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By admin, December 23, 2010 8:11 PM

Definition:
Magicjack is really a simple device that uses the standard SIP protocol to make inbound and outbound calls.
Thus if we know the details of this we can put it an Asterisk server or sipphone, eg iphone etc.

MagicJack uses standard SIP, so once you determine your SIP information (username/password/etc), you can use your MagicJack service with any standard VoIP device.
The easiest way to uncover your MagicJack SIP information is to use Fiddler.
Since all MagicJack passwords are very simple, you could also very easily brute force them using cain, although this technique is a bit more involved than using fiddler.

MagicJack Voip settings you will need to put into your device:
Voip Settings:

Username:
Password:
sip port: 5070
proxy: proxy1.yourcityname.talk4free.com:5070
register: yes
make call without reg: yes
answer call without reg: no
register expires: 3600
display name: your phone number

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.

OTHER METHOD

1. download pmdump from http://magicjackhacks.com/downloads/pmdump.exe
2. follow tutorial on how to create a dump file on this page http://magicjackhacks.com/ using pmdump.
3. download “hex workshop” its free. search on google for it. or use any other free hex editor.
4. open the dump file in the hex editor.
5. search for “ProxyUserName”
6. search for “ProxyUserPassword”
7. search for “SIPProxyURI”

the user name will be your phone number with a leading E and ending 01
password is 20 digits
sipproxy is an IP address

First download pmdump and hexeditor

http://magicjackhacks.com <–pmdump
http://www.hexworkshop.com/ cmd(enter) and then connect your magicjack.

The rest with hexeditor is simple, just drag the created file and search the two strings, it worked for me at the 4th try.
My last post is incomplete. Here is what i missed.
It is just the WHEN you are dumping the memory, it should be done inmediatly (5 seconds) after connecting magicjack, and executing the application. It worked for me at the 3th try.
I suggest to keep open CMD while magicjack is loading that way you will have the chance to do it quick.
pmdump -list (after magicjack has begin to run)
pmdump PID anyfilename (3 seconds after)
Then it’ll work.

FILES YOU MAY NEED:

http://www.megaupload.com/?d=SJ6V5SPF

http://rapidshare.com/#!download|221tl|181798719|MagicJack_Utilities_v1.6.zip|3513

http://rapidshare.com/#!download|492tl|196998576|MJInfo.exe|456

REF:

http://www.bauer-power.net/2010/05/how-to-hack-your-magicjack-to-make.html

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By admin, December 23, 2010 7:23 PM

I have searched numerous forums and cannot find any decente
sipphone settings for the Siemens Gigaset Phone for Sipphone / Gizmo5

The setup:

Personal Provider Data
* Authentication Name:  1747xxxxx what ever your gizmo5 number is
* Authentication password: password what ever yours is
* Username:     1747xxxxxx what ever your gizmo5 number is
* Display name:  1747xxxxxx what ever your gizmo5 number is

General Provider Data
* Domain:   proxy01.sipphone.com
* Proxy server address: proxy01.sipphone.com
* Proxy server port:     5060
* Registration Server: proxy01.sipphone.com
* Registrar server port: 5060
* Registration refresh time: 180

Network
* STUN enabled: YES
* STUN server: stun01.sipphone.com
* STUN port: 3478
* STUN refresh time: 240
* NAT refresh time: 20
* Outbound proxy mode: Never (can anyone confirm?)
* Outbound proxy: EMPTY
* Outbound proxy port: 5060

If you are having trouble with missing one part of the audio on a call ie cant hear the other person speaking but can hear and speak
fine otherwise then you should enable STUN.     (can anyone confirm?)
Gizmo5 does not provide an outbound proxy, so you should leave that field blank.

The screen shot:

siemens gigaset sipphone gizmo5 setup

The codecs:

Codecs:
GSM — fixed bit rate, not loss tolerant, narrow band (8khz sampling rate).
iSAC — variable bit rate, loss tolerant, narrow and wideband (8 to 16khz). Varies based on Bandwidth, packet loss, delay
iLBC — variable bit rate, loss tolerant, narrow
PCMA — fixed bit rate (8kHz sampling rate)
PCMU — fixed bit rate (8kHz sampling rate, high band width)
IPCMWB — 16 kHz sampling rate
EG711 (enhanced g711) — fixed bit rate, loss tolerant, narrowband
iPCM — fixed bit rate, loss tolerant, wide band.

I would recommend that you enable
ulaw u711
alaw a711
GSM
iLBC
g729

and disable
g722 (wideband)

I believe but have no confirmation that it does g729 in passthu mode only. (can anyone confirm?)

DTMF

In-band
Incoming stream delivers DTMF signals in-audio independently of codecs

Out-of-band
Incoming stream delivers DTMF signals out-of-audio using either SIP-INFO or RFC-2833 mechanism,
independently of codecs – in this case the DTMF signals are sent separately from the actual audio stream.

SIP-INFO is not recommended for DTMF delivery, since it cannot deliver strokes synchronously with the audio stream, introducing timing artifacts (mainly because it’s delivered using SIP, which is not a real-time mechanism for delivering media). It is very common for public services to NOT support SIP INFO, and it seems unlikely that such services will improve support for this delivery mechanism.

We want to use rfc2833

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By admin, September 4, 2010 6:40 PM

HERE ARE THE AUSTRALIAN SETTINGS FOR THE LINKSYS 3102

AND ALSO THE FAX SETTINGS TO MAKE FAXING WORK ON VOIP

AUSTRALIAN SETTINGS

Dial tone: 400@-19,425@-19,450@-19;10(*/0/1+2+3)

Busy Tone: 425@-19;10(.375/.375/1)

Reorder Tone: 425@-19,425@-29;60(.375/.375/1,.375/.375/2)

Ring Back Tone: 400@-19,425@-19,450@-19;*(.4/.2/1+2+3,.4/2/1+2+3)

MWI Dial Tone: 400@-19,425@-19,450@-19;2(.1/.1/1+2);10(*/0/1+2)

Ring1 Cadence: 60(.4/.2,.4/2)

 

FXS Port Impedence: 220+820||115nF
DTMF Playback Length: .25
Time Zone: GMT+10:00

PSTN

Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2)
 FXO Port Impedance: 220+820||120nF
 PSTN to SPA Gain: 3
 On-Hook Speed: 26ms (Australia)

DIAL PLAN
JUST A NOTE NOT DIAL PLAN NO WORKIE

([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#,:>xx.<:@gw0>|<#,:>*xx<:@gw0>)

FAX 3102 SETTINGS

a) Network Jitter Level – Very High (was medium)
b) Jitter Buffer Adjustment – Disable (was up down)

c) “Call Waiting Serv” : Set this to No
d) “Three Way Call Serv” : Set this to No as well.

a) Set the PreferredCodec to G711u
b) Set “Silence Suppression Enable” to No
c) Make sure that the option FAX Enable T38 is set to Yes. (Default setting is Yes)
d) “Echo Canc Enable” should be set to No
e) Set the option “Fax Passthru Method” to REINVITES.

more infro

https://docs.google.com/View?docid=dmqx96v_3qk8phn

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By admin, August 14, 2010 7:57 PM

[global]
log file = /var/log/samba/log.%m
load printers = no
smb passwd file = /etc/samba/smbpasswd
browseable = no
server string = trixbox PBX
path = /
workgroup = workgroup
os level = 2000
username map = /etc/samba/smbusers
preferred master = no
max log size = 50

[share]
browseable = yes
writeable = yes
write list = root
comment = server
valid users = root
path = /
public = yes

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By admin, August 14, 2010 7:46 PM

system-config-network

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By admin, August 14, 2010 7:34 PM

make a DB in mysql
login to mysql
mysql -u root -p
then
CREATE DATABASE `openfire`;
CREATE USER ‘openfire’@'localhost’ IDENTIFIED BY ‘password’;
GRANT USAGE ON *.* TO ‘openfire’@'localhost’ IDENTIFIED BY ‘password’ WITH MAX_QUERIES_PER_HOUR 0 MAX_CONNECTIONS_PER_HOUR 0 MAX_UPDATES_PER_HOUR 0 MAX_USER_CONNECTIONS 0;
GRANT SELECT, INSERT, UPDATE, DELETE, CREATE, DROP, INDEX, ALTER, CREATE TEMPORARY TABLES ON `openfire`.* TO ‘openfire’@'localhost’;
FLUSH PRIVILEGES;
quit

Comments (2)

By admin, August 12, 2010 7:35 PM

find your CPU architecture
cat /proc/cpuinfo

Goto

http://asterisk.hosting.lv/

DL the right one
# choose codec binary appropriate for your Asterisk version and CPU type, use x86_64 for 64-bit mode, scroll to the end of the list for FreeBSD binaries
# delete old codec_g729/723*.so files (if any) from /usr/lib/asterisk/modules directory
# copy new codec_g729/723*.so files into /usr/lib/asterisk/modules directory
# restart Asterisk

Then connect to the Asterisk console with:
asterisk -r
and type (adjust accordingly):
load codec_g729-ast14-gcc4-glibc-core2.so

You should then see something like:
Loaded /usr/lib/asterisk/modules/codec_g729-ast14-gcc4-glibc-core2.so => (G729/PCM16 (signed linear) Codec Translator, based on IPP)
== Registered translator ‘g729tolin’ from format g729 to slin, cost 1
== Registered translator ‘lintog729′ from format slin to g729, cost 6

Exit from the console with:
quit

Then use nano to edit /etc/asterisk/sip.conf and just below:
allow=ulaw
add:
allow=gsm
allow=g729

# check the codec is loaded with ‘core show translation recalc 10′ on Asterisk console (‘show translation’ in Asterisk 1.2)
# G.723.1 send rate is configured in Asterisk codecs.conf file (Linux Asterisk 1.2, 1.4, 1.6, TRUNK and Callweaver, FreeBSD 7.x Asterisk 1.4 binaries only):

other cool linux commands

cat /proc/cpuinfo
cat /proc/meminfo
dmesg
lspci

Comments (1)

By admin, February 15, 2010 10:11 PM

sip_custom.conf

language=au
videosupport=yes
allow=g729
allow=g723
allow=h261
allow=h263
allow=h263p
useragent = PAP2T

Comments (2)

By admin, November 28, 2009 12:10 AM

How to Secure Trixbox with fail2ban

For usiness purposes Fail2Ban is an excellent solution for protecting at the SIP application layer.

It is worth reading how to set up Fail2Ban for Trixbox

http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk

wget http://downloads.sourceforge.net/project/fail2ban/fail2ban-stable/fail2ban-0.8.4/fail2ban-0.8.4.tar.bz2 -jxf fail2ban*

How to Secure Trixbox by changing passwords

1)FOP, Default password is again : passw0rd

2) mysql default passwords is “passw0rd”

3) FreePBX web interface has the default user: maint with password ” password”

4) Trixbox main web Interface, access is wide open, meaning that anyone who knows your PBX IP address or sub domain, can access it. a) VoIP installers

Changing your default CentOS Password

passwd

You will be asked to enter your old password and to type in your new password twice.

Changing your default FreePBX Password

The default login and password for a newly installed FreePBX (formerly known as AMP) is:

Username: maint
Password: password

To change the default password at the CentOS command prompt type the following command.

passwd-maint

Changing your default FOP Password

edit /etc/amportal.conf

find FOPPASSWORD=passw0rd and change it for something reasonable

amportal restart

Changing your default MeetMe Password

passwd-meetme

It will ask you for your new password twice.

Changing your default System Mail Password

passwd admin

Changing your default MySQL Password

Edit /etc/amportal.conf and change AMPDBUSER=asteriskuser and AMPDBPASS=yourpassword.
Careful, the values at the top of that file are actually commented out (which is idiotic), the real values are at the very bottom of the file.

Edit /etc/asterisk/cdr_mysql.conf and change USER= asteriskuser and PASSWORD=yourpassword.

Edit /etc/astersik/ cbmysql.conf and change DBUSER= asteriskuser and DBPASS=yourpassword.

Lastly, login to the commandline on your TrixBox terminal as root and execute these commands:
amportal stop
mysqladmin -u asteriskuser -p password yourpassword
[then enter your current password for root to confirm the change]
service mysqld restart
amportal start

mysql -u root -p
passw0rd

SET PASSWORD FOR asteriskuser@localhost=PASSWORD(‘newpass‘);

amp111

SUMMARY

DO NOT CHANGE DEFUALT PASSWORDS APART FROM MAINT AND FOP

IT IS JUST TOO HARD TO GET THE WHOLE TRIXBOX WORKING AGAIN!!!!!

PEOPLE WHO MADE TRIXBOX BIT SLOPPY IN THE PASSWORD SECURITY FRONT
WOULD NOT BE THAT HARD TO WRITE A SCRIPT THAT CHANGED ALL THE PASSWORDS
AND RELEVANT FILES