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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.
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Implementation of Asterisk VoIP PBX
Schön, Martin ; Münster, Petr (referee) ; Szőcs, Juraj (advisor)
The thesis is divided into two parts, theoretical and practical. In theory, it analyzed the signaling protocols, the RTP protocol for data transmission, VoIP Asterisk PBX and security options. The practical part describes the installation of the Asterisk PBX, conducted demonstration call forwarding. PBX was associated with an LDAP directory server to manage users and ejabberd for text messaging protocol XMPP. It was configured translation SIP-Skype using the SiSky Gateway and with dialplan SIP-H.323 translation. The study was to develop Dialplan with IVR. It was measured bandwidth of the audio/video codecs in an encrypted/unencrypted connections.
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Grammar Based Automatic Speech Recognizer
Škorvaga, Vojtěch ; Karafiát, Martin (referee) ; Schwarz, Petr (advisor)
This work describes a development of system for network compilation for speech recognition based on Speech Recognition Grammar Specification (SRGS) grammar defined by W3C consortium. Together with the new module, the recognizer was integrated to the FreeSwitch software phone switch using a combination of MRCPv2/SIP/RTP networks protokols and tested.
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Stress Testing of Internet Telephony
Šípek, Martin ; Ilgner, Petr (referee) ; Číka, Petr (advisor)
The thesis deals with the development of a module for JMeter that adds SIP stress testing functionality that was previously missing in the program. The theoretical part of the thesis deals with Internet telephony and codecs used in telephony. The SIP protocol is discussed in detail and the end of this part is devoted to stress testing. The practical part focuses mainly on the development of the proposed module, where the code of the module is described in detail, but it also deals with the installation and configuration of the Asterisk PBX on which the proposed module is tested. The results of this testing are analyzed at the very end of this thesis.
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Stress Testing of Internet Telephony
Šípek, Martin ; Ilgner, Petr (referee) ; Číka, Petr (advisor)
The thesis deals with the development of a module for JMeter that adds SIP stress testing functionality that was previously missing in the program. The theoretical part of the thesis deals with Internet telephony and codecs used in telephony. The SIP protocol is discussed in detail and the end of this part is devoted to stress testing. The practical part focuses mainly on the development of the proposed module, where the code of the module is described in detail, but it also deals with the installation and configuration of the Asterisk PBX on which the proposed module is tested. The results of this testing are analyzed at the very end of this thesis.
|
|
| |
|
|
Grammar Based Automatic Speech Recognizer
Škorvaga, Vojtěch ; Karafiát, Martin (referee) ; Schwarz, Petr (advisor)
This work describes a development of system for network compilation for speech recognition based on Speech Recognition Grammar Specification (SRGS) grammar defined by W3C consortium. Together with the new module, the recognizer was integrated to the FreeSwitch software phone switch using a combination of MRCPv2/SIP/RTP networks protokols and tested.
|
|
|
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.
|
|
|
Implementation of Asterisk VoIP PBX
Schön, Martin ; Münster, Petr (referee) ; Szőcs, Juraj (advisor)
The thesis is divided into two parts, theoretical and practical. In theory, it analyzed the signaling protocols, the RTP protocol for data transmission, VoIP Asterisk PBX and security options. The practical part describes the installation of the Asterisk PBX, conducted demonstration call forwarding. PBX was associated with an LDAP directory server to manage users and ejabberd for text messaging protocol XMPP. It was configured translation SIP-Skype using the SiSky Gateway and with dialplan SIP-H.323 translation. The study was to develop Dialplan with IVR. It was measured bandwidth of the audio/video codecs in an encrypted/unencrypted connections.
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