So, we thought we'd go over a couple of the differences of SIP and IAX and why IAX may suit your needs better.

IAX:

• Only one port needed on firewall
• Works much easier with NAT environments
• Smaller network footprint
• Out of Band DTMF for ease of use
• Created by the Asterisk Team (well supported)

SIP:

• Standard Protocol
• Works with not only Asterisk
• Great if you're not NATing
• Well Supported
• Higher Bandwidth usage than IAX
• Can be a problem with DTMF sometimes

Those are pretty much the bare facts about SIP vs IAX. We're not out to start a war between which one is better, obviously both have their benefits and disadvantages. For us, we like using IAX with any remote office for trunking purposes or for external lines behind firewalls we don't control. For inside the office, we opt for SIP as much nicer phones are available with this protocol.

Which do you prefer?

Don't forget to check out the contest from VOIPSUPPLY while you're at it.

As of 5-31-08 to obtain your sip credentials you will need to dump your memory while magicjack.exe is running in order to view the decrypted password.
All other information can be had with any packet capture program.

Original, and All credit goes here: http://revolution.hackthisbox.com/magicjack/readme

Replace EXXXXXXXXXX01 with your MJ number. Include E and 01.
Replace the proxy proxy1.Atlanta.talk4free.com:5070 with the proxy your MJ registers to and change host=67.90.138.70 to host=YourProxyIPHere.
Replace XXXXXpasswordXXXXX with your password. Currently a 20 character string consisting of numbers and letters. Mine is all uppercase.

~~~~~sip.conf~~~~~

register => EXXXXXXXXXX01:XXXpasswordXXXX@proxy1.Atlanta.talk4free.com:5070

[magicjack]
context=incoming
username=EXXXXXXXXXX01
type=friend
secret=XXXXXpasswordXXXXX
port=5070
nat=yes
insecure=very
host=67.90.138.70
fromuser=EXXXXXXXXXX01
dtmfmode=inband
qualify=2000

~~~~~sip.conf~~~~~

~~~~~extensions.conf~~~~~

[incoming]
exten => YourMJNumber,1,Answer
exten => YourMJNumber,2,Dial(sip/sipura,30,r) ;dial someone...such as an ATA

[MagicJackOutgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@magicjack,30,r)
exten => _1NXXNXXXXXX,2,congestion()
exten => _1NXXNXXXXXX,102,busy()
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup

[sip]
include => MagicJackOutgoing

~~~~~extensions.conf~~~~~

 

 

 

Hello all,

not sure this belongs here, but I'm wondering if anyone else has
received "phone spam" lately. Over the last eight weeks, I've received
over 60 calls to my toll-free numbers, originating from various fake
numbers in the 940 and 956 area codes. When I still *listened* to them,
they began with "for all your office needs go to smtmco.com."

Whenever I call the numbers back (based on caller id), I get one or two
rings, then a click, then dead air. That's also the reason I'm posting
this here -- these folks either have access to a large pool of numbers,
or to a list of "dead" numbers. The last six numbers (three today) are:
940.387.0483
956.982.1640
940.891.6197
956.554.7617
940.891.6099
956.421.3378

The 956 numbers are all SWBell exchanges, while the 940s are GTE/Verizon
exchanges. All calls came in through my toll-free numbers (which in
turn are hosted by vitelity/exgn).

The website referenced isn't loading, but registered to:

Registrant:
Sales Team
321 High School Rd NE, PMB 348
Suite D3
Bainbridge Island, Washington 98110
United States
(800) 921-0136 ; NS1.ORDERSHOPPER.COM

Ordershopper.com also seems a dead-end, as it's registered "by proxy",
as all good spammers do.

I have since established a greylisting function, which simply does the
zapateller and then prompts the caller to call back within ten minutes
to be put through. If the same caller (by ID) calls the same number
within those 10 minutes, he's put through to a person and whitelisted.
If the caller hangs up before hearing the message (during the SIT tone),
the numbers is marked for blacklisting. Outgoing calls are
automatically whitelisted, as are certain (local) area codes. This has
kept the phone silent for a couple of weeks, and afaik, no valid calls
were killed.

But the real question is -- has anyone else seen this?

1. Install madplay

# apt-get install madplay

2. Login to the FreePBX Control Console and create a Music On Hold context. Next you will want to upload all your mp3's to this context.

3. Once your satisfied with the mp3's you've uploaded to your context, you will have to ssh to your box

# ssh your.pbx.ip

4. After logging in, and su'ing to root, go to /etc/asterisk and edit musiconhold_additional.conf. It should end up looking like this. Replace [Flewid] with your Music on Hold context.

[Flewid]
mode=custom
directory=/var/lib/asterisk/mohmp3/Flewid/
application=/usr/bin/madplay -Qzr -o raw:- --mono -R 8000 -a -12
random=yes

5. Now you will have to stop and start asterisk. Something like this should do the trick.

# /etc/init.d/asterisk stop
# /etc/init.d/asterisk start

6. Now when you issue a ps you should see that madplay is now running. You should see something like the following.

# ps aux | grep madplay

asterisk 15921 0.0 0.4 5732 2156 ? S Nov21 0:00 /usr/bin/madplay -Qzr -o raw:- --mono -R 8000 -a -12 Party 1992 Intromusic.mp3 A Touch of Spring.mp3 Ice Frontier.mp3 imphobia.mp3 Aquaphobia - 1993 spring.mp3 Lavender Hill.mp3 M16A.mp3 M16C.mp3 MENU.mp3 Minimum Velocity.mp3 Purple Sky II.mp3 Satellite one..mp3 technology.mp3 Turbulence.mp3 Unreal Symphony.mp3 When the heavens fall.mp3

7. That's it. Now you should have fancy music on hold that won't start at the beginning of every song, and when people are put on music on hold they will hear the song immediately, instead of waiting for it to start.

There is one problem with this method, in that if you decide to upload any more music on hold files to your context, musiconhold_additional.conf will get overwritten, and you will have to apply this fix again.

It would be nice if the freepbx developers added the option of using madplay from the administration interface. But this isn't really too big of a deal since musiconhold is changed infrequently (at least for us).

It wasn't very hard to get this done, and we've submitted a patch to FreePBX. Hopefully they will incorporate this into their next release and also add it for IAX Devices as well.

What This Hack Does:

1. Allow you to specify secret or md5secret for peer
2. Allow you to type plaintext secret in md5secret field, and it will run md5sum

What This Hack Doesn't Do:

1. Add MD5Secret Ability to IAX Devices
2. Work if your asterisk realm is set to something other than "asterisk".
3. Allow device passwords longer than 32 characters.
4. Display Popup warnings if no passwords are entered

How to enable this on your FreePBX / Asterisk 1.4 Installation:

1. Edit the Functions.inc.php in the core modules directory of freepbx
   

   # cd /var/www/admin/modules/core/functions.inc.php
   # cp functions.inc.php functions.inc.php.original
   # nano functions.inc.php

2. Look for the function named "core_Devices_addsip" and replace it with the following:
   

   //add to sip table
   function core_devices_addsip($account) {
   global $db;
   global $currentFile;

   foreach ($_REQUEST as $req=>$data) {
   if ( substr($req, 0, 8) == 'devinfo_' ) {
   $keyword = substr($req, 8);
   if ( $keyword == 'dial' && $data == '' ) {
   $sipfields[] = array($account, $keyword, 'SIP/'.$account);
   } elseif ($keyword == 'mailbox' && $data == '') {
   $sipfields[] = array($account,'mailbox',$account.'@device');
   } elseif ($keyword == 'md5secret' && $data != '') {
   $sipfields[] = array($account, 'md5secret', md5($account.':asterisk:'.$data));
   } else {
   $sipfields[] = array($account, $keyword, $data);
   }
   }
   }

3. Directly following the "core_devices_addsip" function, is the sipfields array. Replace it with the following piece of code:
   

   if ( !is_array($sipfields) ) { // left for compatibilty....lord knows why !
   $sipfields = array(
   //array($account,'account',$account),
   array($account,'accountcode',(isset($_REQUEST['accountcode']))?$_REQUEST['accountcode']:''),
   array($account,'secret',(isset($_REQUEST['secret']))?$_REQUEST['secret']:''),
   array($account,'md5secret', (isset($_REQUEST['md5secret']))? $_REQUEST['md5secret']:''),
   array($account,'canreinvite',(isset($_REQUEST['canreinvite']))?$_REQUEST['canreinvite']:'no'),
   array($account,'context',(isset($_REQUEST['context']))?$_REQUEST['context']:'from-internal'),
   array($account,'dtmfmode',(isset($_REQUEST['dtmfmode']))?$_REQUEST['dtmfmode']:''),
   array($account,'host',(isset($_REQUEST['host']))?$_REQUEST['host']:'dynamic'),
   array($account,'type',(isset($_REQUEST['type']))?$_REQUEST['type']:'friend'),
   array($account,'mailbox',(isset($_REQUEST['mailbox']) && !empty($_REQUEST['mailbox']))?$_REQUEST['mailbox']:$account.'@device'),
   array($account,'username',(isset($_REQUEST['username']))?$_REQUEST['username']:$account),
   array($account,'nat',(isset($_REQUEST['nat']))?$_REQUEST['nat']:'yes'),
   array($account,'port',(isset($_REQUEST['port']))?$_REQUEST['port']:'5060'),
   array($account,'qualify',(isset($_REQUEST['qualify']))?$_REQUEST['qualify']:'yes'),
   array($account,'callgroup',(isset($_REQUEST['callgroup']))?$_REQUEST['callgroup']:''),
   array($account,'pickupgroup',(isset($_REQUEST['pickupgroup']))?$_REQUEST['pickupgroup']:''),
   array($account,'disallow',(isset($_REQUEST['disallow']))?$_REQUEST['disallow']:''),
   array($account,'allow',(isset($_REQUEST['allow']))?$_REQUEST['allow']:'')
   //array($account,'record_in',(isset($_REQUEST['record_in']))?$_REQUEST['record_in']:'On-Demand'),
   //array($account,'record_out',(isset($_REQUEST['record_out']))?$_REQUEST['record_out']:'On-Demand'),
   //array($account,'callerid',(isset($_REQUEST['description']))?$_REQUEST['description']." <".$account.'>':'device'." <".$account.'>')
   );
   }

4. Look for the SIP Temporary Arrays, around line 2973 and add this value. We're not sure if it's required, but it works with it here so we left it.
   

   $tmparr['md5secret'] = array('value' => '', 'level' => 0);

5. Exit and save the file
6. Refresh FreePBX Extension and you should now see md5secret available as an option. This field also appears on the add new sip extension page as well.

You can download the modified functions.inc.php by using this link

Hi Guys,

With the help of one of my contractors (elisand.com), we've developed a little howto for FreePBX and Ubuntu. This should work on LTS and Regular versions of Ubuntu with minimal changes. We install all required software, asterisk 1.4 and FreePBX. It's been working for 2 of our clients without issue, so it was decided that it's time to post it up for the world.

In the interest of keeping it easy to read, I've decided to post it up as word format, and PDF documents. See below to download them.

• FreePBX Ubuntu Doc - Word 2003 Format
• FreePBX Ubuntu Doc - Word 2007 Format
• FreePBX Ubuntu Doc - PDF Format
• FreePBX Ubuntu Doc - Asterisk Init File for Ubuntu
• Download the Latest in PDF format from Free PBX Guide

We hope this helps you as much as it has us. It's taken the time down from around 5-7 hours for the initial install when we created this document, to more like 30-40 minutes including compile time. Swanky!

Please comment below if you have any additions, flames, praise!

Note, If you're looking for current information please see Free PBX Guide for the latest tutorial with updated information from what is located below.
Free PBX Guide

First we'll have to download the asterisk source from their site.

# cd /usr/src
# wget ftp://ftp1.digium.com/pub/asterisk/asterisk-1.4.5.tar.gz
# tar xzvf asterisk-1.4.5.tar.gz

Now we'll have to grab the latest SpanDSP from Soft-Switch. We'll also have to compile this and make sure no other SpanDSP versions are available on our server. The paths may change if you are using /usr/local so edit the following commands appropriately.

# cd /usr/lib
# rm -rf *spandsp*
# cd /usr/include
# rm -rf *spandsp*
# cd /usr/src
# wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre3.tgz
# tar xzvf spandsp-0.0.4pre3.tgz
# cd spandsp-0.0.4pre3
# ./configure && make && make install
# ldconfig -v

Now that SpanDSP has been compiled and installed, we can continue with the integration to Asterisk 1.4.

First we will need to copy the app_txfax.c and app_rxfax.c into the Asterisk Applications directory. This can be done like this:

# cd /usr/src
# cd asterisk-1.4.5
# cd apps
# wget http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/app_rxfax.c
# wget http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/app_txfax.c
# cd ..

Now we'll have to add the SpanDSP support to Asterisk in order to make use of these applications. I prefer to manually apply the patch file from Soft-Switch because it almost always fails. You can try either way you want.

Open up the patch file in a web browser so you can easily see what lines need to change, the patch file is located here

patch file: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/asterisk.patch

Now we'll need to manually apply these lines. Assuming you know something about patch files it should be pretty easy for you to modify the files, I'll briefly outline what needs to be done below.

First, Make sure you are in the Asterisk source directory.

# cd /usr/src/asterisk-1.4.5

Now, We'll have to modify the menuselect dependencies.

# nano build_tools/menuselect-deps.in

Search for RADIUS=@PBX_RADIUS@ and directly below this line put in the following new line

SPANDSP=@PBX_SPANDSP@

Exit and save the file.

Now you'll have to add a line to configure.ac

# nano configure.ac

Search for AST_EXT_LIB_SETUP([RADIUS], and directly below this line, put this in

AST_EXT_LIB_SETUP([SPANDSP], [spandsp Library], [spandsp])

Now, in the same file (configure.ac) we also have to add another line. Search for AST_EXT_LIB_CHECK([RADIUS], and directly below this line, add the following all on one line:

AST_EXT_LIB_CHECK([SPANDSP], [spandsp], [fax_init], [spandsp.h], [-ltiff])

Exit and save the file.

Now our last file to edit is the make options, which you can do like this

# nano makeopts.in

Search for the line like this RADIUS_LIB=@RADIUS_LIB@, add this directly below it

SPANDSP_INCLUDE=@SPANDSP_INCLUDE@
SPANDSP_LIB=@SPANDSP_LIB@

Exit and save the file.

Now we are almost ready to get the compiling going. But first, we have to copy the plc.h file to the Asterisk source so we won't get errors during compiling later on. You can do this as such:

# cd /usr/src/asterisk-1.4.5
# cd includes/asterisk
# cp plc.h plc.h.backup
# cp /usr/include/spandsp/plc.h .

Now because we've added new options to Asterisk, we'll have to bootstrap it

# cd /usr/src/asterisk-1.4.5
# ./bootstrap.sh

Now we re-run configure like the following, your options might be different

# ./configure --with-netsnmp --with-ssl --with-spandsp

Now we should go into the menuselect program to ensure that app_rxfax and app_txfax will be included when you compile Asterisk.

# make menuselect

Ensure that both applications are selected under the Applications menu. Exit the menuselect utility by pressing "x" to save your changes if required.

Now we can continue with the regularly scheduled compiling of the program.

# make
# make install

You should have a brand new copy of app_txfax.so and app_rxfax.so located in /usr/lib/asterisk/modules

Note that you might have to add /usr/local/lib or /usr/lib to your ld.so.conf in /etc in order to make this work, once you add the library path, re-run ldconfig, then try to start asterisk.

Have fun!

# cd asterisk-addons-1.4.2
# cd asterisk-ooh323c
# nano configure

and then I had to hit control-w, and search for libtool.m4, once I found the section that it checked for this problem of the software being differing versions I commented it all out. Basically the whole section starting with the beginning if statement of the libtool check all the way to the bottom fi statement.

Next you'll need to compile the Asterisk-Addons package by doing the following:

# cd ..
# make

Notice that you will be presented with yet another error that it can't find the library file to copy for you when you try to issue the make install after compiling. This is a simple fix for now, just do this.

# cd asterisk-addons-1.4.2
# cd asterisk-ooh323c
# cd .libs
# cp libchan_h323.1.0.1 libchan_h323.so.1.0.1

Now you can continue with the installation by running this

# cd ..
# make install

That's it, you should now have chanooh323 available for your asterisk installation. If you have any difficulty let me know in the comments and I'll try and help.

*NOTE* UPDATED VERSION OF THE SNMP HOWTO FOR ASTERISK 1.6 IS HERE IF YOU NEED IT *NOTE*

Download the slackware + snmp + asterisk howto in english here

Download the slackware + snmp + asterisk howto in portugese here

Leave us any comments if you find omissions or errors.

*NOTE* UPDATED VERSION OF THE SNMP HOWTO FOR ASTERISK 1.6 IS HERE IF YOU NEED IT *NOTE*

• 1 - Soekris NET4801
• 1 - PfSense Embedded Image
• 1 - CF Reader
• 1 - Serial Cable for the soekris
• 1 - CF Card 256 MB

The Steps I took:

1. Backup your old Soekris / Monowall configuration xml file
2. Buy a new CF card so we can keep the old configuration sane so it's plug and play "back to working" incase pfsense doesn't work
3. Plug your cf reader into one of your linux boxes, and do this with the image

   
   # zcat pfSense-1.0.1-Embedded.img.gz | dd of=/dev/sda bs=16k
   

   where /dev/sda is your cf card as seen by the OS. If you are not using a linux box, you can check out the howto's available at the following URL

4. Remove the CF card by unplugging the usb cable then the cf card.
5. Shutdown the Soekris, Remove the case and remove current CF Card, place aside
6. Place the new image card inside and then plug in your cables.
7. Connect serial cable to your linux box and set term to 9600 8N1. I use minicom.
8. Start up minicom on your linux box.
9. Plug in the Soekris
10. If it doesn't boot, you may have to set the bios to have the CF Card as primary.
    This note is thanks to the pfsense digest

    
    To do this enter the bios by pressing ctrl-p on boot and:
    
    set flash=primary
    
    reboot
    

11. Watch as it boots to make sure nothing is funky
12. Now you should see the configuration interface on your pfsense box, through minicom (the serial connection)
13. Setup the LAN and WAN interfaces
14. Connect to the LAN interface via a laptop, and configure the web interface
15. Remove laptop, plug into your switch, and you're good to go.

I like pfsense. It's got a lot more options than m0n0wall, and while it's a little slower on my soekris 4801, it's still a great addition to my network :)