Archive for the ‘PBX’ Category

Secret codes for nokia

Wednesday, June 9th, 2010

*#06#IMEI number (International Mobile Equipment Identity).
*#0000#Firmware version.
*#92702689#Life timer (W A R 0 A N T Y), The time your phone has spent in sending and receiving calls.
*#62209526#Wireless MAC Address.
*#2820#Bluetooth MAC address.
*#7370#Format phone.
*#7780#Factory Reset
*#7780#
*#7370#

List of Open Source PBX software

Tuesday, April 20th, 2010

Asterisk

FreeSwitch

Bayonne             http://www.gnutelephony.org/index.php/GNU_Bayonne

Yate                      http://yate.null.ro

Call outside through FXO card TDM400

Wednesday, March 24th, 2010

exten => _9X.,1,Dial(ZAP/1/${EXTEN:1})
for round- robin system  it is
exten =>  _9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})

Post installation steps for FreePBX 2.5.1 / AsteriskNow 1.5.0

Tuesday, March 2nd, 2010

Steps to Do

  • # Reset the passords of the default users
  • # Increase the memory limit to 100 M
  • # Change the Asterisk Manager Password
  • # Enter email_id for online update checks
  • #Download the latest FOP version
  • # edit “httpd.conf”

(more…)

Errors which AsteriskNOW displays after the installation

Tuesday, March 2nd, 2010

Cronmanager encountered 1 Errors
The following commands failed with the listed error
/var/lib/asterisk/bin/module_admin listonline (255)
Added 12 hours, 56 minutes ago
(cron_manager.EXECFAIL)

Could not reload FOP server

Could not reload the FOP operator panel server using the bounce_op.sh script. Configuration changes may not be reflected in the panel display.
Added 1 hour, 16 minutes ago
(freepbx.reload_fop)

(more…)

Understanding Digium Cards naming scheme

Wednesday, February 24th, 2010

Naming method looks like

  • “CCC X S O E/B “
  • CCC = is the interface type ,
  • TDM => PCI
  • AEX => PCI Express

It is read as

  • X => X is the no of port , it can be 4,8,24
  • S => S is the no of extensions or FXS lines
  • O => is the no of trunk lines
  • E/B can be either E which means hardwre echo cancellation module
  • B => Only Base module

My selected card is TDM-808EF
ie PCI card, with 8 port , 8 FXO , with echo cancellation

List of GPL based IP-Based PBX systems

Thursday, February 18th, 2010

I have come across three GPL based IP-Based PBX systems
They are

  • 1> Asterisk
  • 2> Call Weaver
  • 3> FreeSwith
  • 4> Yate

There are GUI for the above Telephone Systems they are

For Asterisk

  • FreePBX => http://www.freepbx.org/
  • Trixbox => http://trixbox.org/
  • Elastix    => http://www.elastix.org/
  • Askozia   => http://www.askozia.com/
  • Briker      => http://www.briker.org/
  • EasyAsterisk => http://www.easyasterisk.org/

For FreeSwitch

  • http://www.fusionpbx.com

For Yate
I have not come across any work , If any one know about it please do let me know

For CallWeaver

I have not come across any work , If any one know about it please do let me know

What I understood about SS7

Wednesday, February 17th, 2010

SS7 is an example of CCS (Common Channel Signalling ) Protocol

  • It is not a single protocol
  • It is a suite of Protocol
  • It is not an access protocol
  • It is a net work protocol
  • It is point to multipoint

ISUP(ISDN user part) and TUP (Telephone User Part)  are in the call control layer

There is MTP (Message transyer layer ) which has three layers:-MTP1 , MTP2 , MTP3

Here in MTP1 (OSI -Physical layer ) has E1,T1

My opinion
So we have E1 , T1 links available in MTP1 layer , is this enough for the PBX . ie If we take the E1 link from MTP1 to the foneBRIDGE , it should work.

Etisalat :- Primary Rate Access (PRI)

Wednesday, February 17th, 2010

A 2Mbps pipe from the exchange to your business provides thirty channels, each at speeds of 64Kbps. They can be used for voice and data dial-up services.
With ISDN PRI, almost all modern voice and data communication systems from PBXs, mainframe and distributed systems through to your company LAN/WAN can be connected to your ISDN service.
The use of routers, multiplexers, ISDN controllers and video/audio conferencing units is also supported.

Additionally, Digital Telephone Switches can use our Direct Inward Dialling (DID) facility over PRI. DID offers a single interface for up to 30 concurrent calls, as well as for up to two hundred free DID extensions.