Modified items

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freeswitch-application-db-1.6.20-1.lbn25.x86_64: mod_db implements an API and dialplan interface to a database backend for FreeSWITCH. The database can either be in sqlite or ODBC. It also provides support for group dialing and provides database backed limit interface.

freeswitch-application-curl-1.6.20-1.lbn25.x86_64: Provide FreeSWITCH dialplan access to CURL

freeswitch-application-conference-1.6.20-1.lbn25.x86_64: Provide FreeSWITCH Conference Bridge Services.

freeswitch-application-cidlookup-1.6.20-1.lbn25.x86_64: Provide FreeSWITCH access to third party CallerID Name Databases via HTTP

freeswitch-application-callcenter-1.6.20-1.lbn25.x86_64: Provide Automated Call Distribution capabilities for FreeSWITCH

freeswitch-application-blacklist-1.6.20-1.lbn25.x86_64: Provide black/white listing of various fields used for routing calls in FreeSWITCH

freeswitch-application-avmd-1.6.20-1.lbn25.x86_64: Provide an voicemail beep detector for FreeSWITCH

freeswitch-application-abstraction-1.6.20-1.lbn25.x86_64: Provide an abstraction to FreeSWITCH API calls

freeswitch-1.6.20-1.lbn25.x86_64: FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow. We support various communication technologies such as SIP, H.323 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk. We also support both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 16 or 32 kilohertz and can bridge channels of different rates. FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms. Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including sipXecs, OpenSER, Asterisk, CodeWeaver and OpenPBX.

flite-2.0.0-31.lbn25.x86_64: Flite (festival-lite) is a small, fast run-time speech synthesis engine developed at CMU and primarily designed for small embedded machines and/or large servers. Flite is designed as an alternative synthesis engine to Festival for voices built using the FestVox suite of voice building tools.

erlang-xmpp-1.1.21-1.lbn25.x86_64: XMPP is an Erlang XMPP parsing and serialization library, built on top of Fast XML.

codec2-devel-0.8-1.lbn25.x86_64: Codec 2 is an open source (LGPL licensed) speech codec for 2400 bit/s and below. This package contains the development files required to compile programs that use codec2.

celt-0.11.3-2.lbn25.x86_64: CELT (Constrained Energy Lapped Transform) is an ultra-low delay audio codec designed for realtime transmission of high quality speech and audio. This is meant to close the gap between traditional speech codecs (such as Speex) and traditional audio codecs (such as Vorbis).

broadvoice-0.1.0-2.lbn25.x86_64: broadvoice is a library for the BroadVoice 16 and 32 speech codecs.

adns-1.4-19.lbn25.x86_64: adns is a resolver library for C (and C++) programs. In contrast with the existing interfaces, gethostbyname et al and libresolv, it has the following features: - It is reasonably easy to use for simple programs which just want to translate names to addresses, look up MX records, etc. - It can be used in an asynchronous, non-blocking, manner. Many queries can be handled simultaneously. - Responses are decoded automatically into a natural representation for a C program - there is no need to deal with DNS packet formats. - Sanity checking (eg, name syntax checking, reverse/forward correspondence, CNAME pointing to CNAME) is performed automatically. - Time-to-live, CNAME and other similar information is returned in an easy-to-use form, without getting in the way. - There is no global state in the library; resolver state is an opaque data structure which the client creates explicitly. A program can have several instances of the resolver. - Errors are reported to the application in a way that distinguishes the various causes of failure properly. - Understands conventional resolv.conf, but this can overridden by environment variables. - Flexibility. For example, the application can tell adns to: ignore environment variables (for setuid programs), disable sanity checks eg to return arbitrary data, override or ignore resolv.conf in favour of supplied configuration, etc. - Believed to be correct ! For example, will correctly back off to TCP in case of long replies or queries, or to other nameservers if several are available. It has sensible handling of bad responses etc.

ptlib-2.10.10-16.fc24.x86_64: PTLib (Portable Tools Library) is a moderately large class library that has it's genesis many years ago as PWLib (portable Windows Library), a method to product applications to run on both Microsoft Windows and Unix systems. It has also been ported to other systems such as Mac OSX, VxWorks and other embedded systems. It is supplied mainly to support the OPAL project, but that shouldn't stop you from using it in whatever project you have in mind if you so desire.

pjsua-2.4.5-8.lbn25.x86_64: pjsua is an open source command line SIP user agent (softphone) that is used as the reference implementation for PJSIP, PJNATH, and PJMEDIA. Despite its simple command line appearance, it does pack many features!

pjproject-2.4.5-8.lbn25.x86_64: This package provides the Open Source, comprehensive, high performance, small footprint multimedia communication libraries written in C language for building embedded/non-embedded VoIP applications. It contains: - PJSIP - Open Source SIP Stack - PJMEDIA - Open Source Media Stack - PJNATH - Open Source NAT Traversal Helper Library - PJLIB-UTIL - Auxiliary Library - PJLIB - Ultra Portable Base Framework Library - PJSUA2 - Object Oriented abstractions layer for PJSUA

opal-3.10.10-9.fc23.x86_64: Open Phone Abstraction Library, implementation of the ITU H.323 teleconferencing protocol, and successor of the openh323 library.

linphone-devel-3.12.0-2.git74fd021.lbn25.x86_64: Linphone is a SIP compliant audio & video phone. It can be used to run calls over the internet. It has a gtk+ and console interface. This package contains development files for linphone.

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