National Repository of Grey Literature 85 records found  1 - 10nextend  jump to record: Search took 0.00 seconds. 
Implementing of GSM modem in PBX Asterisk
Benýšek, Jiří ; Krajsa, Ondřej (referee) ; Šilhavý, Pavel (advisor)
Short Message Service (shortly SMS) is the most widely used type of communication systems. The main advantages are that allow a fast exchange of messages between devices, a very good availability through GSM and a reasonable price. Nowadays the SMS service support has expanded to include other technologies such as a service of the information navigation and the remote connection. The master‘s thesis concentrates on the Short Message Service, deals with basic principles and statements using by this service. The topic of the thesis is software PBX Asterisk and its possibility of SMS implementation, especially verification of SMS processing goes through the PSTN. After the basic introduction the master‘s work deals with the installation and configuration of the server. The main focus is on an installation of the operating system with an additional pack including necessary libraries and modules for a correct working of the server. The following section is paying attention to the Asterisk server configuration, especially a hardware card installation which is necessary for a connection with analog telephones, done by Bluetooth connections, set up user’s profiles of the SIP protocol and create a dial plan. This is followed by a verification of SMS option of the implementation and communication with GSM modem which is used as a gate for an exchange SMS between PSTN and GSM network. The last chapter of this master‘s thesis comes with the aimed results.
Billing system and call monitoring for PBX Asterisk
Depiak, Petr ; Šilhavý, Pavel (referee) ; Krajsa, Ondřej (advisor)
This master's thesis is focused on developement of billing system with the options of monitoring individual calls for software exchange Asterisk. Billing of calls is adaptible with the help of group of individual rules, consisting of tariff impulses, numerical prefix, with help of outgoing trunk and cost of the billed unit. The first part of this work is focused on instalation, configuration and preparation of individual components of the billing system. In this work is explained the architecture of the billing system and highlighted the purpose of work of the model database. Next we focused on the purpose and the principal system invidual function of the system including solution. At last there is a simple manual to operate the system with the help of created web interface.
Project of VoIP exchange for commercial usage
František, Jiří ; Hanák, Pavel (referee) ; Janovič, Filip (advisor)
The aim of this thesis is create VOIP central based on Asterisk with billing system A2Billing. First of all is describe process of instalation programing instrument and provisioning pbx for local and outgoing calls. Short describe to billing of calls and praktical show of completely setting of system are follows. Administrator and user web based interface is described to. At the end of thesis is shown different technology of VOIP phones which was use in thesis. Setting of this phones is briefly described.
Communication systems based on IP telephony
Zimek, Josef ; Kapoun, Vladimír (referee) ; Herman, Ivo (advisor)
My master’s thesis is focused on designing and creating communication network, which provides communication between two independent networks through encrypted tunnel. My solution is based on routers formed by older personal computers with FreeBSD like a operating system. Between routers is created static encrypted tunnel by using IPSec protocol. Voice services provides packet oriented exchange Asterisk with support of signaling protocol SIP. This solution can be used eg. for connecting remote branch to headquarters of company and then can branch utilize shrared resources. To headquarters can connect also remote workers from their home. In this case are used SSL certificates to authentication of user. This scenario is very required today.
Securing IP PBX against attacks and resistance testing
Kakvic, Martin ; Šedý, Jakub (referee) ; Šilhavý, Pavel (advisor)
This diploma thesis focuses on attacks on PBX Asterisk, FreeSWITCH and Yate in LTS versions. In this work was carried out two types of attacks, including an attack DoS and the attack Teardown. These attacks were carried out using two different protocols, SIP and IAX. During the denial of service attack was monitored CPU usage and detected if its possible to establish call and whether if call can be processed. The Security of PBX was build on two levels. As a first level of security there was used linux based firewall netfilter. The second level of security was ensured with protocols TLS and SRTP.
Asterisk VoIP private branch exchange and its distributions
Melichar, Ondřej ; Komosný, Dan (referee) ; Papež, Nikola (advisor)
This master’s thesis delves into the possibilities of the open-source Private Branch Exchange Asterisk, elaborates on its features and compares it with several other distros. The term SIP stack is explained here with the mention of two of its representatives. Further in the thesis, the security risks of the VoIP technology are explained, and specific attacks are described and then realized. As a part of the testing process, the possibilities of a custom module and its following implementation are explored, as well as the portability between the individual distros and its proper functioning.
Modular web interface for Asterisk PBX
Moučka, Martin ; Rášo, Ondřej (referee) ; Krajsa, Ondřej (advisor)
This bachelor thesis is dealing with various ways of configuring Asterisk PBX and creating a modular web interface. This interface’s goal is to simplify the configuration of PBX and allow expanding its functions by adding new modules. Apart from a simple configuration, interface contains a self care zone where user can check his call history and keep his own phone book. As part of this work, there are two modules especially useful in environment of technical support and call centers. Those modules provide an opportunity to create interactive voice response menu, calling automat and queues management. Interface depends on Asterisk in version 13 which is the newest version with long term support. The application is secured by user account management with role assigning. The role can be modified by permitting only specified actions.
Performance limits, reliability and security of open source PBX
Bednár, Jakub ; Šedý, Jakub (referee) ; Šilhavý, Pavel (advisor)
The aim of this thesis is to install and to configure three Open source PBXes Asterisk, Freeswitch and YATE. Furthermore, the aim is to realize the performance test and stability tests on three different HW configurations with the tester Spirent Abacus 5000. The scripts in bash were created to monitor PBX performance. Another part of the study is to analyze and to compare PBX security and to compare the Open Source PBX with a proprietary PBX Alcatel-Lucent OXE.
SIP implementations in Asterisk open source PBX
Bednář, Vít ; Krajsa, Ondřej (referee) ; Šilhavý, Pavel (advisor)
The thesis compares native SIP stack with PJSIP stack in the open source telephone private branch exchange (PBX) Asterisk. First, there are described both SIP protocol and Asterisk application. Subsequently, the architecture, new function support and the stacks setting possibilities are explored. For different exchange scenarios several commented configuration files are presented. The stacks are tested using Spirent TestCenter C1 software thereafter. In conclusion, selected properties are assessed and new PJSIP stack benefits are summarized. In addition, the laboratory assignment is attached.
Deployment of SIP Server at the FIT for IP Telephony
Hýbner, Lukáš ; Ráb, Jaroslav (referee) ; Matoušek, Petr (advisor)
Master's Thesis is engaged on possibilities connect SIP server to telephone network on FIT. The main reason is that employees can call to university, when they are out of faculty. For resolution we will use SIP server Asterisk, which will be serving as authorization server for users. Next Asterisk will ensure transmission numbers to SIP address with ENUM. In the practical part we will verify the functionality.

National Repository of Grey Literature : 85 records found   1 - 10nextend  jump to record:
Interested in being notified about new results for this query?
Subscribe to the RSS feed.