VoIP Blog - Page 812
Asterisk AGI MacinTalk
Asterisk AGI MacinTalk I just wrote an AGI that lets you call MacinTalk as follows exten => exten,pri,agi,macintalk|The text you want spoken goes here Put the following into /var/lib/asterisk/agi-bin/macintalk #!/bin/bash trap ‘rm /tmp/agi$$_*; exit’ INT TERM HUP #declare…
Read More »Festival
From the web site: Festival is a general multi-lingual speech synthesis system developed at CSTR. It offers a full text to speech system with various APIs, as well an environment for development and research of speech synthesis techniques….
Read More »Sphinx
Sphinx Description From Sourceforge: Sphinx is a speaker-independent large vocabulary continuous speech recognizer under Berkeley’s style license. It is also a collection of open source tools and resources that allows researchers and developers to build speech recognition system….
Read More »CDRTool
CDRTool https://www.ag-projects.com/CDRTool.html Evaluation and commercial licenses available. CDRTool is an Operational System Support (OSS) for service providers. CDRTool allows real time web access to Call Detail Records generated by VoIP switches, gateways or network access equipment using RADIUS…
Read More »D-Link DVG-1402s
Dlink webpage for DVG-1402s 2 FXS port VoIP gateway (NAT/firewall/QoS, 4 LAN ports) Product Features: “¢ Built-in Firewall Protects Your Network “¢ Connect up to Two Standard Telephones for Low-Cost Calling “¢ Compatible with Multiple Call Features Including…
Read More »get data
Usage: GET DATA <file to be streamed> [timeout] [max digits] Stream the given file, and recieve DTMF data. This is similar to stream file, but this command can accept and return many DTMF digits, while stream file returns…
Read More »Asterisk tips openclose.agi
How to play different greetings based on time and date Lets play a nighttime or daytime greeting and then continue in the current context. Place this in your agi-bin directory (usually /var/lib/asterisk/agi-bin): !/bin/sh Done by Alex Lopez (alex.lopez…
Read More »SIPcall
UK VOIP to/from PSTN service From website SIPCall enables SIP users to make and receive telephone calls to and from the telephone network (PSTN) irrespective of their geographical location. SIPCall not only offers call charge free IP to…
Read More »Asterisk indications BE
Asterisk indications for Belgium Insert this into indications.conf and set the Asterisk cmd SetLanguage to be. [be] description = Belgium ringcadance = 1000,3000 dial = 425 busy = 425/500,0/500 ring = 425/1000,0/3000 congestion = 425/167,0/167 callwait = 1400/175,0/175,1400/175,0/3500…
Read More »Voxeo
Voxeo Improves customer service and lowers communications costs by automating and connecting common enterprise phone calls with Interactive Voice Response IVR or Voice over IP (VOIP) solutions. Voxeo’s VOIP and IVR hosting, turnkey platforms, and developer services –…
Read More »OpenSER
Page Contents About OpenSER OpenSER Headlines Headlines Archive OpenSER Events 2008 Events Archive OpenSER Capabilities Download Documentation Tutorials Other Resources OpenSER Modules v1.0.0 v1.1.0 v1.2.0 v1.3.0 v1.4.0 Deploying OpenSER Practical Examples Platforms Resources See also About OpenSER Please…
Read More »About OpenSER
Please note that OpenSER no longer exists. It was forked into two projects, Kamailio and OpenSIPS. The two projects are currently very similar, given that they are built on the same code base, however the two projects have…
Read More »Asterisk perl library
The Asterisk PERL Library helps you in developing Asterisk AGI applications that support your dialplan Asterisk Manager applications that can help you monitor Asterisk You’ll find the Asterisk perl library here: FreeBSD misc/p5-Asterisk port In the distribution, there…
Read More »Asterisk H324M
Asterisk as 3G-H.324M ? SIP Gateway Update to this page: A new Wiki page is created to collect compatibility information about app_h324m: Asterisk app_h324m compatibility Update to this page: As with 2008-01-16 (and long before), H324M calls work…
Read More »H.263
H.263 is a video codec designed by the ITU-T as a low-bitrate encoding solution for videoconferencing. It was first designed to be utilized in H.324 based systems (PSTN and other circuit-switched network videoconferencing and videotelephony), but has since…
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