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TCDialer 1.0

TCDialer 1.0


TCDialer is a DTMF dialer. more>>
TCDialer is a DTMF dialer. Designed to resemble telephone keypad. Useful for those who end up in a hotel with a pulse phone.

DTMF generator for those without a touch-tone phone.

Push the buttons. Hear the DTMF tones. Pretty simple.

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Download (0.019MB)
Added: 2006-10-20 License: GPL (GNU General Public License) Price:
1101 downloads
mgetty+sendfax 1.1.35

mgetty+sendfax 1.1.35


mgetty+sendfax is an intelligent getty with fax and voice modem support. more>>
Mgetty+sendfax is a collection of programs to send and receive faxes in a unix environment using a class 2.0 or 2 (theyre different) faxmodem. vgetty is an extension to mgetty, distributed with it, that implements incoming voice call handling for certain voice-capable modems, with new ones added regularly, if specs are available.
More specifically, the program `mgetty allows you to use a class 2.0 or 2 fax modem for receiving faxes and handling external logins without interfering with outgoing calls.
`sendfax is a standalone program which sends fax files. `vgetty is an extended version of mgetty that can answer the telephone like an answering machine and record a voice-mail message (if it finds one), or perform `mgettys fax or data call handling otherwise.
The mgetty+sendfax distribution includes vgetty and a good-sized gob of utility programs that help you manage faxes and voice messages.
Main features:
- send faxes directly or using shell scripts (easily integrated into other applications).
- do "fax polling", this means you can call the weather station and get them to send you a fax containing the current weather map. (Not all modem manufacturers implement this feature in their modems!)
- create a "fax queue", outgoing faxes get sent automatically, the user is informed by mail about the result.
- `mgetty allows you to use a single modem line for receiving calls and dialing out.
- `mgetty knows about "smart" modems, and will make sure that the modem is always in a defined state (specific modem initialization possible)
- Incoming calls are answered manually (`RING -> `ATA -> `CONNECT) instead of using auto-answer (`ATS0=1), this way the modem wont pick up the phone when the machine is down or logins are not allowed. (but see note below for ISDN/digital modems)
- mgetty completely replaces getty and/or uugetty. Like uugetty, supports lock files in a fashion compatible with almost all known versions of UUCP (HDB/BNU, SVR4, V7, Taylor in various flavours). uugetty has some features mgetty doesnt support; see "How does mgetty differ from uugetty?" below.
- mgetty supports System V style gettydefs terminal configurations.
- mgetty can receive class 2 faxes (if your modem supports it).
- mgetty knows about incoming FidoNet calls.
- mgetty has extensive logging / debugging features
- do "fax poll sending", that is, you can setup your machine as fax poll server, to send some fax pages to "fax poll" callers. (Send informations about your system, the current wheather map, ...). Be warned, even less modems support this feature.
- mgetty can selectively refuse calls based upon CallerID, if your modem supports it, and youre subscribed to the service. CallerID is also logged.
- mgetty has facilities to allow you to refuse incoming FAXes when available disk space is low.
- mgetty knows about incoming PPP calls, and can hand them off to the PPP-daemon, without requiring a login/password sequence. This feature is also known as AutoPPP
- behaves like a normal answering machine for human callers
- automatic fax reception when a T.30 calling tone is detected
- If the caller isnt a human or fax, a data connect is attempted, if this is successful, the caller will get a normal login
- does not interfere with dialouts
- remote playback of messages via a DTMF code
- toll saver -- if there are new messages, pick up the phone earlier, this way you can hang up in time to avoid a useless call
- message light - the autoanswer LED of your modem (if it has one) is turned on if there are new messages
- easy playback - on some modems, you can play back the new messages just by pressing DATA/VOICE
- using a speech synthesizer is possible - add the date and time to messages (not included by default). The scripts show how to use a speech synthesizer like rsynth, but it is not included in the package. To use this feature, you need a voice modem for that; a converter from the pvf format to the rmd (raw modem data) format exists. This is not true for all supported modems.
- voice conversion utilities - play messages on /dev/audio (Not for all supported modems, some voice modems use a proprietary format)
- and more, more features available through the voice library/mvm
Enhancements:
- This release adds class 1/1.0 fax reception (still considered experimental).
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Added: 2006-02-25 License: GPL (GNU General Public License) Price:
1342 downloads
ivam2 0.3

ivam2 0.3


ivam2 is an automatic phone answering machine software for ISDN and Linux. more>>
ivam2 is an automatic phone answering machine software for ISDN and Linux. It is the completely rewritten successor of ivam featuring many additions.

ivam2 is seperated in two parts: the core daemon written in C and the automate logic coded in Python. The latter is pluggable and may be replaced by different implementations on a per phone number basis (both caller and callee). This makes ivam2 a very powerful application server for telephony services.

The software is very scalable, multiple ISDN channels may be controlled from a single daemon. To write telephony applications for ivam2 is not complicated. In fact, they are simple executables which read audio data of the caller from STDIN and write audio data for the caller to STDOUT.

DTMF sequences may be read from a FIFO special file. A framework for writing telephony applications in Python is provided, a simple answering machine script based on this framework as well. You are free to write applications in other languages such as Perl or C.

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Download (0.30MB)
Added: 2006-05-17 License: GPL (GNU General Public License) Price:
1255 downloads
JVoiceXML 0.5.5

JVoiceXML 0.5.5


JVoiceXML is an implementation of VoiceXML 2.1, the Voice Extensible Markup Language. more>>
JVoiceXML is an implementation of VoiceXML 2.1, the Voice Extensible Markup Language. JVoiceXMLs specification can be found at http://www.w3.org/TR/2005/CR-voicexml21-20050613/ as an extension to VoiceXML 2.0, specified at http://www.w3.org/TR/voicexml20/.

VoiceXML is designed for creating audio dialogs that feature synthesized speech, digitized audio, recognition of spoken and DTMF key input, recording of spoken input, telephony, and mixed initiative conversations.

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Download (0.64MB)
Added: 2007-03-01 License: LGPL (GNU Lesser General Public License) Price:
968 downloads
chan_misdn 0.2.1

chan_misdn 0.2.1


chan_misdn is a channel driver for the open source PBX Asterisk for using ISDN BRI/PRI devices that are supported by mISDN. more>>
chan_misdn is a channel driver for the open source PBX Asterisk for using ISDN BRI/PRI devices that are supported by mISDN. chan_misdn is the new ISDN Layer for Linux.
Main features:
- NT and TE mode
- PP and PMP mode
- BRI and PRI (with BNE1 and BN2E1 Cards)
- DTMF Detection in HW+mISDNdsp (much better than asterisks internal!)
- Display Messages to Phones (which support display msg)
- HOLD/RETRIEVE/TRANSFER on ISDN Phones : )
- Screen/ Not Screen User Number
- Basic EchoCancellation
- Volume Control
- Crypting with mISDNdsp (Blowfish)
- Data (HDLC) callthrough
- Data Callin (with app_ptyfork +pppd)
- echo cancellation
- some other
Enhancements:
- Release management was changed.
- This is the first stable version.
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Download (0.12MB)
Added: 2005-12-06 License: GPL (GNU General Public License) Price:
1418 downloads
Pocket Linux 2.51

Pocket Linux 2.51


Pocket Linux is an almost minimal, one floppy linux system designed to quickly convert PC workstation into a secure linux. more>>
Pocket Linux is an almost minimal, one floppy linux system designed to quickly convert PC workstation into secure linux-based workstation using ssh to connect to remote host (other networking clients are also supported).
It supports bootp for determining host IP and other network parameters (theres also manual configuration possible, but bootp is recommended).
In addition to workstations equipped with a network card (ethernet or arcnet), you can also use Pocket Linux on a PC equipped with a modem. Modem is automatically detected and then PPP connection is made.
The idea came up some time in 1996 or so. The distribution then was not perfect, but still it shown it was a great idea. It wasnt maintained for about year or so, until I took it up again in the early January 1998. After a complete rebuild Pocket Linux 2.00 was released. It soon gained a huge number of happy users, whose ideas helped its development.
The aim is to provide a small and efficient workstation that autoconfigures as much as possible and lets securely use the network from almost everywhere.
Current version is a nice attempt and future ones will enhance the automation and support for various network equipment and protocols, becoming a total solution. Future plans also include side projects like one floppy router.
In order to understand some of the config options its useful to know something about operations that are done during bootup (in order to automatically configure the network). These are, in order (the later attempts are made if the earlier ones dont set-up the network):
- attempt to setup the network using BOOTP
- attempt to reuse previous manual configuration
- modem detection
- attempt to setup modem conection
Most of the config options switches these operations on and off.
Main menu
You can choose the following network configuration commands from the main menu (only the ones that make sense in the present context are displayed):
- Options - allows setting few binary parameters controlling automatic network configuration and modem handling.
- BOOTP query - attempts to configure network (ethernet or arcnet) using BOOTP. Normally its automatically done during bootup, but this can be switched off.
- Manual configuration - allows manual setting of network configuration parameters (ethernet or arcnet).
- Detect modem - detects serial port the modem is on and its parameters (transfer rate, initialization commands).
- Dial PPP using predefined configuration - creates modem connection (PPP) using one of (up to ten) remembered configurations. By default only TPSA (0202122) configuration is remembered.
- Dial PPP using new configuration - creates new PPP configuration and sets up a modem connection using the newly created config.
- Disconnect PPP - disconnects modem connection.
- Mount /usr via NFS - mounts remote /usr filesystem via NFS. It will be automatically mounted during each Pocket Linux bootup if its turned on in configuration options.
- Exit - Do not config the network - exits the program without configuring the network.
Configuration options
There are following options available:
- Probe network with BOOTP - switches automatic BOOTP probing during bootup on and off. On by default.
- Reuse manual network configs - if on, an attempt is made to restore network configuration during bootup. Netconf remembers 10 most recent manual configurations along with network cards MAC addresses. If cards MAC address matches one of the remembered ones, assigned configuration is used. On by default.
- Automatically setup PPP - switches automatic attempt to create modem connection during bootup on and off. Its made with the first config on PPP configs list. On by default.
- Reuse modem configuration - if on, modem detection is not performed during bootup - instead most recently used modem configuration is used. Off by default.
- Pulse dialing - switches dialing mode used for modem connections between tone dialing (DTMF) and pulse dialing. Default is off (that is tone dialing).
- Automount disk partitions - switches on and off automatical disk partitions mounting (ext2 and vfat filesystems) and swap partition activation during bootup. On by default.
- Add swap file if low memory - switches on and off automatical swap file creation during bootup. Swap file is created if, and only if, theres less than 16 MB memory available (including potentially activated swap partitions) and theres a disk partition on which it could be created available. On by default.
- Automount NFS /usr - switches on and off /usr filestem mounting via NFS during bootup. NFS path to the filestem must be set using "Mount /usr filesystem via NFS" command in main menu. Off by default.
Manual network card configuration
You can enter network configuration parameters in this window:
- This machines IP - enter IP number for this computer here
- Network mask - enter netmask here. If omitted, mask will be calculated based on IP (which will not necessarily be right).
- Broadcast address - enter network broadcast address here. If omitted broadcast address will be calculated based on IP (not mask! - which will not necessarily be right).
- Default gateway - enter IP address of default network gateway (router) here.
- Nameserver IP(s) - enter one or more (separated by spaces) name servers IP addresses here. Can be omitted, but then domain names couldnt be used.
- Default domain(s) - you can enter one or more (separated by spaces) domain names to search host if incomplete domain names will be used. Its optional.
PPP configuration
You can enter modem connection configuration parameters in this window:
- Config name - config name (used in existing configuration selection menu).
- Phone number - phone number to dial (eg. 0w202122 for TPSA access modems).
- Username - username to send to remote server
- Password - password to send to remote server
- Nameserver IP(s) - enter one or more (separated by spaces) name servers IP addresses here. Can be omitted, but then domain names couldnt be used.
- Default domain(s) - you can enter one or more (separated by spaces) domain names to search host if incomplete domain names will be used. Its optional.
Because of permanent configuration that is kept on the floppy you should remember to:
- dont write protect the floppy
- dont remove the floppy from the drive (at least during network configuration)
Enhancements:
- bugfixes in netconf reuse code
- disk partitions automounting, swap partitions autoactivating
- automatic swap file creation
- extended support for NFS mountable /usr
- PS/2 mouse support
- new startup logo
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Download (1.44MB)
Added: 2005-05-11 License: GPL (GNU General Public License) Price:
1640 downloads
Intelligent Wardialer 0.07

Intelligent Wardialer 0.07


Intelligent Wardialer is a war dialer used for auditing your PSTN (phone) network. more>>
Intelligent Wardialer is a "war dialer" used for auditing your PSTN (phone) network. Its features include random/sequential dialing, Voice over IP using the IAX2 (Intra-Asterisk eXchange) protocol, ASCII flat file and MySQL logging, a curses-based front end, key stroke marking, multiple modem support, several methods of "tone detection", save/load state, banner detections (to determine remote system types) and blacklist support.
Main features:
- Full and Normal logging: Full logging records all possible events during dialing (busy signals, no answers, carriers, etc). By default it only records things that we might find interesting (carriers, possible telco equipment).
- ASCII flat file and MySQL logging: You can log to a traditional ASCII flat file, and record information into a MySQL database.
- Dials randomly or sequentially.
- Remote system identification: When finding a remote modem and connecting, iWar will remain connected and attempt to identify the remote system type.
- Key stroke marking: When actively "listening" to iWar work, if you hear something interesting, you can manually "mark" it by hitting a key. You can also enter a "note" about something you find interesting.
- Multiple modem support, because... well, hey - this is "Unix". iWar will support as many modems you can hook up
- Nice "curses" based display. This means that if youre using iWar from a Linux console or a VT100 based terminal, it should work fine. Its not a escape sequence kludge, but true "curses".
- Full control over the modem: Unlike other kludges, iWar doesnt just open the modem as a typical "file". It controls the baud rate, parity, and CTS/RTS (Hardware flow control) DTR (Data terminal ready). This is important for controlling the modem and making it preform the way you want it to during scanning. For example, DTR hang ups.
- Blacklisted phone number support: For numbers the system should never dial.
- Save state: If within the middle of a "wardialing" session you want to quit, you can save the current state to a file. This allows you to come back later and restart iWar where you left off. (via the -l option)
- Load pre-generated numbers: You can load a file (via the -L option) of numbers that you want to dial. This is useful if you want to load numbers generated by another routine (perl/shell script/etc).
- Tone location, if your modem supports it. iWar uses two different methods. The traditional "ATDT5551212w;" (Toneloc) and "silence" detection.
- Records remote system banners on connection for later review
- iWar can be used to attack PBXs and Voice mail systems
- Terminal window so you can watch modem interactions and carrier results in real time
- Support the IAX2 (Intra-Asterisk eXchange) "Voice over IP" (VoIP) protocol. This allows you to scan without the need of additional hardware! To my knowledge, iWar is the first war dialer with VoIP functionality
- In IAX2 mode, iWar acts as a "full blown" VoIP client. In this mode, key 0-9, * and # play there DTMF equivalents. In this mode, you can also directly "talk" (using a microphone) with the remote target if so desired.
- In IAX2 mode, if your VoIP provider supports it, you can "set" your caller ID number (caller ID spoofing).
- Comes with complete source code and is released under the GNU General Public License.
Enhancements:
- Major bugfixes were made for BSD type systems.
- Some other minor bugs were also fixed. VoIP IAX2 (Intra-Asterisk eXchange) support was added.
- With this, you can scan with no additional hardware (such as an analog modem).
- Instead, calls are placed over the Internet.
- DTMF support for when under IAX2 mode was added.
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Added: 2006-01-16 License: GPL (GNU General Public License) Price:
1399 downloads
POE::Filter::FSSocket 0.07

POE::Filter::FSSocket 0.07


POE::Filter::FSSocket is a POE filter that parses FreeSWITCH events into hashes. more>>
POE::Filter::FSSocket is a POE filter that parses FreeSWITCH events into hashes.

SYNOPSIS

#!/usr/bin/perl

use warnings;
use strict;

use POE qw(Component::Client::TCP Filter::FSSocket);
use Data::Dumper;

POE::Component::Client::TCP->new(
RemoteAddress => 127.0.0.1,
RemotePort => 8021,
ServerInput => &handle_server_input,
Filter => POE::Filter::FSSocket,
);

POE::Kernel->run();
exit;

my $auth_sent = 0;
my $password = "ClueCon";

sub handle_server_input {
my ($heap,$input) = @_[HEAP,ARG0];

print Dumper $input;


if($input->{Content-Type} eq "auth/request") {
$auth_sent = 1;
$heap->{server}->put("auth $password");
} elsif ($input->{Content-Type} eq "command/reply") {
if($auth_sent == 1) {
$auth_sent = -1;

#do post auth stuff
$heap->{server}->put("events plain all");
}
}
}

POE::Filter::FSSocket parses output from FreeSWITCH into hashes. FreeSWITCH events have a very wide range of keys, the only consistant one being Content-Type. The keys are dependant on the type of events. You must use the plain event type as that is what the filter knows how to parse. You can ask for as many event types as you like or all for everything. You specify a list of event types by putting spaces between them ex: "events plain api log talk"
Currently known event types (Event-Name):

CUSTOM
CHANNEL_CREATE
CHANNEL_DESTROY
CHANNEL_STATE
CHANNEL_ANSWER
CHANNEL_HANGUP
CHANNEL_EXECUTE
CHANNEL_BRIDGE
CHANNEL_UNBRIDGE
CHANNEL_PROGRESS
CHANNEL_OUTGOING
CHANNEL_PARK
CHANNEL_UNPARK
API
LOG
INBOUND_CHAN
OUTBOUND_CHAN
STARTUP
SHUTDOWN
PUBLISH
UNPUBLISH
TALK
NOTALK
SESSION_CRASH
MODULE_LOAD
DTMF
MESSAGE
CODEC
BACKGROUND_JOB
ALL

Currently handled FreeSWITCH messages (Content-Type):

auth/request
command/response
text/event-plain
api/response (data in __DATA__ variable)
log/data (data in __DATA__ variable)POE::Filter::FSSocket is a POE filter that parses FreeSWITCH events into hashes.

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Download (0.007MB)
Added: 2007-04-18 License: Perl Artistic License Price:
919 downloads
Cornfed SIP User Agent 1.1.4

Cornfed SIP User Agent 1.1.4


Cornfed SIP User Agent is a SIP Softphone. more>>
Cornfed SIP User Agent is a Session Initiation Protocol (SIP) based softphone for your IBM-compatible Personal Computer running the Linux operating system.
The Cornfed SIP User Agent allows you to make Internet phone calls using an Advanced Linux Sound Architecture (ALSA) or Open Sound System (OSS) sound card with speakers and microphone as your telephone handset.
Main features:
- Supports SIP (RFC 3261), SDP (RFC 2327), and RTP (RFCs 3550 and 3551)
- Automated detection of Residential Gateways using Network Address Translation (NAT)
- Supports Digest authentications for registrations and outbound INVITEs
- Support for loose proxy routing using Record-Route and Route headers
- Handles forking of outbound INVITEs by proxies
- Supports re-INVITEs for changes to media transport
- Supports G.711 mu-Law and a-Law voice codecs
- Supports RFC 2833 DTMF tone generation
- Supports SIP compact header forms
- Gnome GUI and CLI clients
- Multi-threaded implementation
The Cornfed SIP User Agent is provided free of charge for personal use for users of the Linux operating system. The program is provided as a binary distribution only. The Cornfed SIP User Agent is specifically designed with embedded and mobile wireless devices in mind.
Development is under way to bring this client to other platforms. If you are interested in licensing this technology for your commercial application, please contact Cornfed Systems at 410-404-8790.
Enhancements:
- Minor bug fixes.
- Changed readline functionality to used raw input rather than cooked mode.
- Fixed bugs with display of status messages that caused client to hang.
- Disabled removal of registration on client exit.
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Added: 2007-07-03 License: Free for non-commercial use Price:
529 downloads
WANPIPE 2.3.4-13 / 3.1.3 Beta

WANPIPE 2.3.4-13 / 3.1.3 Beta


WANPIPE S-series is a family of intelligent multi-protocol WAN and ADSL adapters that support data transfer rates up to 8Mbps. more>>
WANPIPE S-series is a family of intelligent multi-protocol WAN and ADSL adapters that support data transfer rates up to 8Mbps.
All WAN protocols supported by WANPIPE are implemented in firmware and run on the card. An advantage of an intelligent adapter is that it offloads the system CPU and improves stability.
By adding a Sangoma WAN/ADSL component to the Linux kernel, one can create a powerful multi-T1/ADSL router/firewall with proven reliability of Linux. Sangoma S-series cards support an optional on board T1/E1 CSU/DSU that eliminates all external components of a traditional routing solution: i.e. T1/E1 line can be directly connected to the card.
WANPIPE supports the following protocols, ATM, ADSL, Frame Relay, PPP, MULTILINK PPP, CHDLC, X25(API), BitStreaming (API), BiSync(API), and SDLC(API).
Furthermore, WANPIPE supports custom API development such as: Credit card verification, Voice-over IP, Satellite Comm. All device drivers are part of the standard Linux Kernel distribution.
Whats New in 3.1.3 Beta Development Release:
- Drivers can now work with 1, 2, 5, and 10ms chunk sizes.
- Numerous major bugs have been fixed, including issues with echo cancellation, DTMF synchronization, E1/CRC4 mode, and startup.
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Added: 2007-08-11 License: GPL (GNU General Public License) Price:
807 downloads
Verona 0.9.3

Verona 0.9.3


Verona is a portable VOIP toolkit based on eXosip, oSIP, oRTP, curl, and ffmpeg. more>>
Verona is a portable VOIP toolkit based on oRTP, oSIP, eXosip, curl, and ffmpeg.
The toolkit includes a phApi module, which is a high level VOIP API that allows a developer to create voice/video over IP applications in a dozen lines of code.
It includes an example of a minimalistic SIP user agent that supports audio and video transmission. The toolkit is basically the VOIP engine used by the openwengo project.
Enhancements:
- This release optimizes and simplifies the build system.
- It adds support to cross-compile for ARM Linux.
- It adds support for transmission of DTMF in the SIP INFO packet.
- It is compatible with the latest version of Intels IPP.
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Added: 2006-10-25 License: GPL (GNU General Public License) Price:
1095 downloads
VOCP 0.9.3

VOCP 0.9.3


OCP is a complete messaging solution for voice modems, with voicemail, fax, email pager, DTMF command shell. more>>
Much more than an answering machine, VOCP transforms your computer into a full-featured call answering and voice messaging system. With the VOCP System, you can create an unlimited number of voicemail, pager and command shell boxes which callers will navigate using their touch tone telephone.
You can send and receive faxes, listen to your email using text-to-speech, filter and redirect calls based on caller ID information, run programs through the telephone and more.
VOCP acts as a call answering and voice messaging system. When a call comes in on the line connected to the voice modem, VOCP answers and the caller is placed in the root box. From there, the caller may navigate the system tree by using the DTMF keys on the telephone.
Most boxes will play messages that present menus which lead to other boxes, which the caller selects by entering a single digit on the telephone keypad. Callers may jump to any box withing the system by entering *, followed by the box number and, optionally, followed by the # key.
The leaf nodes, or terminal points, in the system end the call - this is where the caller will leave a voice or pager message, send or receive a fax.
A user may also choose to log into the system, in order to retreive voicemail messages or to run programs using the command shells. This is accomplished by accessing the special box number 999 (by entering *999#).
Callers navigate the system using a touch-tone phone and may send and receive faxes, voice mail, and pager messages, listen to text/HTML email messages, or execute configured programs on the host and hear the resulting output.
Main features:
- The core VOCP voice messaging system
- The VOCP Call Center
- The XVOCP graphical message retrieval interface
- The VOCPhax graphical faxing interface
- The VOCPweb remote web interface
- The VOCP BoxConf configuration interface
- a large number of utility programs (see the /usr/local/vocp/bin/README for details)
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Added: 2005-04-20 License: GPL (GNU General Public License) Price:
1648 downloads
SvxLink 070415

SvxLink 070415


The SvxLink project aims to develop a flexible general purpose voice services system for ham radio use. more>>
The SvxLink project aim to develop a flexible general purpose voice services system for ham radio use. The svxlink server consists of a core that handles the connection to the tranceiver.
The transceiver audio is connected to the PC through the sound card and the PTT is controlled by a pin in the serial port. The core can be configured to act as a repeater controller or to operate on a simplex channel.
The voice services are loaded into the core as plugins called modules in SvxLink lingo. Existing voice services are: Help - a help system, Parrot - a module that plays back everything you say, EchoLink - connect to other EchoLink stations and TclVoiceMail - a simple voice mail system. The project also includes an EchoLink client GUI application (Qtel).
EchoLink is an amateur radio invention (well actually it is just a modified verison of IP telephony) to link radio transceivers together over the Internet. You must have an amateur radio license to use it. The original EchoLink software can be found at http://www.echolink.org/.
However, this software only support the Windows operating system and it is closed source. SvxLink is FREE software released under the GPL license.
Qtel is only an EchoLink client program. It does not have the sysop mode. That is, it can not be connected to a transceiver and act as a link. For the latter, use the svxlink server.
Enhancements:
- Support for remote receivers linked over the Internet has been added.
- To be able to use multiple receivers, a simple signal strength detector was implemented.
- CTCSS tone transmit has been implemented and the CTCSS squelch detector has been improved, which makes it possible to run SvxLink with the squelch open.
- DTMF muting and squelch tail elimination has been added, and there are a couple of other changes and bugfixes.
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Added: 2007-04-15 License: GPL (GNU General Public License) Price:
925 downloads
Linphone 1.7.1

Linphone 1.7.1


Linphone is a Web phone with a GNOME interface. more>>
Linphone is a web phone that it let you phone to your friends anywhere in the whole world, freely, simply by using the internet. The cost of the phone call is the cost that you spend connected to the internet.
To call somebody, you must provide to linphone a SIP URL : It is something like toto@machine.com, where toto is a linux user that runs linphone, and machine.com is the name of a host on a network. If you dont know the machines name you can specify simply an IP address in dot notation (as 192.0.0.1)
Linphone is mostly sip compliant. It works successfully with these implementations:
- eStara softphone (commercial software for windows)
- Pingtel phones (with DNS enabled and VLAN QOS support disabled).
- Hotsip, a free of charge phone for Windows.
- Vocal, an open source SIP stack from Vovida that includes a SIP proxy that works with linphone since version 0.7.1.
- Siproxd is a free sip proxy being developped by Thomas Ries because he would like to have linphone working behind his firewall. Siproxd is simple to setup and works perfectly with linphone.
- Partysip aims at being a generic and fully functionnal SIP proxy. Visit the web page for more details on its functionalities.
Linphone may work also with other sip phones, but this has not been tested yet.
Linphone uses the SIP protocol to establish calls, for that reason it cannot work with H323 phones, because SIP and H323 are different and opposite protocols. H323 phones are Netmeeting (for windows), Gnome-meeting (Unix), OpenPhone...
Main features:
- Works with the Gnome Desktop under linux, (maybe some others Unixes, but this has never been tested). Nevertheless you can use linphone under KDE, of course !
- Since version 0.9.0, linphone can be compiled and used without gnome, in console mode, by using the program called "linphonec"
- Works as simply as a cellular phone. Two buttons, no more.
- Linphones includes a large variety of codecs (G711-ulaw, G711-alaw, LPC10-15, GSM, and SPEEX). Thanks to the Speex codec it is able to provide high quality talks even with slow internet connections, like 28k modems.
- Understands the SIP protocol. SIP is a standardised protocol from the IETF (http://www.ietf.org), that is the organisation that made most of the protocols used in the internet. This guaranties compatibility with most SIP - compatible web phones.
- You just require a soundcard to use linphone.
- Other technical functionnalities include DTMF (dial tones) support though RFC2833 and ENUM support (to use SIP numbers instead of SIP addresses).
- Linphone is free software, released under the General Public Licence.
- Linphone is documented: there is a complete user manual readable from the application that explains you all you need to know.
- Linphone includes a sip test server called "sipomatic" that automatically answers to calls by playing a pre-recorded message.
Enhancements:
- This version fixes a compilation error, an incorrect icon path and updates the cz translation.
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Added: 2007-04-18 License: GPL (GNU General Public License) Price:
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