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A-O-A My name is Abdullah Sabri. I recently completed my BE(IT) From UET Texila. My Final year project"Implementation of VOIP network over NGN". The main... here is my CV ..

sabri — Thu, 08 Dec 2011 15:15:32 +0200

M.ABDULLAH SABRI

OBJECTIVE
To contribute in the growth and vision of a dynamic corporation while utilizing my potential for achieving excellence, enhancing my technical and managerial skills.

EDUCATION ▼
1. BEIT [2007-2011] (2.54 * CPA 2=60%)
UET Taxila, Pakistan.

Major Courses Studied:
COURSES
Mobile And Satellite Communication
Circuit Analysis
Software Engineering
Operating System
Computer Networks
Advance Computer Network
Object Oriented Programming
Data Structure   Computer Architecture

2.   DAE (INFORMATION TECHNOLOGY [2006] 1ST DIVISION (68.75%)

KRL Institute of Computer
Sciences and Information Technology,
Kahuta., Pakistan.

3. SSC. MATRICULATION [2003] 2ND DIVISION (58.11%)

FINAL YEAR PROJECT

IMPLEMENTATION OF VOIP NETWORK OVER NGN (NEXT GERNATION NETWORK)
Linux /SQL Data Base
Description:
The Project is primarily concerned design of system and performs fully VoIP network services. The basic objective of this Open source project is to implement a VoIP network using NGN ideology. The network will be able to handle both SIP/IP and non SIP/IP communication. This network will be integrated with PSTN using some gateway. An application server will also be integrated for different type of services. The Billing server (a2billing) will be used for AAA (Authentication, Authorization & Accounting). This network will meet the demands of NGN and 3 G Networks. Using SER (SIP Express Router) And Asterisk we can make thousands of concurrent calls per second on very much low cost.SER uses standardized protocols, this allows end-users and system administrators to have more control and even develop communication services that have been traditionally unchangeable and limited. Our Proposed project will provide the advanced communication services like voice calls; video calls voice .mail services, phone to phone calls, pc to phone calls, phone to pc calls with billing solution. This task has delivered the fully functional VoIP solution having many integrated features like,
•   Call waiting
•   Call forwarding
•   Music on Hold
•   Audio calls
•   video calls
•   Interactive Voice Response (IVR) and have fully billing solution.

SEMESTER PROJECTS

•   Library Management System in MS Access.
•   Video Shop in C Language.
•   Library Management System in Oracle data bases.
•   Online Merit List Generator Website using C#/ASP.Net
•   Online Book Shop Using PHP

SKILLS

Languages C# , C ,C++ , Java , Oracle

Software Microsoft windows XP,Window 7,Linux (Centos 5.4)

Microsoft Office ,Power Point , Excel , Excess , Fireworks, Asterisk Configuration ,Open SER configuration, Trix Box ,Asterisk Billing (A2billing)

Web HTML ,CSS ,Java Script ,Php , MySql , Photoshop, Dream weaver
And currently working on YII Php Frame Work (Making Report for my company and will Present to all employees).
Other Effective presentation and communication skills, Group work.

PERSONAL INFORMATION

Father Name : Abdul Hakim
CNIC # : 3730231094387
D-O-B : 05-11-1987
Martial Status : Single
Religion : Islam
Nationality : Pakistani

References will be furnished as and when required.

Posted to: A-O-A My name is Abdullah Sabri. I recently completed my BE(IT) From UET Texila. My Final year project"Implementation of VOIP network over NGN". The main... here is my CV .. Show Thread | Post Reply

Re: Asterisk Pakistan Website

unluckyprince — Thu, 16 Jun 2011 09:50:40 +0300

It seems good to be a part of such a great community.

Posted to: Asterisk Pakistan Website Show Thread | Post Reply

Re: Asterisk Located in Toronto ,Pakistani user cant register

shah — Sun, 05 Jun 2011 07:13:38 +0300

5060 port by default blocked in pakistan

Change port ...................... 5060

Posted to: Asterisk Located in Toronto ,Pakistani user cant register Show Thread | Post Reply

Asterisk Traning!...

Arshad — Tue, 19 Apr 2011 15:46:33 +0300

I am new in this and i am ready to learn Asterisk* with AGI Programming kindly update me when and where i can learn Asterisk*.... Awaiting your kind response, Thanks. 0322 235 3643........

Posted to: Asterisk Traning!... Show Thread | Post Reply

asterisk configuration

hashaam — Fri, 25 Feb 2011 10:07:31 +0200

hello.....i new to asterisk......i have installed asterisk in my local machine in linux.....i have also installed a virtual computer...virtualbox in this system......i than installed 2 twinkle phones....one in my local machine and other in the virtual machine......and i have registered them both with asterisk........there exists a problem which is that i am able to call from virtual machine to local machine and they both connect but when i try to call from host machine to virtual machine it does not work......
following are my sip.config and extension.config files.......here billy is in host machine and blaine is in virtual machine.....

sip.config:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[blaine]
type=friend
context=phones
host=dynamic

[billy]
type=friend
context=phones
host=dynamic
...........................................

extension.config:

[globals]

[general]
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]

[internal]
exten => s,1,Verbose(1|Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/billy,30)
exten => 1000,n,Hangup()

exten => 1002,1,Verbose(1|Extension 1002)
exten => 1002,n,Dial(SIP/blaine,30)
exten => 1002,n,Hangup()

[phones]
include => internal
include => default

Posted to: asterisk configuration Show Thread | Post Reply

Asterisk Training

Shomail — Fri, 07 Jan 2011 09:40:05 +0200

Crystal consulting is pleased to announce that it would be holding Asterisk workshops in Karachi. This hands-on training session is designed for people with or without prior Linux or Asterisk experience, will get you started quickly and efficiently with the fastest growing telephony platform.

What: Asterisk Training by Crystal Consulting

Where: Karachi, Pakistan.

Duration: 30 to 35 hours

For details please visit our website at:

www.crystalconsulting.pk/asterisk-training-2/

Alternatively for more information you can

training@crystalconsulting.pk

Or call us at :

0300-3497670

Posted to: Asterisk Training Show Thread | Post Reply

Re: Urdu Numbers in Asterisk

zeeshan — Fri, 22 Oct 2010 15:11:39 +0300

Hi i am Total new this field i need help i am ready to Pay money if you are leaving in Karachi just tell me number i will call you please response fast

Thanks

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

Re: Urdu Numbers in Asterisk

zeeshan — Fri, 22 Oct 2010 15:11:35 +0300

Hi i am Total new this field i need help i am ready to Pay money if you are leaving in Karachi just tell me number i will call you please response fast

Thanks

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

TDM800 for sale

pakion — Mon, 11 Oct 2010 21:05:09 +0300

Hi and Sallam to all.

Dear i have one Asterisk 8 Ports TDM800 Module for sale any one interested do call me.

Pakion
92-333-2131218

Posted to: TDM800 for sale Show Thread | Post Reply

Digium Wildcard TE405P for sale

fshals — Sat, 18 Sep 2010 14:31:38 +0300

Digium Wildcard TE405P

The TE405P is the next generation of Digium hardware that improves performance and scalability through bus mastering architecture. Use the telephony card selector to identify a card that fits your requirements.

The TE405P supports both E1, T1, J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1, T1, and J1 equipment and allows inexpensive T1 channel banks to connect with E1 circuits. Because the TE405P improves I/O speed by up to 10 times, the result is reduced CPU usage and increased card density per server.

Digium has designed the TE405P to be fully compatible with existing software applications and it is fully integrated with the Asterisk Open Source PBX/IVR platform. Also, the open source driver supports an API interface for custom application development. With the combination of Digium Hardware and Asterisk software, numerous combinations of telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications.

The TE405P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features.

The TE405P is for use only with a 5.0 volt PCI slot. The TE410P is for use only with a 3.3 volt PCI slot - typically available on newer server motherboards and in 64-bit PCI bus architectures.

AOA ,
Dear members i have this card now i want to sell it it is used card with extension slot i am using it from last year now i want to sell it if any one intrested then mail me on faisalshals@gmail.com , faisalshals@yahoo.com & hotmail.com same
Thanks regards
Syed Faisal Saleem

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Re: Asterisk Located in Toronto ,Pakistani user cant register

astpanj — Tue, 14 Sep 2010 11:27:04 +0300

What method using to connect to your asterisk server. SIP or what?

Posted to: Asterisk Located in Toronto ,Pakistani user cant register Show Thread | Post Reply

Re: rejected because extension not found

astpanj — Tue, 14 Sep 2010 11:14:53 +0300

Sip.conf
[PHONE1]
type=friend
context=default
regexten=1234
callerid="User1" <1234>
host=dynamic
disallow=all
allow=ulaw

Context default does not exist in your extensions.conf so change it to context=local. If it does not work you may come back here.

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