This document provides information about installing and configuring Asterisk PBX software on a Linux system. It discusses installing necessary hardware like an X100P FXO card and TDM400P FXS card. It describes configuring the zaptel and zapata files to define the channels for each card. Finally, it briefly discusses setting up SIP by editing the sip.conf and extensions.conf files to register Asterisk with a SIP proxy.
The document discusses IPv6 networking. It provides an overview of IPv6 addressing and packet structure, describing IPv6 addresses, extension headers, and fragmentation. It also examines potential weaknesses in OS IPv6 protocol stacks and techniques for bypassing firewalls and intrusion detection systems when using IPv6.
The document discusses various attacks that are possible against the AoE (ATA over Ethernet) storage protocol due to its lack of authentication and security features. Some key attacks mentioned include replay attacks, unauthenticated disk access by reading and writing directly to disks, creating an AoE proxy to reroute traffic, and denial of service attacks. The document warns that AoE deployments could be vulnerable if not properly segmented from untrusted networks.
This document provides information about RSTS/E, an early operating system commonly used on PDP computers. It describes how to identify an RSTS/E system upon connecting, explains the login process using a Project-Programmer Number and password, and lists some basic system commands like HELP, DIRECTORY, and BYE once logged in. The document aims to describe the basics of interacting with an RSTS/E system for beginners.
- Hidden or administrative shares are share names with a dollar sign ($) appended to their names. They do not display in the network browse list.
- The seven layers in the Open System Interconnection model are: Application, Presentation, Session, Transport, Network, Data link, and Physical.
- The address resolution protocol (ARP) is used to associate the 32 bit IP address with the 48 bit physical address, while the reverse address resolution protocol (RARP) allows a host to discover its Internet address when it knows only its physical address.
This document provides instructions for installing and configuring TinyOS on Ubuntu Linux. It outlines downloading TinyOS from its website, adding the TinyOS repositories, configuring environment variables, and compiling and pushing applications to sensor motes. Shell scripts are provided to simplify connecting motes, compiling modules, and pushing compiled code. Running the Java listener and GUI are also explained for receiving and viewing sensor data from motes over a TCP/IP network connection.
In this workshop we will make a brief introduction to the basics of networking: IP addresses, MAC addresses, DNS, DHCP. Concepts as a router, gateway and firewall are explained. Then we will see in practice how to share files on a local network (NFS, Samba), establish a FTP connection, or log on to another (Linux) machine remotely (SSH, VNC, RDP). Finally, we review some useful networking tools like ping, netstat, lookup, port scan, traceroute, whois.
This document provides information on networking topics in Linux including:
- How to connect to Linux systems using SSH and things that can be done from the Linux command line interface
- IP addressing and subnet masking
- Setting up networks and creating permanent network configuration files
- Network troubleshooting tools like traceroute, nmap, netstat
- Reasons why network software may not work like firewalls blocking ports or network speed issues
- An overview of VPNs versus proxy servers and how each works
The document discusses IP Telephony and its infrastructure. It begins with an acknowledgement and background on telecommunications in Bangladesh. It then discusses combining wireless networks with VoIP. The key components of a VoIP infrastructure include softswitches, signaling gateways, and media gateways. Protocols like SIP are used to carry signaling information over IP networks.
The document discusses IPv6 networking. It provides an overview of IPv6 addressing and packet structure, describing IPv6 addresses, extension headers, and fragmentation. It also examines potential weaknesses in OS IPv6 protocol stacks and techniques for bypassing firewalls and intrusion detection systems when using IPv6.
The document discusses various attacks that are possible against the AoE (ATA over Ethernet) storage protocol due to its lack of authentication and security features. Some key attacks mentioned include replay attacks, unauthenticated disk access by reading and writing directly to disks, creating an AoE proxy to reroute traffic, and denial of service attacks. The document warns that AoE deployments could be vulnerable if not properly segmented from untrusted networks.
This document provides information about RSTS/E, an early operating system commonly used on PDP computers. It describes how to identify an RSTS/E system upon connecting, explains the login process using a Project-Programmer Number and password, and lists some basic system commands like HELP, DIRECTORY, and BYE once logged in. The document aims to describe the basics of interacting with an RSTS/E system for beginners.
- Hidden or administrative shares are share names with a dollar sign ($) appended to their names. They do not display in the network browse list.
- The seven layers in the Open System Interconnection model are: Application, Presentation, Session, Transport, Network, Data link, and Physical.
- The address resolution protocol (ARP) is used to associate the 32 bit IP address with the 48 bit physical address, while the reverse address resolution protocol (RARP) allows a host to discover its Internet address when it knows only its physical address.
This document provides instructions for installing and configuring TinyOS on Ubuntu Linux. It outlines downloading TinyOS from its website, adding the TinyOS repositories, configuring environment variables, and compiling and pushing applications to sensor motes. Shell scripts are provided to simplify connecting motes, compiling modules, and pushing compiled code. Running the Java listener and GUI are also explained for receiving and viewing sensor data from motes over a TCP/IP network connection.
In this workshop we will make a brief introduction to the basics of networking: IP addresses, MAC addresses, DNS, DHCP. Concepts as a router, gateway and firewall are explained. Then we will see in practice how to share files on a local network (NFS, Samba), establish a FTP connection, or log on to another (Linux) machine remotely (SSH, VNC, RDP). Finally, we review some useful networking tools like ping, netstat, lookup, port scan, traceroute, whois.
This document provides information on networking topics in Linux including:
- How to connect to Linux systems using SSH and things that can be done from the Linux command line interface
- IP addressing and subnet masking
- Setting up networks and creating permanent network configuration files
- Network troubleshooting tools like traceroute, nmap, netstat
- Reasons why network software may not work like firewalls blocking ports or network speed issues
- An overview of VPNs versus proxy servers and how each works
The document discusses IP Telephony and its infrastructure. It begins with an acknowledgement and background on telecommunications in Bangladesh. It then discusses combining wireless networks with VoIP. The key components of a VoIP infrastructure include softswitches, signaling gateways, and media gateways. Protocols like SIP are used to carry signaling information over IP networks.
1. The document introduces VoIP concepts and presents Asterisk as a free and open source PBX software solution that is well-suited for implementing VoIP networks in developing regions.
2. It discusses challenges in developing regions like lack of technical knowledge and affordable infrastructure, and how VoIP solutions like Asterisk can help address these issues by providing flexibility.
3. The document provides an overview of topics covered like basic VoIP concepts, how to set up an Asterisk PBX, equipment options, and presents a case study of introducing VoIP services.
The document provides information to help understand the differences between Windows XP Home and Professional editions. Key differences include:
- XP Pro includes features like backup software, dynamic disks, IIS, and encrypted file system that XP Home does not have.
- XP Pro supports up to two processors while XP Home only supports one.
- XP Pro allows systems to be domain members and supports group policies, while XP Home does not.
- Only XP Pro supports upgrades from Windows 2000/NT and will have a 64-bit version for Itanium systems.
This document provides an overview and objectives for an E-Learning training course on building a complete PBX with Asterisk. The training will cover understanding and installing Asterisk, building a simple PBX with SIP phones and SIP trunks, configuring features like dialplans, IVRs and voicemail, and integrating applications. By the end of the course, students will have hands-on experience building their own basic Asterisk system. Various chapters include objectives, presentations on topics, and instructions for labs to gain practical experience configuring Asterisk.
Time Sensitive Networking in the Linux Kernelhenrikau
Time Sensitive Networking provides mechanisms for sending data accross the network with very low latency, low jitter and low framedrops, opening up a whole range of new applications.
This talk primarily focuses on media, but the driver should be interesting for industrial applications and automotive as well.
SinoV-AP1000 asterisk IPPBX user manualCherry Jiang
Start from 2004, SinoVoIP Factory focus on Asterisk card, GSM Gateway, Opensource IPPBX, POE.
Should any support just kindly msg sinovoip@foxmail.com
We will full-support you in this field.
Thank you!
Cherry
Setting up VoIP management server using a communication framework and let the users from LTE networks to register and make voice calls over IP system as well as video session.
This document provides instructions for setting up and using the A400E analog telephony card with DAHDI on Linux. It includes information on hardware setup such as power supply and module installation. It also covers software installation and configuration of the A400E card with Asterisk and DAHDI drivers. The document contains safety instructions, an overview of Asterisk and the A400E card features, and chapters on hardware setup, software installation and configuration, specifications, and pin assignments.
The document discusses configuring VoIP telephony systems using the Yeastar S-Series IPPBX, including choosing an appropriate S-Series model based on company size, configuring basic network and system settings, setting up extensions and registering SIP phones and soft clients, and provisioning IP phones automatically to extensions.
It provides guidance on initial setup and configuration of the S-Series IPPBX to set up a functioning VoIP system for a business, covering hardware installation, network configuration, system prompts, extensions, and registering SIP endpoints.
The configuration process involves selecting an S-Series model, connecting modules, setting the network
The document describes Takeoka's Raspberry Pi cluster consisting of 3 nodes with a total of 6 processor cores. The cluster provides low-cost computing power and acts as a gateway and file server for Takeoka's home network. Takeoka hopes that future technologies like MRAM will enable new "normally off computing" architectures with even lower power consumption by allowing complete powering off of systems when not in use.
The document discusses techniques for hacking UNIX operating systems and provides instructions on how to gain unauthorized access. It begins by explaining the different versions of UNIX and how making them all compatible will make hacking easier. It then describes how to identify a UNIX system by its standard login and password prompts. The document provides usernames and likely passwords to try and gain initial access. It also discusses how to send anonymous email and view the password file. The overall purpose is to teach methods for compromising UNIX security without authorization.
By AWS & Espressif
Learn about Amazon FreeRTOS and the Espressif Systems’ ESP32-DevKitC. Determine how this fits into an IoT solution and look at examples of using cloud services in IoT microcontroller-based devices.
The document discusses open source VoIP and Asterisk. It summarizes that Asterisk is the most popular open source IP PBX option. While some enterprises were hesitant to adopt open source for phone systems due to concerns over support and compatibility, Asterisk has proven viable for large implementations. Asterisk can integrate analog and IP phones to replace traditional PBXs in a cost-effective manner.
Networking is divided into layers for modularity and standardization. The OSI model has 7 layers, while TCP/IP typically uses a 4 layer model. Each layer encapsulates the packet from the previous layer, adding a header. Packets are called frames, datagrams or segments depending on the layer. IP addresses are 32-bit numbers that specify networks and hosts. Subnet masks define the network and host portions. Routers use route tables to determine the next hop for packets based on the destination address.
Creating a console cable & making a console connectionIT Tech
The document provides instructions for creating a console cable and connecting to the console port of a Cisco EX90 unit. It lists the necessary tools which include a wire stripper, solder, soldering iron, DB9 enclosure, RS232 to TTL/CMOS converter, and an RJ45 cable. It then provides 7 steps for creating the console cable by stripping wires, soldering the converter, installing the enclosure, locating the EX90 console port, connecting the cable, connecting the DB9 to a computer, and configuring a terminal emulator. The process is similar for the Cisco EX60.
Optimizing the Audio Decoding Based Upon Hardware Capability: An Android NUPl...INFOGAIN PUBLICATION
Performance in android based devices is a major concern. A number of frame-drops can be seen while playing a 4k video on a less powerful device. A multi-purpose android set-top box should be capable of playing the 4k videos alongside of its basic functionalities. Sometimes the hardware allotted for the set-top box may not be up to the mark. Hence, it is difficult to handle 4k videos at a smoother rate. To tackle the above problem we have come up with the solution which tunnels the compressed/uncompressed audio track directly to the Smart-TV based upon its capability. The solution is built for ST’s set top box. In this paper we discuss basics of set top box, android audio architecture and the solution for frame drop backed by the experimental results and by using standard testing methods.
Optimizing the Audio Decoding Based Upon Hardware Capability: An Android NUPl...INFOGAIN PUBLICATION
Performance in android based devices is a major concern. A number of frame-drops can be seen while playing a 4k video on a less powerful device. A multi-purpose android set-top box should be capable of playing the 4k videos alongside of its basic functionalities. Sometimes the hardware allotted for the set-top box may not be up to the mark. Hence, it is difficult to handle 4k videos at a smoother rate. To tackle the above problem we have come up with the solution which tunnels the compressed/uncompressed audio track directly to the Smart-TV based upon its capability. The solution is built for ST’s set top box. In this paper we discuss basics of set top box, android audio architecture and the solution for frame drop backed by the experimental results and by using standard testing methods.
Subnetting of IPv4 ip address that help you to solve every type of ip address with any one of the class you want to subnet,and have a basic introduction of IPv6 ,and why, Ipv5 is not used.
This document summarizes the network design for ABC Company which has nine offices across multiple countries. The network will use a mesh topology with redundant fiber connections between offices. Servers located in Tokyo, Paris, and Detroit will provide 24/7 connectivity. Each office will use star configurations with fiber connections to switches on each floor. Cisco routers, firewalls, and blade servers will provide routing and security. VOIP phone systems and Cisco WebEx will enable conferencing. The network aims to support file transfers within 20 seconds between workstations using 1Gbps connections where possible.
Implementation of IPSec VPN on Cisco routers and Configuring it on ISP. (1)Vanitha Joshi
This document discusses implementing an IPsec VPN on Cisco routers and ISPs. It begins with introductions to TCP/IP concepts like layers, IP, and TCP. It then discusses existing VPN implementation methods and proposes using IPsec VPN. IPsec provides authentication and encryption of IP packets to secure communications. The document outlines the TCP/IP internet architecture including layers, IP, and protocols. It provides details on IP datagram format and fields. Finally, it discusses network topologies like bus, ring, and star, and components like hubs, switches, routers, and gateways.
The document provides a high-level overview of how the Internet works by explaining key concepts like Internet addresses (IP addresses), protocol stacks, packets, networking infrastructure, and domain names. It describes how computers on the Internet are assigned unique IP addresses and how messages are broken into packets that travel through a protocol stack consisting of application, transport, internet and hardware layers. It also outlines the physical infrastructure that connects computers, including connections to internet service providers, backbones, routers and network access points that packets traverse to reach their destination address on another computer connected to the Internet.
The document discusses the evolution and integration of IP telephony with traditional PSTN networks over six stages. It describes alternatives to legacy PBX systems such as using IP telephony to replace inter-building connections or fully replacing the PBX. The document also discusses concepts like least-cost routing, IP telephony gateways, packet-based switches, and integrating VoIP with existing PBX systems.
This document provides a summary of integrating IP telephony into the public switched telephone network (PSTN) environment. It discusses the evolution of PSTN from analog to digital networks and the emergence of time division switching. It then outlines the architecture of a soft switch solution for integration, including edge, core, control, and application layers. Finally, it discusses existing practices for integration, such as replacing tandem exchanges, and considers the specific context of integrating IP networks in Bangladesh.
1. The document introduces VoIP concepts and presents Asterisk as a free and open source PBX software solution that is well-suited for implementing VoIP networks in developing regions.
2. It discusses challenges in developing regions like lack of technical knowledge and affordable infrastructure, and how VoIP solutions like Asterisk can help address these issues by providing flexibility.
3. The document provides an overview of topics covered like basic VoIP concepts, how to set up an Asterisk PBX, equipment options, and presents a case study of introducing VoIP services.
The document provides information to help understand the differences between Windows XP Home and Professional editions. Key differences include:
- XP Pro includes features like backup software, dynamic disks, IIS, and encrypted file system that XP Home does not have.
- XP Pro supports up to two processors while XP Home only supports one.
- XP Pro allows systems to be domain members and supports group policies, while XP Home does not.
- Only XP Pro supports upgrades from Windows 2000/NT and will have a 64-bit version for Itanium systems.
This document provides an overview and objectives for an E-Learning training course on building a complete PBX with Asterisk. The training will cover understanding and installing Asterisk, building a simple PBX with SIP phones and SIP trunks, configuring features like dialplans, IVRs and voicemail, and integrating applications. By the end of the course, students will have hands-on experience building their own basic Asterisk system. Various chapters include objectives, presentations on topics, and instructions for labs to gain practical experience configuring Asterisk.
Time Sensitive Networking in the Linux Kernelhenrikau
Time Sensitive Networking provides mechanisms for sending data accross the network with very low latency, low jitter and low framedrops, opening up a whole range of new applications.
This talk primarily focuses on media, but the driver should be interesting for industrial applications and automotive as well.
SinoV-AP1000 asterisk IPPBX user manualCherry Jiang
Start from 2004, SinoVoIP Factory focus on Asterisk card, GSM Gateway, Opensource IPPBX, POE.
Should any support just kindly msg sinovoip@foxmail.com
We will full-support you in this field.
Thank you!
Cherry
Setting up VoIP management server using a communication framework and let the users from LTE networks to register and make voice calls over IP system as well as video session.
This document provides instructions for setting up and using the A400E analog telephony card with DAHDI on Linux. It includes information on hardware setup such as power supply and module installation. It also covers software installation and configuration of the A400E card with Asterisk and DAHDI drivers. The document contains safety instructions, an overview of Asterisk and the A400E card features, and chapters on hardware setup, software installation and configuration, specifications, and pin assignments.
The document discusses configuring VoIP telephony systems using the Yeastar S-Series IPPBX, including choosing an appropriate S-Series model based on company size, configuring basic network and system settings, setting up extensions and registering SIP phones and soft clients, and provisioning IP phones automatically to extensions.
It provides guidance on initial setup and configuration of the S-Series IPPBX to set up a functioning VoIP system for a business, covering hardware installation, network configuration, system prompts, extensions, and registering SIP endpoints.
The configuration process involves selecting an S-Series model, connecting modules, setting the network
The document describes Takeoka's Raspberry Pi cluster consisting of 3 nodes with a total of 6 processor cores. The cluster provides low-cost computing power and acts as a gateway and file server for Takeoka's home network. Takeoka hopes that future technologies like MRAM will enable new "normally off computing" architectures with even lower power consumption by allowing complete powering off of systems when not in use.
The document discusses techniques for hacking UNIX operating systems and provides instructions on how to gain unauthorized access. It begins by explaining the different versions of UNIX and how making them all compatible will make hacking easier. It then describes how to identify a UNIX system by its standard login and password prompts. The document provides usernames and likely passwords to try and gain initial access. It also discusses how to send anonymous email and view the password file. The overall purpose is to teach methods for compromising UNIX security without authorization.
By AWS & Espressif
Learn about Amazon FreeRTOS and the Espressif Systems’ ESP32-DevKitC. Determine how this fits into an IoT solution and look at examples of using cloud services in IoT microcontroller-based devices.
The document discusses open source VoIP and Asterisk. It summarizes that Asterisk is the most popular open source IP PBX option. While some enterprises were hesitant to adopt open source for phone systems due to concerns over support and compatibility, Asterisk has proven viable for large implementations. Asterisk can integrate analog and IP phones to replace traditional PBXs in a cost-effective manner.
Networking is divided into layers for modularity and standardization. The OSI model has 7 layers, while TCP/IP typically uses a 4 layer model. Each layer encapsulates the packet from the previous layer, adding a header. Packets are called frames, datagrams or segments depending on the layer. IP addresses are 32-bit numbers that specify networks and hosts. Subnet masks define the network and host portions. Routers use route tables to determine the next hop for packets based on the destination address.
Creating a console cable & making a console connectionIT Tech
The document provides instructions for creating a console cable and connecting to the console port of a Cisco EX90 unit. It lists the necessary tools which include a wire stripper, solder, soldering iron, DB9 enclosure, RS232 to TTL/CMOS converter, and an RJ45 cable. It then provides 7 steps for creating the console cable by stripping wires, soldering the converter, installing the enclosure, locating the EX90 console port, connecting the cable, connecting the DB9 to a computer, and configuring a terminal emulator. The process is similar for the Cisco EX60.
Optimizing the Audio Decoding Based Upon Hardware Capability: An Android NUPl...INFOGAIN PUBLICATION
Performance in android based devices is a major concern. A number of frame-drops can be seen while playing a 4k video on a less powerful device. A multi-purpose android set-top box should be capable of playing the 4k videos alongside of its basic functionalities. Sometimes the hardware allotted for the set-top box may not be up to the mark. Hence, it is difficult to handle 4k videos at a smoother rate. To tackle the above problem we have come up with the solution which tunnels the compressed/uncompressed audio track directly to the Smart-TV based upon its capability. The solution is built for ST’s set top box. In this paper we discuss basics of set top box, android audio architecture and the solution for frame drop backed by the experimental results and by using standard testing methods.
Optimizing the Audio Decoding Based Upon Hardware Capability: An Android NUPl...INFOGAIN PUBLICATION
Performance in android based devices is a major concern. A number of frame-drops can be seen while playing a 4k video on a less powerful device. A multi-purpose android set-top box should be capable of playing the 4k videos alongside of its basic functionalities. Sometimes the hardware allotted for the set-top box may not be up to the mark. Hence, it is difficult to handle 4k videos at a smoother rate. To tackle the above problem we have come up with the solution which tunnels the compressed/uncompressed audio track directly to the Smart-TV based upon its capability. The solution is built for ST’s set top box. In this paper we discuss basics of set top box, android audio architecture and the solution for frame drop backed by the experimental results and by using standard testing methods.
Subnetting of IPv4 ip address that help you to solve every type of ip address with any one of the class you want to subnet,and have a basic introduction of IPv6 ,and why, Ipv5 is not used.
This document summarizes the network design for ABC Company which has nine offices across multiple countries. The network will use a mesh topology with redundant fiber connections between offices. Servers located in Tokyo, Paris, and Detroit will provide 24/7 connectivity. Each office will use star configurations with fiber connections to switches on each floor. Cisco routers, firewalls, and blade servers will provide routing and security. VOIP phone systems and Cisco WebEx will enable conferencing. The network aims to support file transfers within 20 seconds between workstations using 1Gbps connections where possible.
Implementation of IPSec VPN on Cisco routers and Configuring it on ISP. (1)Vanitha Joshi
This document discusses implementing an IPsec VPN on Cisco routers and ISPs. It begins with introductions to TCP/IP concepts like layers, IP, and TCP. It then discusses existing VPN implementation methods and proposes using IPsec VPN. IPsec provides authentication and encryption of IP packets to secure communications. The document outlines the TCP/IP internet architecture including layers, IP, and protocols. It provides details on IP datagram format and fields. Finally, it discusses network topologies like bus, ring, and star, and components like hubs, switches, routers, and gateways.
The document provides a high-level overview of how the Internet works by explaining key concepts like Internet addresses (IP addresses), protocol stacks, packets, networking infrastructure, and domain names. It describes how computers on the Internet are assigned unique IP addresses and how messages are broken into packets that travel through a protocol stack consisting of application, transport, internet and hardware layers. It also outlines the physical infrastructure that connects computers, including connections to internet service providers, backbones, routers and network access points that packets traverse to reach their destination address on another computer connected to the Internet.
The document discusses the evolution and integration of IP telephony with traditional PSTN networks over six stages. It describes alternatives to legacy PBX systems such as using IP telephony to replace inter-building connections or fully replacing the PBX. The document also discusses concepts like least-cost routing, IP telephony gateways, packet-based switches, and integrating VoIP with existing PBX systems.
This document provides a summary of integrating IP telephony into the public switched telephone network (PSTN) environment. It discusses the evolution of PSTN from analog to digital networks and the emergence of time division switching. It then outlines the architecture of a soft switch solution for integration, including edge, core, control, and application layers. Finally, it discusses existing practices for integration, such as replacing tandem exchanges, and considers the specific context of integrating IP networks in Bangladesh.
VoIP can help improve rural development in Bangladesh in several sectors:
1. In agriculture, VoIP allows farmers to access information on crop quality, prices, and international markets through video conferencing.
2. In education, VoIP enables communication between rural schools and educators in cities through video conferencing and information sharing.
3. In healthcare, VoIP can help implement telemedicine services so rural doctors can consult with specialists in cities.
4. VoIP can also improve inter-district communication, transportation management, weather reporting, and establish call centers to boost the economy.
The document discusses how IP telephony can provide voice communication services to rural areas in Bangladesh. It describes how IP-based networks are well-suited for quickly implementing telephone infrastructure in rural regions. Specifically, the document outlines various IP network architectures that could be used to deliver rural telephony services and discusses technical aspects of implementing VoIP systems, including considerations around reliability, quality of service, emergency calls, and security.
This document discusses IP telephony solutions for banking institutions. It begins by providing background on banks and what they are. It then defines and discusses IP telephony, Voice over IP (VoIP), and IP-PBX systems. It provides examples of how IP telephony solutions can benefit banks, including call center applications and video conferencing. It also discusses how these solutions can help banks save money and improve efficiency.
This document summarizes the invitation for proposals from eligible Bangladeshi entities for licenses to provide international gateway services (IGW) and interconnection exchange (ICX) services in Bangladesh. Key details include:
- The Bangladesh Telecommunication Regulatory Commission (BTRC) will issue 3 IGW licenses and invites proposals by November 21, 2007.
- Licensing guidelines and application forms can be obtained from BTRC for 50,000 Taka. Proposals must follow all terms and conditions.
- BTRC will also issue licenses for other services like international internet gateway and one entity can apply for multiple licenses but will only receive one.
- A pre-bid meeting will be held on November 4, 2007 for
This document provides a case study on using an IP PBX solution over WiMAX technology. It begins with an abstract explaining that the paper will discuss IP PBXs and transmitting voice over IP networks, as well as how WiMAX can provide universal access. The body of the document then defines IP PBXs, their advantages, standards used, and how WiMAX can serve as the backbone for various voice and data applications like IP PBXs across different usage scenarios such as education and rural networks. It provides diagrams of sample network topologies and concludes that WiMAX enables flexible, high-bandwidth delivery of real-time applications and services.
1) Mobile operators are pursuing fixed-mobile convergence (FMC) using IP Multimedia Subsystem (IMS) platforms to deliver voice services over both fixed and mobile networks. However, as high-speed internet access comes to mobile phones, VoIP services will threaten mobile operators.
2) FMC allows mobile operators to leverage their large customer base initially. But as 3G data services and WiFi networking expand, VoIP providers like Skype will be able to directly compete.
3) To survive long-term, mobile operators must split their business into a mobile access provider and branded internet services, similar to how AOL transitioned from an integrated ISP/content provider. They need to develop internet brands now
This document discusses implementing Voice over IP (VoIP) and IP Multimedia Subsystem services over WiMAX wireless networks. It addresses introducing VoIP and multimedia transmission over wireless, using soft switching for compatibility with WiMAX. It also discusses challenges like ensuring voice quality, security, and E911 support. Finally, it explores services like video on demand that WiMAX networks can provide using IP Media Subsystem technologies.
This document is a project report on proposing an IP Telephony solution for disaster management in coastal areas of Bangladesh. It discusses protocols like SIP, H.323, and RTP that could be used to provide voice and video calling services. The report outlines a system design with remote branches connected to a main database server via internet, PSTN, and radio links to provide resilient communications even during disasters. The goal is to enable communication between coastal areas and the rest of the country when traditional networks are damaged.
The document discusses bandwidth requirements for IP telephony solutions in Bangladesh. It examines factors that influence bandwidth needs like codecs, sample rates, packet overhead. The G.729A codec requires the least bandwidth at 16 kbps per call. To serve a 50 seat call center would require a 800 kbps connection. The government needs to plan adequate bandwidth to support growing IP telephony demands.
This document discusses how IP-enabled contact centers can simplify multisite management. It provides an overview of how IP technology allows enterprises to consolidate their contact center infrastructure and applications into a single, centralized system. This centralized system can then manage contact centers across multiple locations in an integrated way. Specifically, the document discusses how IP-enabled contact centers can streamline contact center management, financial management, and IT management for multisite enterprises. It also provides examples of how companies like ebookers.com have successfully leveraged IP technologies to simplify their multisite contact center operations.
This document discusses Voice over Internet Protocol (VoIP) and its use in mobile communication networks. It provides details on VOIP functionality, implementation, reliability, quality of service, difficulties with faxing, integration into the global telephone numbering system, use on mobile phones and handheld devices, security considerations, and adoption of VOIP technology. The document examines the benefits of using VOIP in mobile networks, including IP backbone networks, redundancy, and technical requirements for supporting IP traffic. It also outlines VoIP architecture and provides references.
The document discusses VOIP issues in Nepal and potential solutions. It notes that while VOIP technology is not prohibited, providing end user telephony services is. It also discusses how VOIP started in Nepal due to high international calling rates by NTC and lack of competition. Potential solutions proposed include speeding telecom liberalization, discouraging technology-specific licensing, separating NTC into different units, and monitoring international telecommunications policies to prevent illegal use of technologies.
The document discusses setting up a call center in Bangladesh and available open source solutions. It notes that call centers handle large volumes of phone calls and can be used for customer service, sales, and more. Virtual call centers allow agents to work remotely while open source options like asterCRM provide call center software that integrates with Asterisk-based VoIP systems. The document provides an overview of call center technology, administration, types of calls centers and considers what is needed to start a call center in Bangladesh, including available training programs.
This document discusses quality of service (QoS) in voice over internet protocol (VoIP). It defines key terms like QoS and discusses why QoS is important for VoIP. Specifically, it states that QoS is needed to deliver high quality voice services over the internet by prioritizing VoIP packets to reduce delays. It also outlines some methods for implementing QoS, including at the network edge, and standards like bandwidth allocation that help ensure reliable VoIP call quality.
This document proposes a WiMAX VoIP solution for Bangladesh that would provide cost-effective communication services. It recommends establishing multi-purpose service centers in divisional headquarters that are connected via a fiber optic backbone and WiMAX networks. The centers would offer services like telemedicine, education, training, and support various industries. Implementing this infrastructure could promote rural development and economic opportunities in Bangladesh through affordable broadband and voice services.
This document discusses implementing a voting system through IP telephony. Key points include:
1. IP telephony equipment like phones, servers and gateways would be used to set up the system and allow voting through phone keypads or dedicated voting pads.
2. A voting server with a database would collect and count votes, ranking candidates based on vote totals.
3. Challenges include applying various voice and data network standards to a converged network and ensuring the same reliability as traditional voice networks.
4. Providing voting services through IP telephony could be profitable for service providers and beneficial for users through reduced costs.
This document provides an overview of the design and implementation of a wireless IP telephony system for telemedicine services in Bangladesh. It discusses:
1. The history and components of telemedicine and how IP telephony works.
2. The proposed design of a prototype wireless IP telephony system for telemedicine, which would enable integrated voice and data transmission for flexible medical services.
3. Some of the technical problems and economic aspects of implementing such a wireless IP telemedicine system, as well as the benefits to patients, practitioners, and telemedicine services.
Salesforce Integration for Bonterra Impact Management (fka Social Solutions A...Jeffrey Haguewood
Sidekick Solutions uses Bonterra Impact Management (fka Social Solutions Apricot) and automation solutions to integrate data for business workflows.
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1. [ASTERISK] April 15, 2008
Project on IP telephony
Prepared by :
Fatah Uddin
ID # 072831056
Ete ~ 605
Section ~ 02
Prepared for :
Dr. mashiur Rahman
Assistant Professor, Department of Engineering
North South University
Department of Engineering
North South University
1
2. [ASTERISK] April 15, 2008
What is Asterisk?
Asterisk PBX, from now on just called Asterisk, is Linux based, Open Source and free PBX
software. Asterisk is a complete PBX in software. It runs on Linux and provides all of the
features you would expect from a PBX and more. Asterisk does voice over IP in three protocols,
and can interoperate with almost all standards-based telephony equipment using comparatively
inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, and Call Queuing. It has support for three-way calling, caller ID
services, ADSI, SIP and H.323 (as both client and gateway).
Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and
analog telephony equipment, Asterisk supports a number of hardware devices, most notably all
of the hardware manufactured by Asterisk's sponsors, Digium. Digium has single and quad span
T1 and E1 interfaces for interconnection to PRI lines and channel banks. In addition, an analog
FXO card is available, and more analog interfaces are in the works.
Also supported are the Internet Line Jack and Internet Phone Jack products from Quicknet.
Asterisk supports a wide range of TDM protocols for the handling and transmission of voice
over traditional telephony interfaces. Asterisk supports US and European standard signaling
types used in standard business phone systems, allowing it to bridge between next generation
voice-data integrated networks and existing infrastructure. Asterisk not only supports traditional
phone equipment, it enhances them with additional capabilities.
Using the IAX Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across
disparate networks. While using Packet Voice, it is possible to send data such as URL
information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk provides a central switching core, with four APIs for modular loading of telephony
applications, hardware interfaces, file format handling, and codecs. It allows for transparent
switching between all supported interfaces, allowing it to tie together a diverse mixture of
telephony systems into a single switching network.
Asterisk is primarily developed on GNU/Linux for x/86. It is known to compile and run on
GNU/Linux for PPC. Other platforms and standards based UNIX-like operating systems should
be reasonably easy to port for anyone with the time and requisite skill to do so. Asterisk is
available in the testing and unstable debian archives, maintained thanks to Mark Purcell.
Who Made Asterisk?
Asterisk was originally written by Mark Spencer of Digium dba Linux Support Services, Inc.
Code has been contributed from Open Source coders around the world, and testing and bug-
patches from the community have provided invaluable aid to the development of this software.
2
3. [ASTERISK] April 15, 2008
Installing Linux for Asterisk
In order to use Asterisk you are going to have to use Linux. If you are a Linux ‘guru’ then you
might want to skip this section and just take a look at the dependencies. This section deals with
installing a Linux system for use with Asterisk. I make no excuses or arguments for the chosen
distribution, Red Hat 8; I have no intention of discussing the pros and cons of the many different
distributions available. In other words, if you don’t like Red Hat, stop bitching and find a
windows user to torment.
X100P Installation
I hope I can make the assumption that you are capable of adding a new card to your system, if
not get someone who knows what they are doing to help out. Oh and please wear a wrist strap.
The X100P card is what we call an FXO (Foreign Exchange Office) card. Unless you know
about telecoms then that probably didn’t mean much. What it means to you and me is that you
can plug a cable into the phone socket on your wall and into the back of the X100P; this is your
incoming line from your telecoms provider (KPN/BT/Bell etc).
You may have noticed that there are 2 sockets on the back of the X100P; one marked as the line
interface and the other as a phone interface. Plug in an appropriate cable between the wall socket
and the line interface on the card. If you have a spare phone then plug this into the phone
interface on the card too. It is always good to have a phone plugged into this interface because in
the event of asterisk failing, or a power cut the card actually still allows access to the PSTN line.
Obviously if you decide to use a phone that is not powered from the phone line, if you have a
power cut, it will not work.
TDM400P Installation
The TDM400P card is an FXS (Foreign Exchange Station) card; it’s what you plug standard
analog phones into to use them with asterisk.
Configuration
Once the cards have been physically plugged into your Asterisk machine, power it up again.
Once you’ve logged in you’ll need to configure the cards. It’s worth pointing out that you do not
have to have one of each card type but this section will make the assumption that you have got
one of each so that we can cover a much fuller configuration.
The first thing you’ll need to do is modprobe for the cards, basically this makes your linux
system ‘poke’ the card via its driver
The order in which you do the modprobe’s IS important. If you modprobe the FXO (modprobe
wcfxo) card first then it will be channel 1, if you modprobe the FXS (modprobe wcfxs) card first
then its first port will be channel 1, the second channel 2 and so on…
Now you need to tell asterisk which ports are for what… edit /etc/zaptel.conf
3
4. [ASTERISK] April 15, 2008
Note to exit vi hit the escape key then type : wq to write the file and quit. If you want to quit
without saving use q! instead of wq. My zaptel.conf .
The loadzone and defaultzone are fairly obvious and are simply your ISO
country code – Not all country codes are available, but to be honest I was
running with ‘us’ for a couple of months here in Holland with no issues..
The first 3 lines are very important so pay attention.
Just to confuse you more FXO ports use FXS signaling and FXS ports use FXO signaling. So
from the above configuration you can see that port 1 (fxsks=1) is actually an FXO card (X100P)
and ports 2 and 3 are on the FXS card (TDM20B). Note that you do not have to have the cards in
this order, but they must match the order you modprobed them. Since I have a single port FXO
card and a 2 port FXS (with the ability to add 2 more ports) it makes sense to put the FXO card
as the first, otherwise I’d have to change my configuration files a bit when (if) I added more FXS
ports.
Save your file and then edit /etc/asterisk/zapata.conf
This is where you set the configuration for each of the channels on the cards (you can think of
the channels as lines if you like). It’s important to note that the configuration in zapata.conf is
what I would call ‘backwards’ – i.e. you set the features for a channel, then
you assign the channel to them. There is actually a very good reason for this. We are only
dealing with a couple of channels so it’s not a big deal, but imagine if you were dealing with
128, 256 or more… Doing it ‘backwards’ allows you to set the configuration for all the channels
and assign them all at once (making slight modifications if required)
When you edit the file you may see things like
these are all pretty specific to your environment, they take either yes or no as the argument.
Hopefully what they do is obvious, for example:
gives you a call waiting tone if you are on the phone and someone else rings. Many of these
features need to be supported by your local telco (Bell/BT/KPN..etc)
At the bottom of this section are example zaptel and zapata config files that should work
immediately if your configuration MATCHES mine – i.e. you have 1 X100P and 1 TDM20B.
Some of the options available in this configuration file are as follows:
More than likely you will want to turn on echo cancellation
The gain options allow us to increase or decrease the ‘volume’ if the devices we have appear to
quite or loud – to be honest you should not really need to change the rx/tx gain settings under
normal circumstances.
Groups allow us to bundle devices together to act more or less as one, got example in Dial
strings you can use g<device-group-number>.
4
5. [ASTERISK] April 15, 2008
Setting up SIP
The files we are particularly interested in are sip.conf and extensions.conf. The first thing we will
need to do is edit sip.conf - Use whichever editor you are familiar with under Linux - I use vi. –
Note to exit vi hit the escape key then type : wq to write the file and quit. If you want to quit
without saving use q instead of wq. If you've ever looked at a windows .ini file you'll notice the
format is very similar. In the [general] section you need to add a line to register asterisk with
your sip proxy (eg ix66).An important point here, if you do not have a sip aware firewall and are
just using port forwarding then ensure that your context points to somewhere like ‘invalidcalls’.
If you do not do this then someone could call one of your extensions direct from the Internet. If
you had an FXO card in the machine, this could lead to them being able to make PSTN calls!!
A couple of things to note here, obviously the sip address "me@mysipproxy.com" is one that
someone would normally call you on. The /1000 at the end of the 'register =>' line is actually the
extension that asterisk will use for this address, ie when someone calls me@mysipproxy.com
extension 1000 will ring.
If you re registering with FreeworldDialup then you will need to add your password to the
register line too: e.g.
You now need a section to define the proxy information:
That part is fairly straight forward using these guidelines:
[section name] - the section name is the bit after the @ in your 'register =>' line
type = - this is a peer.
host= - the ip address of the ASTERISK server.
fromuser - username used for registration
secret - password for this account if required, if no password is required comment it out
using a ; in front
fromdomain
We now need to define a couple of phones. I use Snom 100's, but you may just want to use
sjphone (http://www.sjlabs.com) or something like that.
Since the configuration for both phones will be similar I'll only deal with 'phone1'. First of all
the [phone1] is important, it's what you will use in your extensions file to identify this physical
phone when issuing Dial commands. (e.g. Dial(SIP/phone1,20,tr) - we'll get to this later).
The type= is going to be a friend in this instance, peer is used when Asterisk is contacting a
proxy, user is used for phones that can only make calls, and friend acts as both a peer and a user.
Set your host=dynamic and the IP address in the defaultip= entry too. If your host has an entry
in your DNS then you just enter the machines name in the host= field. Unless you have a good
reason to leaving it as dynamic and using the IP given to the phone (fixed or DHCP allocated) is
perfectly ok.
The username and secret fields are only used when the username is not the same as the client.
You should be able to safely ignore this for the moment.
5
6. [ASTERISK] April 15, 2008
dtmfmode= is somewhat trial and error, it depends on your physical phone or software, I've
found that my Snom 100's need rfc2833 for DTMF tones to be heard, and SJphone needs inband.
Essentially it's trial and error, but once you establish which is needed for what type of phone, it
applies to all. This should not be installation specific, ie I've just told you that Snom 100's use
rfc2833, so you can use those too..(for the Snoms also change the phone config.
mailbox= is the voicemail mailbox that is checked for messages, this information is then passed
to the phone, on a Snom 100 phone it results in an envelope icon. Usually you'll have more than
one mailbox, one for each user or extension.
context= is extremely important you should, in the first instance make this the same for all your
sip clients. If a phone is not in a a valid context you will not be able to use it. I've used 'sip' you
can use whatever you like, but make sure they are the same, you will need to make an entry in
your extensions.conf file (which we will get to later)
callerid= is pretty self explanatory, however, note that the callerid must be in the format above,
ie "text" <number> it's the whole line that's the callerid, not just the bit in quotes. Essentially the
name goes in the quotes and the number between the < and >.
Once you've set up these phones you'll need to add some information in your extensions.conf
file:
# vi extensions.conf
You'll see there is a lot of information here, for the moment ignore this. Go to the bottom of the
file and add [sip] and press return. This needs to be the same as the context= section for your
phones defined in sip.conf surrounded by square brackets. If I had set the context= to wibble
(context=wibble) then this section would be called [wibble].
Next we'll add a couple of extensions for the phones defined in sip.conf
exten => 1,1,Dial(SIP/phone1,20,tr)
exten => 2,1,Dial(SIP/phone2,20,tr)
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
Add the above lines in the [sip] section. The format of the exten lines is fairly simple:
exten => extension number, command priority, command
Since an extension may have a number of commands there may be multiple entries for an exten.
We'll do some more complex stuff later but for the moment save this file.
When you start asterisk without any parameters it runs as a daemon. Enter the following
command:
# asterisk
6
7. [ASTERISK] April 15, 2008
If you get the error
ERROR[8192]: File asterisk.c, Line 1249 (main): Asterisk already running on
/var/run/asterisk.ctl. Use 'asterisk -r' to connect.
then asterisk is already running and you should use the command
# asterisk –r
to connect to the daemon, adding vvvvvvvvvgc to this will give us debug information too ie:
# asterisk –vvvvvvvgrc
If the daemon was running and you got the error type
reload
and hit enter at the prompt. If you did not get the error you do not need to do anything, however
you might want to attach to the daemon anyway using.
# asterisk –vvvvvvvgrc
You will need to configure your phones to connect to the asterisk machine. This varies from
device to device here is an example for Snom phones:
Under Home > Settings > SIP > Lines
Add the Name as phone1, the Account as phone1 and the Registrar as the IP of your asterisk pbx
and click save. Do the same for your second phone, replacing 'phone1' with 'phone2'.You should
now have 2 phones registered to asterisk (it may take a few seconds to actually happen). If you
like you can check by typing:
sip show peers
from the asterisk console, this should result is a display a little like the one below
Name/username Host Mask Port Status
phone1/phone1 192.168.1.4 (D) 255.255.255.255 5060 Unmonitored
7
8. [ASTERISK] April 15, 2008
phone2/phone2 192.168.1.3 (D) 255.255.255.255 5060 Unmonitored
Go to 'phone1' and dial '2', phone2 should start ringing. If not double check everything. If it does,
hang up and then go to phone2 and dial '1'. phone1 should start ringing. When someone calls the
address in your sip.conf (eg me@mysipproxy.com) both phones should ring. Of course all this is
wonderful if you only want to talk internally, but how can you talk to other people over the
internet? If you've registered with a proxy for example FreeworldDialup add an entry like this:
[fwd]
exten => _8.,1,Dial,SIP/${EXTEN-1}@fwd.pulver.com,tr
Next add
include => fwd
in the [sip] section of your extensions.conf. This will allow you to dial 8 then the users
FreeworldDialup number e.g. 8100001 would dial FWD's voicemail. On hard phones, this is
much easier than having to type a full sip address. Got some friends you call often over the
internet? just set up an exten line for them:
exten => 4002,1,Dial,SIP/keith@his-sip-address.com
Now you just dial 4002 and it will call them...
Voicemail - Please leave a message after the tone...
Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time.
What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need
to set up voicemail to capture all those messages if we miss them.
The first thing we need to do is create the mailbox for Asterisk to use, thankfully there is a little
utility to do this:
# /usr/src/asterisk/addmailbox
You'll be prompted for a mailbox number,
Enter mailbox number:
8
9. [ASTERISK] April 15, 2008
for now enter 9999 and hit return. This will copy a number of files required for the mailbox.
Voicemail files can be found in
/var/spool/asterisk/vm/
each mailbox has is a subdirectory from this point, so mailbox 9999 is
/var/spool/asterisk/vm/9999/
You don't really need to worry about these files, but at least you now know where they are on
your system. The next thing we need to do is modify a few files so that the mailbox can be used.
We need to edit the config files, so go to their location:
# cd /etc/asterisk
The first of the files we are interested in is voicemail.conf, edit the file
# vi voicemail.conf
The section we are interested in is [default]. If, which is likely, you have the sample configs
installed then you will see a number of entries in this file. You can comment them out if you like
by placing a semi-colon ';' in front of the line.
[default]
;1234 => 4242,Example Mailbox,root@localhost
Add the line
9999 => 1234,<your name in here>,<your email address here>
The format of the file is pretty simple:
mailbox number => mailbox password, mailbox description, mailbox user
email address
Once you have added the line for the 9999 mailbox, save the file. Next we'll need to edit out
sip.conf to make sure that the phones we set up earlier are going to use the correct mailbox for
notification.
9
10. [ASTERISK] April 15, 2008
# vi sip.conf
find the entries for [phone1] and [phone2] and change the lines that read:
mailbox=1000 ; Mailbox for message waiting indicator
to read
mailbox=9999 ; Mailbox for message waiting indicator
Note that this will be of most use if your phones support either a message waiting indicator of
some sort.
Save the file. For the moment we are going to use the same mailbox for both phones, but you can
set up one for each if you like. Now edit your extensions.conf file
# vi extensions.conf
locate the [sip] section where you added the entries for your extensions earlier in this document,
the lines should look similar to these:
exten => 1,1,Dial(SIP/phone1,20,tr)
exten => 2,1,Dial(SIP/phone2,20,tr)
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
we are going to add a few more lines to the exten => 1000 entry to add voicemail:
exten => 1000,2,VoiceMail,u9999
exten => 1000,102,VoiceMail,b9999
add these lines after the entry for exten => 1000, it should look like this
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
exten => 1000,2,VoiceMail,u9999
exten => 1000,102,VoiceMail,b9999
What have we done? Well, the exten entries are like a list of things to do when we get a call. In
this case we have said:
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
10
11. [ASTERISK] April 15, 2008
When extension 1000 rings (exten =>1000) , the first thing we do (exten =>1000,1) is dial
phone1 and phone 2 (exten =>1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)) and make it ring for
20 seconds.
exten => 1000,2,VoiceMail,u9999
If the extension is not answered in the 20 seconds, the second entry will be executed, which is
voicemail. The mailbox is specified by the u9999 at the end of the line. Now, this is all 'fine and
dandy' but what if you are actually on the phone when the call comes in? Well, this is what the
next line is for
exten => 1000,102,VoiceMail,b9999
There are 2 things to note here, the first is that the priority number has jumped way up to 102.
This is a feature of Asterisk, and a useful one at that. When the call comes in and Asterisk tries to
dial the extension 1000, if you are on the phone, Asterisk will jump to the current priority + 101
(n + 101). This gives us a priority of 102. The second thing to note is that the mailbox number at
the end of the line is preceded by a b (b9999) this indicates that the busy message should be
played to the user, and then the user should be allowed to leave a message.
This is all very well, people can leave you messages, but at the moment you can't listed to them!
Not really useful is it. We need to set up an extension so that you can actually get to the
messages. there are 2 ways to set this up. The first is useful in home environments where you are
only really going to have one mailbox or where you trust people completely: I say this because
the first way to do it is to effectively not use the password.(I'll show you the other was in a
moment). Under the exten => entries you've just added add the following lines.
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain,s9999
I've 'padded' this out so that you can follow the chain of events when extension 1001 is dialed.
Follow this sequence in this document (don;t actually try it on your phones yet)
When extension 1001 is dialed the following things happen in the following order,
1. The phone gets a ringing tone
2. There is a 2 second wait (the phone is still getting the ringing tone)
3. The call is answered and goes straight to the voicemail enu for mailbox 9999
Note the user doesn't get asked for a password. The second way to do this is to omit the ',s9999'
from the line giving:
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12. [ASTERISK] April 15, 2008
exten => 1001,3,VoicemailMain
When extension 1001 is dialed this time the following things happen in the following order,
1. The phone gets a ringing tone
2. There is a 2 second wait (the phone is still getting the ringing tone)
3. The call is answered and asked the user to enter a mailbox number, then a password. In our
examples above you would enter 9999 as the mailbox and 1234 as the password (remember, we
added the entry in voicemail.conf)
Save your extensions.conf file ... Sorry but we've been doing quite a bit here and I want to make
sure you remember to save the file. None of the changes you have made are functional yet, if you
went to one of your phones and tried to dial extension 1001, you'd more than likely get a
message telling you that the number was unavailable. We need to tell Asterisk to reload our
.conf files.. So as before (and yes, I used cut and paste :P ):
When you start asterisk without any parameters it runs as a daemon. Enter the following
command:
# asterisk
If you get the error
ERROR[8192]: File asterisk.c, Line 1249 (main): Asterisk already running on
/var/run/asterisk.ctl. Use 'asterisk -r' to connect.
then asterisk is already running and you should use the command
# asterisk -r
to connect to the daemon, adding vvvvvvvvvgc to this will give us debug information too
ie:
# asterisk -vvvvvvvgrc
If the daemon was running and you got the error type
reload
and hit enter at the prompt. If you did not get the error you do not need to do anything, however
you might want to attach to the daemon anyway using.
# asterisk -vvvvvvvgrc
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13. [ASTERISK] April 15, 2008
Now go to phone 1 and dial extension 1000, let phone 2 ring, don't answer it... You should hear
a voice announce that
"The person at extension 1000 is not available..."
After the beep, leave yourself a message and then hang-up. Then dial extension 1001 and listen
to your own message, make sure it all works.. Remember, when we first started, we set up a
registration with FWD and allocated extension 1000 to it? If you are using Snom phones and the
voicemail sounds 'funny', like the person is underwater add the following lines to your sip.conf
under the general section
disallow=all
allow=ulaw
allow=alaw
Save the file, go to the Asterisk console and type
reload
and try calling the voicemail again...it should sound a lot better. Sadly this uses more bandwidth,
but Snom has assured me that there will be a fix for this in their next firmware release.. (I have
no date on when that will be)
You're probably getting quite caught up in Asterisk by now, so take a look at the documentation,
see if the sections we've covered make more sense to you now...Well, that's all for now, next
time we'll look at Call parking, and meetme extensions.
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