to intervene directly in the queues.conf agents in the form of static
member => Technology Resource [, Malus]
– Also e.g. member => Zap / 2 – them (may be used several times, see queues.conf). But this can result in problems with joinempty and leavewhenempty, since these agents always be available, even if it is in fact not at their apparatus. It also has the disadvantage that always finds an agent is assigned to an apparatus and not from another apparatus from register.
We therefore prefer to use dynamic form and arrange queue support in the form:
member => Agent / AgentenNr
two agents 1001 and 1002 to:
member => Agent/1001
member => Agent/1002
Wednesday, 20 November 2013
Clean up Asterisk sample config files
sed '/^;/d' manager.conf.sample | sed '/^[ \t]/d' | sed '/^$/d' > manager.conf
Friday, 20 September 2013
Asterisk vs FreeSWITCH
SOURCE: http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk and FreeSWITCH are
B2BUA (Back-to-back user agents). They act as a user agent for two or
more ends, and use a common protocol like the Session Initiation
Protocol (SIP). The B2BUA is responsible for handling all SIP signaling
between the end-points, route calls, transfer calls, translate codecs
between the call legs and offer value-added features to the call such as
voicemail and call transfer.
In other words, they are telephony platforms designed to facilitate the creation of voice and chat driven products. FreeSWITCH in particular, can scale from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the call flow.
Isn’t FreeSWITCH a soft-switch and Asterisk a PBX? What is the difference?
Extracted from the FreeSWITCH FAQ:
"A PBX is an entity that allows a private company to have its own mini phone company providing services like voicemail, extensions and conferencing to phones. The primary focus of a PBX is for multiple phones to find each other and communicate. A soft-switch is a software application that can connect phone lines from one network to another, often routing calls from one protocol to another or to a termination point such as a PBX. FreeSWITCH has the potential to also implement a PBX but it is not mandated to. Think of it as a lower level application than a PBX. It’s possible to load several modules into FreeSWITCH to make it behave exactly like a cluster of PBX applications. This is entirely easier than trying to make a single monolithic PBX behave as a switch especially when much of the PBX functionality is permanently built into the core of the PBX application."
My first experience with Asterisk at RAC
My first real Asterisk experience began earlier this year when my boss, Adil Wali, asked me to set up a PBX server for RAC. I had done some previous Asterisk work so I already had some experience with it, but this was the first time I had deployed Asterisk in a large-sized company.
The time came and I deployed an Asterisk server for RAC. Right now it does inbound/outbound, IVR (Interactive Voice Response), conferences, voicemail, call forwarding and almost everything else you would expect from a nice VoIP (PBX) server. There are some things that some of us would like to have that are not currently available in the Asterisk software, such as dialing out from conferences and soft-based conferences. A few employees have asked me for the option to dial out from conferences but I cannot yet add this to our configuration because the feature doesn't exist in Asterisk. It is a technical limitation of Asterisk but it is available on FreeSWITCH. This is one of the reasons that FreeSWITCH would be better suited for us.
I have a fair amount of Asterisk knowledge now. I have used it for some time and I know its merits and its problems. The nice thing about Asterisk is that it's really easy to set up and configure the way you want. The bad thing about it is that it has some internal problems on its core and sometimes it gives you unexpected problems. It also lacks some of the features we need.
Asterisk issues and bugs I have found in the software
I ran into some problems when I tried to deploy Asterisk on a VPS first (Amazon EC2). When I tried to do conferencing, I needed the MeetMe application that provides conferences, and MeetMe depends on Zaptel (which is another piece of software that provides the timing that MeetMe needs). I tried to install Zaptel, but Zaptel refused to work on XEN (the virtualization software that our VPS uses). So I didn't have any other choice but to go with a dedicated server for our VoIP needs. FreeSWITCH works great on a VPS with all the conferencing features, and everything out of the box - no zaptel, ztdummy or anything that could interfere is needed.
I tried very hard to make Asterisk work with the conference features on the VPS (Amazon EC2). I also tried with other conference applications like app_conference. This doesn't require an external timing source like MeetMe does, but it lacks some features that we needed, such as sound notifications. MeetMe had this but MeetMe wasn't an option on the VPS because of its technical requirements. I also tried to get support on Asterisk and their response to my problem was that it was "irrelevant".
This was the first time I started asking myself, "Why do I need a ztdummy driver (Zaptel) in order to use the conference module? Why can't the conference module use it's own internal timing source?" I thought there was something wrong with this design. That was when Brian West (also know as bkw_ - a fellow core-developer and friend of mine) told me about FreeSWITCH and its nice soft-based conference module. That was the first time I heard about FreeSWITCH and when I saw the light.
I also experienced DTMF problems and sound corruption with Asterisk. This happens because the Asterisk core is broken. I tried doing the same things in FreeSWITCH and everything worked great.
After all of the work that I put into making Asterisk work on our server, I keep discovering more and more problems with it. The first problem I discovered after everything was working was the DTMF problem. One day an employee asked me why his password wasn't working. I went and looked at the server and his password was there and everything looked good. Then I tried to debug it and I put a function on the dialplan that returned the digits that you send to the server and to my surprise, the server was printing bad/broken digits. So after a lot of research, I have come with the conclusion that the DTMF in Asterisk was broken.
I reported a bug about this issue, but it is still open and not resolved two months after I reported it: bugs.digium.com/view.php
One workaround for this problem was to use rfc2833compensate=yes in the sip configuration, in order to compensate the DTMF digits or RTP packets... this fixed it with X-Lite but the problem still happens with Zoiper. I'm sure this is the reason that passwords fail for conferences sometimes.
Another problem I had was with sound. The sound gets corrupted after a while with the Speex codec. Again I reported a bug of this: bugs.digium.com/view.php
I have also experienced some deadlocks with Asterisk as reported here: bugs.digium.com/view.php
I have tried the same things with FreeSWITCH and I haven't experienced the problems. It all went very smoothly and FreeSWITCH has some features that Asterisk doesn't, such as dialing out from conferences.
I came to the point where I realized that we need a better solution. Asterisk has too many problems for our company. I decided to go with FreeSWITCH in order to have a high quality PBX.
On the way to a better VoIP solution with FreeSWITCH
The learning curve on FreeSWITCH for me wasn't that bad. However, I didn't know so it was a bit tedious for me at first. I now realize that the XML configuration is a good thing. Regular expressions in the dialplan make it succinct and the XML makes adding customized features very easy and straightforward. FreeSWITCH is a class-five soft-switch that can be molded the way you want, from a PBX to carrier-grade soft-switch to anything you can ever imagine, and it's very extensible. The design that its creator went with is excellent. It is a masterpiece and there is nothing in the open source world or in the proprietary world that comes even close to it. I think that it is a great piece of work.
One of the big advantages of FreeSWITCH is that it's core is better designed, with a lot of care in the details. The primary author of FreeSWITCH is a former developer that spent a lot of time working with the flawed Asterisk design and finally gave up and decided to build a VoIP system that would be stable, scalable and extensible. The stable part means that we'll all be able to sleep well at night.
One of the nice things about FreeSWITCH is the community. The developers themselves and the users all hang in one channel - #freeswitch (at freenode). The mailing list is also a very nice place. In both places they are very friendly and supportive, unlike the Asterisk/Digium community. They are also open-minded. At one point I asked for a feature (mod_yaml) and the creator of FreeSWITCH (Anthony Minessale) came up with this feature in less than 3 hours - without even knowing what YAML was when the feature was asked for. It was really impressive. He came up with a whole new feature that he hadn't even heard of yet, while the Asterisk devs can't even fix little bugs. Here is the proof of this: lists.freeswitch.org/pipermail/freeswitch-users/2008-June/003954.html
Another nice thing about FreeSWITCH is that it doesn't suffer from the "Not Invented Here" (NIH) syndrome that Asterisk suffers from. FreeSWITCH uses the Sofia SIP stack, a 100% RFC compliant SIP stack, which is very complete, robust and mature. Nokia also actively contributes in this stack along with the FS developers. They also use XML and PCRE (Perl Compatible Regular Expressions) for the dialplan, the Apache Portable Runtime (APR) and SQLite. FreeSWITCH has the ability to load scripts written in LUA, Python, Perl, PHP and XML in the dialplan as "applications” so you have a lot of flexibility and freedom in how you want to configure your VoIP system.
I am very happy that I made the decision to switch to FreeSWITCH. Employees in the company have already reported better audio quality and stability. FreeSWITCH is clearly the winner here. The project is still young and it has a long way to go, but the core and the foundations are very mature and strong and that makes it a great platform.
A few articles
FreeSwitch Attempts To Punctuate Asterisk! FreeSwitch and SipXecs and Second Generation IP PBX FreeSwitch - Trying on the Big Boy Pants Asterisk vs. FreeSWITCH (anders.com) FreeSWITCH, a Second-Generation Open-Source Communications Platform digg.com/software/Open_Source_VoIP_Asterisk_or_FreeSwitch digg.com/software/an_open_source_class_five_telephony_switch
In other words, they are telephony platforms designed to facilitate the creation of voice and chat driven products. FreeSWITCH in particular, can scale from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the call flow.
Isn’t FreeSWITCH a soft-switch and Asterisk a PBX? What is the difference?
Extracted from the FreeSWITCH FAQ:
"A PBX is an entity that allows a private company to have its own mini phone company providing services like voicemail, extensions and conferencing to phones. The primary focus of a PBX is for multiple phones to find each other and communicate. A soft-switch is a software application that can connect phone lines from one network to another, often routing calls from one protocol to another or to a termination point such as a PBX. FreeSWITCH has the potential to also implement a PBX but it is not mandated to. Think of it as a lower level application than a PBX. It’s possible to load several modules into FreeSWITCH to make it behave exactly like a cluster of PBX applications. This is entirely easier than trying to make a single monolithic PBX behave as a switch especially when much of the PBX functionality is permanently built into the core of the PBX application."
My first experience with Asterisk at RAC
My first real Asterisk experience began earlier this year when my boss, Adil Wali, asked me to set up a PBX server for RAC. I had done some previous Asterisk work so I already had some experience with it, but this was the first time I had deployed Asterisk in a large-sized company.
The time came and I deployed an Asterisk server for RAC. Right now it does inbound/outbound, IVR (Interactive Voice Response), conferences, voicemail, call forwarding and almost everything else you would expect from a nice VoIP (PBX) server. There are some things that some of us would like to have that are not currently available in the Asterisk software, such as dialing out from conferences and soft-based conferences. A few employees have asked me for the option to dial out from conferences but I cannot yet add this to our configuration because the feature doesn't exist in Asterisk. It is a technical limitation of Asterisk but it is available on FreeSWITCH. This is one of the reasons that FreeSWITCH would be better suited for us.
I have a fair amount of Asterisk knowledge now. I have used it for some time and I know its merits and its problems. The nice thing about Asterisk is that it's really easy to set up and configure the way you want. The bad thing about it is that it has some internal problems on its core and sometimes it gives you unexpected problems. It also lacks some of the features we need.
Asterisk issues and bugs I have found in the software
I ran into some problems when I tried to deploy Asterisk on a VPS first (Amazon EC2). When I tried to do conferencing, I needed the MeetMe application that provides conferences, and MeetMe depends on Zaptel (which is another piece of software that provides the timing that MeetMe needs). I tried to install Zaptel, but Zaptel refused to work on XEN (the virtualization software that our VPS uses). So I didn't have any other choice but to go with a dedicated server for our VoIP needs. FreeSWITCH works great on a VPS with all the conferencing features, and everything out of the box - no zaptel, ztdummy or anything that could interfere is needed.
I tried very hard to make Asterisk work with the conference features on the VPS (Amazon EC2). I also tried with other conference applications like app_conference. This doesn't require an external timing source like MeetMe does, but it lacks some features that we needed, such as sound notifications. MeetMe had this but MeetMe wasn't an option on the VPS because of its technical requirements. I also tried to get support on Asterisk and their response to my problem was that it was "irrelevant".
This was the first time I started asking myself, "Why do I need a ztdummy driver (Zaptel) in order to use the conference module? Why can't the conference module use it's own internal timing source?" I thought there was something wrong with this design. That was when Brian West (also know as bkw_ - a fellow core-developer and friend of mine) told me about FreeSWITCH and its nice soft-based conference module. That was the first time I heard about FreeSWITCH and when I saw the light.
I also experienced DTMF problems and sound corruption with Asterisk. This happens because the Asterisk core is broken. I tried doing the same things in FreeSWITCH and everything worked great.
After all of the work that I put into making Asterisk work on our server, I keep discovering more and more problems with it. The first problem I discovered after everything was working was the DTMF problem. One day an employee asked me why his password wasn't working. I went and looked at the server and his password was there and everything looked good. Then I tried to debug it and I put a function on the dialplan that returned the digits that you send to the server and to my surprise, the server was printing bad/broken digits. So after a lot of research, I have come with the conclusion that the DTMF in Asterisk was broken.
I reported a bug about this issue, but it is still open and not resolved two months after I reported it: bugs.digium.com/view.php
One workaround for this problem was to use rfc2833compensate=yes in the sip configuration, in order to compensate the DTMF digits or RTP packets... this fixed it with X-Lite but the problem still happens with Zoiper. I'm sure this is the reason that passwords fail for conferences sometimes.
Another problem I had was with sound. The sound gets corrupted after a while with the Speex codec. Again I reported a bug of this: bugs.digium.com/view.php
I have also experienced some deadlocks with Asterisk as reported here: bugs.digium.com/view.php
I have tried the same things with FreeSWITCH and I haven't experienced the problems. It all went very smoothly and FreeSWITCH has some features that Asterisk doesn't, such as dialing out from conferences.
I came to the point where I realized that we need a better solution. Asterisk has too many problems for our company. I decided to go with FreeSWITCH in order to have a high quality PBX.
On the way to a better VoIP solution with FreeSWITCH
The learning curve on FreeSWITCH for me wasn't that bad. However, I didn't know so it was a bit tedious for me at first. I now realize that the XML configuration is a good thing. Regular expressions in the dialplan make it succinct and the XML makes adding customized features very easy and straightforward. FreeSWITCH is a class-five soft-switch that can be molded the way you want, from a PBX to carrier-grade soft-switch to anything you can ever imagine, and it's very extensible. The design that its creator went with is excellent. It is a masterpiece and there is nothing in the open source world or in the proprietary world that comes even close to it. I think that it is a great piece of work.
One of the big advantages of FreeSWITCH is that it's core is better designed, with a lot of care in the details. The primary author of FreeSWITCH is a former developer that spent a lot of time working with the flawed Asterisk design and finally gave up and decided to build a VoIP system that would be stable, scalable and extensible. The stable part means that we'll all be able to sleep well at night.
One of the nice things about FreeSWITCH is the community. The developers themselves and the users all hang in one channel - #freeswitch (at freenode). The mailing list is also a very nice place. In both places they are very friendly and supportive, unlike the Asterisk/Digium community. They are also open-minded. At one point I asked for a feature (mod_yaml) and the creator of FreeSWITCH (Anthony Minessale) came up with this feature in less than 3 hours - without even knowing what YAML was when the feature was asked for. It was really impressive. He came up with a whole new feature that he hadn't even heard of yet, while the Asterisk devs can't even fix little bugs. Here is the proof of this: lists.freeswitch.org/pipermail/freeswitch-users/2008-June/003954.html
Another nice thing about FreeSWITCH is that it doesn't suffer from the "Not Invented Here" (NIH) syndrome that Asterisk suffers from. FreeSWITCH uses the Sofia SIP stack, a 100% RFC compliant SIP stack, which is very complete, robust and mature. Nokia also actively contributes in this stack along with the FS developers. They also use XML and PCRE (Perl Compatible Regular Expressions) for the dialplan, the Apache Portable Runtime (APR) and SQLite. FreeSWITCH has the ability to load scripts written in LUA, Python, Perl, PHP and XML in the dialplan as "applications” so you have a lot of flexibility and freedom in how you want to configure your VoIP system.
I am very happy that I made the decision to switch to FreeSWITCH. Employees in the company have already reported better audio quality and stability. FreeSWITCH is clearly the winner here. The project is still young and it has a long way to go, but the core and the foundations are very mature and strong and that makes it a great platform.
A few articles
FreeSwitch Attempts To Punctuate Asterisk! FreeSwitch and SipXecs and Second Generation IP PBX FreeSwitch - Trying on the Big Boy Pants Asterisk vs. FreeSWITCH (anders.com) FreeSWITCH, a Second-Generation Open-Source Communications Platform digg.com/software/Open_Source_VoIP_Asterisk_or_FreeSwitch digg.com/software/an_open_source_class_five_telephony_switch
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