Archivio per la Categoria 'Voip'

GTK Porting Asterisk CTI

English Version
astcti_gtk.jpg astcti_gtk_lin.jpg

Ho iniziato da qualche settimana una ristrutturazione generale di Asterisk CTI. La nuova versione, per la quale un rilascio è previsto orientativamente per i primi mesi del 2009, rappresenta una rivoluzione rispetto alla prima versione del progetto.

Le principali novità sono:

  • Approccio completamente rinnovato: centralizzazione sul server della configurazione dei servizi e dei client abilitati ad essere notificati delle chiamate, nonchè degli applicativi da lanciare sulle postazioni.
  • Software completamente riscritto in C# con mono, usando le librerie GTK. Questo consentirà di avere un software realmente multipiattaforma.
  • Client, denominato AstCTIClient, riscritto in GTK e multipiattaforma.
  • Tutta la configurazione potrà essere gestita da un software chiamato AstCTIConfigurator. e verrà salvata in un database MySQL.
  • Le configurazioni verranno salvate in un file XML caricato all’avvio da AstCTIServer: questo renderà il server indipendente da eventuali failures del database MySQL.
  • Supporto multilanguage.

Naturalmente si tratta di un progetto Open Source sotto licenza GPL e a breve provvederò a caricare i sorgenti su una apposita pagina di progetto su google-code. Attendo vostri commenti, feedback e, sopratutto una partecipazione al progetto!

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

A108 Octal T1 / E1 / J1

a108.jpg

Caratteristiche tecniche:

  • Eight T1/E1 ports with a single PCI interface for high performance voice and data applications.
  • TDM clocking mode: Network timing can be passed from a network -connected DS0 to any or all of the other ports. Both T1 and E1 are supported simultaneously, making it possible to mix T1 channel banks and E1 networks with full channel synchronization.
  • Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1.
  • Support for the Asterisk™, OPAL™ Yate, FreeSwitch™ PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications.
  • All Sangoma’s AFT products, including the A108 card use the same base PCI interface card, and the same professionally engineered firmware on the same family of Field Programmable Gate Arrays.
  • A108, A108d PCI: Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupt sharing without manual tuning.
  • A108-X, A108d-X PCI Express: 1 Lane PCI Express bus.
  • Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis.
  • Short 2U compatible mounting clips included for installation in 2U rackmount servers.
  • High quality, tested RJ45 cables included. Power: 800mA peak, operational 300mA max at +3.3v or 5v.Temperature range: 0 – 50C. Autosense compatibility with 5v and 3.3v PCI busses.
  • Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available.
  • 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention.
  • Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems.
  • Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI.
  • T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications.
  • Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU.
  • EDAC™ (patent pending) technology will be integrated to drastically reduce the cost of echo cancellation.
  • Field upgradeable hardware allows for new TDM-related functions to be added as they become available.
  • Raw bitstream interfaces can be used to support arbitrary non-standard line protocols such as non-byte aligned monosynch or bisynch.
  • WAN data connection is supported by Sangoma’s standard WANPIPE® routing stack, and is completely independent of TDM voice application for total system reliability.
  • WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25.
  • Optional DSP daughterboard on the A108d
  • G.168-2002 echo cancellation in hardware.
  • 1024 taps/128ms tail per channel on all 256 channels.
  • DTMF decoding and tone recognition.
  • Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction.

Pagina del produttore: A108

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

A104 Quad T1 / E1 / J1

a104.jpg

Caratteristiche tecniche:

  • Four T1/E1 ports with a single PCI or PCI-Express interface optimized for high performance voice and data applications.
  • Mix T1 channel banks and E1 networks with full channel synchronization. TDM clocking mode lets network timing to be passed from a network-connected DS0 to any or all of the other ports, so both T1 and E1 are supported simultaneously.
  • Support for Asterisk®, Yate™, CallWeaver™ and FreeSwitch™ PBX/IVR Projects, as well as other open source and proprietary PBX, Switch, IVR, or VoIP gateway applications.
  • All of Sangoma’s AFT products use the same base PCI interface card. Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis.
  • Includes high-quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack-mount servers.
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available.
  • Autosense compatibility with 5V and 3.3V PCI busses.
  • Line decoding: HDB3, AMI, B8ZS.
  • Framing: CRC-4, Non CRC4, ESF, SF, D4T1/E1. Also compatible with Japan’s J1.
  • A104-X and A104D-X PCI Express: 1 Lane PCI Express bus.
  • Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V.
  • Temperature range: 0–50 °C.
  • 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
  • Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems.
  • Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI.
  • T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications.
  • Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU.
  • Uses raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch.
  • WANPIPE® routing stack is completely independent of TDM voice application for total system reliability.
  • WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25.
  • Optional DSP Hardware Echo Canceller Daughterboard
  • G.168–2002 echo cancellation in hardware.
  • 1024 taps/128 ms tail per channel on all channel densities.
  • DTMF decoding and tone recognition.
  • Voice quality enhancement: Octasic® music protection, acoustic echo control, and adaptive noise reduction.

Pagina del produttore: A104

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

AFT A102 / A102D Dual T1 / E1 / J1

a102.jpg

Caratteristiche tecniche:

  • Two T1/E1 ports with a single PCI or PCI-Express interface optimized for high performance voice and data applications.
  • Mix T1 channel banks and E1 networks with full channel synchronization. TDM clocking mode lets network timing to be passed from a network-connected DS0 to any or all of the other ports, so both T1 and E1 are supported simultaneously.
  • Support for Asterisk®, Yate™, CallWeaver™ and FreeSwitch™ PBX/IVR Projects, as well as other open source and proprietary PBX, Switch, IVR, or VoIP gateway applications.
  • All of Sangoma’s AFT products use the same base PCI interface card. Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis.
  • Includes high-quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack-mount servers.
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available.
  • Autosense compatibility with 5V and 3.3V PCI busses.
  • Line decoding: HDB3, AMI, B8ZS.
  • Framing: CRC-4, Non CRC4, ESF, SF, D4T1/E1. Also compatible with Japan’s J1.
  • A102-X and A102D-X PCI Express: 1 Lane PCI Express bus.
  • Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V.
  • Temperature range: 0–50 °C.
  • 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
  • Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems.
  • Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI.
  • T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications.
  • Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU.
  • T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications.
  • Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU.
  • Uses raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch.
  • WANPIPE® routing stack is completely independent of TDM voice application for total system reliability.
  • WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25.

Sito del produttore: A102

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

AFT A101 / A101D Single T1 / E1 / J1

a101.jpg

Caratteristiche tecniche:

  • One T1/E1 port with optimum PCI Express or PCI interface for high performance voice and data applications.
  • Line decoding: HDB3, AMI, B8ZS.
  • Framing: CRC-4, Non CRC4, ESF, SF, D4. Also compatible with Japan’s J1.
  • Support for AsteriskTM, YateTM, and FreeSwitchTM PBX/IVR Projects, as well as other open source and proprietary PBX/Switch/IVR/VoIP gateway applications.
  • All of Sangoma’s AFT products use the same base PCI interface card and the same professionally engineered firmware on the same family of Field Upgradeable Gate Arrays.
  • Fully compatible with all commercially available motherboards—proper PCI-standard interrupt sharing without manual tuning.
  • A101-X and A101D-X PCI Express: 1 Lane PCI Express bus.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis.
  • Includes high quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack mount servers.
  • Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V.
  • Temperature range: 0–50 °C.
  • Autosense compatibility with 5 V and 3.3 V PCI busses.
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available.
  • 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
  • Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems.
  • Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI.
  • T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications.
  • Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU.
  • Use raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch.
  • WANPIPE® routing stack is completely independent of TDM voice application for total system reliability.
  • WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25.
  • Optional DSP daughterboard on the A101D
  • G.168–2002 echo cancellation in hardware.
  • 1014 taps/128ms tail per channel on all 256 channels.
  • DTMF decoding and tone recognition.
  • Voice quality enhancement: Octasic music protection, acoustic echo control, and adaptive noise reduction.

Sito del produttore: AFT A101

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

A500 BRI

a500.jpg

Caratteristiche tecniche:

  • From 2 to 24 ports are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers — simply invert the colour-coded module.
  • Supports Asterisk®, Yate™,   FreeSwitch™,   CallWeaver™, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications.
  • Single synchronous PCI and PCI Express interface for all 24 BRI interfaces.
  • Six ports per RemoraTM card.
  • Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis.
  • Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables included.
  • 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
  • Autosense compatibility with 5 V and 3.3 V PCI busses.
  • Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts.
  • Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available.
  • Power: 800mA peak, operational 300mA max at +3.3 V or 5 V.
  • Temperature range: 0 – 50°C.
  • Optimized DMA stream and hardware-level HDLC handling unload the host CPU.
  • Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch.
    WANPIPE® supports certified, field tested, and reliable Frame Relay, PPP, HDLC and X.25.

Pagina del produttore: A500 BRI

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

A400 FXO/FXS

A400.jpg

Caratteristiche tecniche:

  • From 2 to 48 ports supported, mixing FXO and FXS interfaces as required.
  • Support for the Asterisk™, Yate™, FreeSwitch™, CallWeaver™, OPAL™ PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications.
  • Single synchronous PCI interface up to 48 FXO/FXS ports.
  • Connects to punch block using a standard 12-line colour coded telephone cable. (Included.)
  • Dimensions: 2U height form factor: 290mm x 55mm for use in a 2U chassis.
  • Short 2U compatible mounting clips available for installation in 2U rack mount servers.
  • 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention.
  • Autosense compatibility with 5 V and 3.3 V PCI busses.
  • Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupts sharing.
  • Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available.
  • Power: 800mA peak, operational 300mA max at +3.3 V or 5 V.
  • Temperature range: 0 – 50C.
  • Optional DSP daughterboard on the A400D
    • G.168-2002 echo cancellation in hardware
    • 1024 taps/128ms tail per channel on all channel densities
    • DTMF decoding and tone recognition
    • Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction.
    • No CPU Load

Sito del produttore: A400 FXO/FXS

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

A200 FXO/FXS

A200.jpg

Caratteristiche tecniche:

  • From 2 to 24 ports supported, mixing FXO and FXS interfaces, as required.
  • Support for the Asterisk™, Yate™, FreeSwitch™, CallWeaver™, OPAL™ PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications.
  • Single synchronous PCI interface for all 24 FXO/FXS ports.
  • Four RJ11 ports per Remora™ card.
  • Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis.
  • Short 2U compatible mounting clips available for installation in 2U rack-mount servers.
  • 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention.
  • Autosense compatibility with 5 V and 3.3 V PCI busses.
  • Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts.
  • Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features can be added when they become available.
  • Power: 800mA peak, operational 300 mA max at +3.3 V or 5 V.
  • Temperature range: 0 – 50C.
  • Optional DSP Echo Canceller Daughterboard on the A200D
    • G.168-2002 echo cancellation in hardware
    • 1024 taps/128ms tail per channel on all channel densities
    • DTMF decoding and tone recognition
    • Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction.
    • No CPU Load

Sito del produttore: A200 FXO/FXS

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

GXP2020

grandstram-gxp2020.jpg

Specifiche tecniche:

  • Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
  • Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af)
  • Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
  • Backlit 320×160 high resolution graphic LCD with multi-level grey scales
  • 6 lines indicators with individual SIP account profiles
  • 7 programmable hard keys and 4 XML programmable context-sensitive soft keys
  • Multi-line support of up to 13 call appearance lines with dual-color LED indicators
  • Expandable to additional 112 lines through expansion key-modules
  • Support Caller ID display or block, per call or permanent
  • Call waiting, hold, mute, transfer (blind or attended), forward, and more
  • Multi-party conferencing (up to 5-way)
  • And many more enterprise grade features
  • Full duplex speakerphone w/advanced audio echo cancellation
  • Headset jack (2.5 mm and RJ11) for broad headset compatibility
  • Secure and automated provisioning for mass deployment
  • SRTP and TLS (pending) for privacy protection
  • And many more enterprise grade features

Sito del produttore: GXP2020

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

GXP2000

grandstream-gxp2000.jpg

Specifiche tecniche:

  • Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
  • Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
  • Multiline support of up to 11 lines indicators (expandable to 112 lines through expansion key-module)
  • Graphical LCD to display up to 8 lines and 22 characters per line
  • Dual 10/100Mbps Ethernet ports
  • 2.5mm headset jack (versions 0.3 and 0.4 – 3.5mm headset jack)
  • Support Caller ID display or block, per call or permanent
  • Call waiting, hold, mute, transfer (blind or attended),forward, and more
  • 3-way conferencing
  • Integrated Power-over-Ethernet (802.3af)
  • And many more enterprise grade features

Sito del produttore: GXP2000

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Tag: Sistemi di Call Center | Centralino Voip | Consulente Asterisk Napoli | Centralino Telefonico | Voip | Asterisk CTI | PBX | Telefoni IP | Networking | Linux


Sviluppo di IVR, sistemi di callcenter, PBX Voip.

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