Pingtel Phones

Image

sipXphone � The open source SIP softphone


sipXphone is a SIP softphone available for Windows (download) and Linux platforms ( sipXphone for Gentoo Linux). It is part of the sipX family of projects from SIPfoundry licensed under the L-GPL open source license. The sipXphone was formerly known as the Pingtel xpressa hard phone and instant xpressa softphone.

The sipXphone is an enterprise-grade open source SIP VoIP softphone for personal computer desktops and laptops. This voice-over-IP (VoIP) softphone delivers exceptional functionality, intuitive user interface, and audio quality to the Microsoft Windows and Linux Desktop environment.

The sipX softphone is typically used by road warriors and telecommuters who want to access an office phone when they travel. It provides users with the ability to carry their office phone and all of its features, applications, and directories on their laptop, while establishing connectivity to the Enterprise SIP server using the IPSec or SSL VPN capability already used for corporate email.

The sipXphone also offers integration between a user's phone and computer-based CRM, SFA, and other applications. The sipXphone is also appropriate for office workers who would prefer a computer-based phone that doesn't require any desk space.

Using the sipXphone it is possible for users to access speed dialing, corporate LDAP directories, call handling, call logging, dial from Microsoft Outlook or other contact managers, as well as other productivity enhancing applications. It includes a Java Virtual Machine that allows custom applications to be added very easily.

The phone is fully skinnable (sipXphone Screenshot).

Downloads and Documentation


Basic Phone Functionality


  • Multiple simultaneous calls per phone
  • Virtually unlimited speed dial numbers
  • Intuitive, graphically-assisted telephony features:
    • Hold
    • Transfer, consultative and blind
    • Call forwarding � no answer, busy, deflection (dynamic), do not disturb (static)
    • Multi-party conferencing with on-phone bridging
    • Call log
    • Redial
    • Caller ID
    • Ring mute
    • Hands-free microphone mute
    • Configurable internal and external dial plans
    • Dial number correction
  • User-specified ring sounds via .wav files

Easy-to-Use


  • Graphical user interface simplifies both traditional and next-gen features
  • Palm-size LCD image area (160x160 bitmapped) with sharp contrast grayscale, 11 soft keys, applications/help/menu button and "scroll knob"
  • 8 "hard" keys for call transfer, hold, conference, mute, headset, speakerphone, volume up, volume down
  • Standard 12-key dialing pad
  • Message waiting indicator light, one button retrieval
  • Browser interface for many phone features and functions

Customizable Java� Application Environment


  • Embedded PersonalJava VM
  • Java application framework for phone control
  • Development Kit (xDK) with Programmer's Reference Manual for specific APIs and xDK Guidelines for User Interface Design for recommendations on designing the user interface
  • JTAPI and STAPI to call control and audio systems
  • Window Toolkit (xWT) with GUI widgets
  • Java Naming and Directory Interface (JNDI) and Java Database Connector (JDBC) for directory and database integration
  • Remote Method Invocation (RMI) for distributed apps
  • Java security and internationalization support
  • Programming support by Java visual development toolkits

IP Telephony with Intelligence


  • Environment for applications that involve collaboration between a phone and PC
  • Microsoft Windows 98/NT4/2000/XP desktop libraries
  • Applications for personalizing ring tone and assigning ring tones based on caller ID
  • Web integration with a variety of network services for tracking packages, stocks and weather

Audio Quality


  • Codecs: G.711 A-law and u-law
  • Voice activity detection with silence suppression and comfort noise generation (available only for the Pingtel supported version as GIPS requires a license)
  • Packet loss compensation (Pingtel supported version)
  • Dynamic jitter buffer size to minimize latency and maximize audio quality (Pingtel supported version)
  • G729ab and GIPS iLBC low bit rate codec for speech quality equivalent to G.729 and G.723.1 but with superior packet loss robustness (Pingtel supported version)

Thorough Implementation of SIP


  • IETF RFC 3261 plus multiple draft SIP call control extensions
  • Voice main message waiting indication (MWI) � Internet Draft draft-ietf-sipping-message-waiting-01.txt
  • Out-of-band DTMF tones, as described in RFC 2833
  • Call transfer: consultative transfer (REFER and Replaces) as described by draft-ietf-sip-refer-05.txt and draft-ietf-sip-replaces-02.txt; blind transfer (REFER) as described by draft-ieft-sip-cc-transfer-05.txt
  • Locating SIP Servers � use of DNS SRV records for maintaining connections as described in RFC 3263
  • Codec Negotiation � use of Offer/Answer model for SDP as described in RFC 3264
  • Event Notification � use of SIP Events framework as described in RFC 3265 for presence, instant messaging, message waiting, configuration and other event packages
  • Early media (SDP in 180/183)
  • Delayed SDP (SDP in ACK)
  • Re-INVITE: Codec change, hold, off-hold, session timer
  • Call forwarding (302 redirect): Unconditional Forward, Forward on No Answer, Forward on Busy
  • Hold and Off Hold
  • REGISTER with refresh
  • Bridged conferencing
  • Supported header field
  • Route/Record-Route header fields
  • Configurable RTP/RTCP ports
  • Configurable SIP ports
  • UDP/TCP

Reliable, Fast And Secure


  • Proven performance on both Windows and Linux
  • Digest Authentication: Proxy, Incoming and Outbound - MD5, MD5-sess & QOP

Hardware Requirements


  • 600 MHz Pentium III processors and higher
  • Windows 98, 2000, NT 4 with Service Pack 4 and XP (the current Windows binary is only tested for XP)
  • 256 MB of RAM or more
  • Software driver for sound card that supports full duplex audio
  • Headset with microphone


Pingtel Hardphones




They have sold their hardware line of business, for some reason, rumored to 3com, and, as such, there is very little information about using their hardphone with Asterisk. This Pingtel Hardphone page is the start of agregating that information. All who have experience with their rather nice-looking, full-featured hardphone with Asterisk, kindly improve this page.


Pingtel Softphones


Press Release As of 8/16/04 Pingtel sold its hardware division (xpressa SIP hardphone) to focus soley on supporting the open source sipX PBX solution and the sipXphone softphone.

See also Pingtel, SIPfoundry, sipX, sipX Architecture



Where to buy




Image

sipXphone � The open source SIP softphone


sipXphone is a SIP softphone available for Windows (download) and Linux platforms ( sipXphone for Gentoo Linux). It is part of the sipX family of projects from SIPfoundry licensed under the L-GPL open source license. The sipXphone was formerly known as the Pingtel xpressa hard phone and instant xpressa softphone.

The sipXphone is an enterprise-grade open source SIP VoIP softphone for personal computer desktops and laptops. This voice-over-IP (VoIP) softphone delivers exceptional functionality, intuitive user interface, and audio quality to the Microsoft Windows and Linux Desktop environment.

The sipX softphone is typically used by road warriors and telecommuters who want to access an office phone when they travel. It provides users with the ability to carry their office phone and all of its features, applications, and directories on their laptop, while establishing connectivity to the Enterprise SIP server using the IPSec or SSL VPN capability already used for corporate email.

The sipXphone also offers integration between a user's phone and computer-based CRM, SFA, and other applications. The sipXphone is also appropriate for office workers who would prefer a computer-based phone that doesn't require any desk space.

Using the sipXphone it is possible for users to access speed dialing, corporate LDAP directories, call handling, call logging, dial from Microsoft Outlook or other contact managers, as well as other productivity enhancing applications. It includes a Java Virtual Machine that allows custom applications to be added very easily.

The phone is fully skinnable (sipXphone Screenshot).

Downloads and Documentation


Basic Phone Functionality


  • Multiple simultaneous calls per phone
  • Virtually unlimited speed dial numbers
  • Intuitive, graphically-assisted telephony features:
    • Hold
    • Transfer, consultative and blind
    • Call forwarding � no answer, busy, deflection (dynamic), do not disturb (static)
    • Multi-party conferencing with on-phone bridging
    • Call log
    • Redial
    • Caller ID
    • Ring mute
    • Hands-free microphone mute
    • Configurable internal and external dial plans
    • Dial number correction
  • User-specified ring sounds via .wav files

Easy-to-Use


  • Graphical user interface simplifies both traditional and next-gen features
  • Palm-size LCD image area (160x160 bitmapped) with sharp contrast grayscale, 11 soft keys, applications/help/menu button and "scroll knob"
  • 8 "hard" keys for call transfer, hold, conference, mute, headset, speakerphone, volume up, volume down
  • Standard 12-key dialing pad
  • Message waiting indicator light, one button retrieval
  • Browser interface for many phone features and functions

Customizable Java� Application Environment


  • Embedded PersonalJava VM
  • Java application framework for phone control
  • Development Kit (xDK) with Programmer's Reference Manual for specific APIs and xDK Guidelines for User Interface Design for recommendations on designing the user interface
  • JTAPI and STAPI to call control and audio systems
  • Window Toolkit (xWT) with GUI widgets
  • Java Naming and Directory Interface (JNDI) and Java Database Connector (JDBC) for directory and database integration
  • Remote Method Invocation (RMI) for distributed apps
  • Java security and internationalization support
  • Programming support by Java visual development toolkits

IP Telephony with Intelligence


  • Environment for applications that involve collaboration between a phone and PC
  • Microsoft Windows 98/NT4/2000/XP desktop libraries
  • Applications for personalizing ring tone and assigning ring tones based on caller ID
  • Web integration with a variety of network services for tracking packages, stocks and weather

Audio Quality


  • Codecs: G.711 A-law and u-law
  • Voice activity detection with silence suppression and comfort noise generation (available only for the Pingtel supported version as GIPS requires a license)
  • Packet loss compensation (Pingtel supported version)
  • Dynamic jitter buffer size to minimize latency and maximize audio quality (Pingtel supported version)
  • G729ab and GIPS iLBC low bit rate codec for speech quality equivalent to G.729 and G.723.1 but with superior packet loss robustness (Pingtel supported version)

Thorough Implementation of SIP


  • IETF RFC 3261 plus multiple draft SIP call control extensions
  • Voice main message waiting indication (MWI) � Internet Draft draft-ietf-sipping-message-waiting-01.txt
  • Out-of-band DTMF tones, as described in RFC 2833
  • Call transfer: consultative transfer (REFER and Replaces) as described by draft-ietf-sip-refer-05.txt and draft-ietf-sip-replaces-02.txt; blind transfer (REFER) as described by draft-ieft-sip-cc-transfer-05.txt
  • Locating SIP Servers � use of DNS SRV records for maintaining connections as described in RFC 3263
  • Codec Negotiation � use of Offer/Answer model for SDP as described in RFC 3264
  • Event Notification � use of SIP Events framework as described in RFC 3265 for presence, instant messaging, message waiting, configuration and other event packages
  • Early media (SDP in 180/183)
  • Delayed SDP (SDP in ACK)
  • Re-INVITE: Codec change, hold, off-hold, session timer
  • Call forwarding (302 redirect): Unconditional Forward, Forward on No Answer, Forward on Busy
  • Hold and Off Hold
  • REGISTER with refresh
  • Bridged conferencing
  • Supported header field
  • Route/Record-Route header fields
  • Configurable RTP/RTCP ports
  • Configurable SIP ports
  • UDP/TCP

Reliable, Fast And Secure


  • Proven performance on both Windows and Linux
  • Digest Authentication: Proxy, Incoming and Outbound - MD5, MD5-sess & QOP

Hardware Requirements


  • 600 MHz Pentium III processors and higher
  • Windows 98, 2000, NT 4 with Service Pack 4 and XP (the current Windows binary is only tested for XP)
  • 256 MB of RAM or more
  • Software driver for sound card that supports full duplex audio
  • Headset with microphone


Pingtel Hardphones




They have sold their hardware line of business, for some reason, rumored to 3com, and, as such, there is very little information about using their hardphone with Asterisk. This Pingtel Hardphone page is the start of agregating that information. All who have experience with their rather nice-looking, full-featured hardphone with Asterisk, kindly improve this page.


Pingtel Softphones


Press Release As of 8/16/04 Pingtel sold its hardware division (xpressa SIP hardphone) to focus soley on supporting the open source sipX PBX solution and the sipXphone softphone.

See also Pingtel, SIPfoundry, sipX, sipX Architecture



Where to buy




Created by: admin, Last modification: Mon 23 of Feb, 2009 (22:02 UTC) by smithonvoip
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