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VoIP-info is your go-to website for anything VOIP. This includes VoIP software & hardware, service providers, tips and tricks as well as anything related to voice over IP networks, IP telephony and Internet Telephony. Latest Contributions We love receiving your contributions. If you want to add a page register to create an account and start typing! Upon approval your page will be published and ed
The Asterisk Manager Interface (AMI) allows a client program to connect to an Asterisk instance and issue commands or read events over a TCP/IP stream. Integrators will find this particularly useful when trying to track the state of a telephony client inside Asterisk, and directing that client based on custom (and possibly dynamic) rules. A simple “key: value” line-based protocol is utilized for c
Converting WAV files Asterisk 1.4 In asterisk 1.4 is a conversion application built in, Asterisk file convert. The command converts between different code formats. file convert <file_in.format> <file_out.format> A shell script example: #!/bin/bash # Converts a audio file from alaw to a ulaw rasterisk -x "file convert /tmp/file_in.alaw /tmp/file_out.ulaw" Convert files from the CLI You just recorde
Asterisk and DTMF (TouchTone) As a PBX Asterisk is able to translate the different types of DTMF signalling methods (audio, RFC2833, SIP INFO, IAX2). Variable length DTMF This is a tricky issue: You cannot expect to have control over the duration of DTMF signals when using SIP INFO in general, or RFC2833 before Asterisk 1.4.0: For example cell phone carriers transmit keys pressed on your GSM phone
STUN (Simple Traversal of UDP through NATs (Network Address Translation)) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as ‘Session Traversal Utilities for NAT’. Note: The STUN RFC states: This protocol is not a cure-all for the problems associated with NAT. STUN enables a device to find out its public IP address and
Asterisk an open source framework for building communications applications. It runs on Linux, BSD and OS X and allows you to build a PBX given sufficient Linux and telephony know how. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call
How fast/big must my machine be in order to serve my needs? The only rule of thumb we appear to be able to provide is this: Asterisk 1.2 start to run into problems around 220 concurrent SIP calls. Asterisk 1.4 scales much better and can handle nearly double the call setups/second as well as total concurrent traffic. Moreover, Early testing of Asterisk 1.6 using hash tables shows a SIP performance
When Asterisk starts an AGI script, it feeds the channel variables to the script on standard input. The variable names are prefixed with “agi_” and are separated from their values by a colon and a space. Though the actual channel variables may be in the upper case, the names passed to an AGI script are all lower case. Global variables are not passed to the AGI script in this manner. You must get t
Open Source VOIP applications, both clients and servers. SIP Proxies Sip I/O Lightweight sip proxy, location server, and registrar SBO SIP Proxy Bypass All types of Internet Firewall JAIN-SIP Proxy Mini-SIP-Proxy A very tiny perl POE based SIP proxy MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack MySIPSwitch SIP Proxy server which allows using multiple S
Created by: oej,Last modification on Sun 14 of Sep, 2008 [09:36 UTC] by haamed Open Source VOIP applications, both clients and servers. Open source means all source code is available!! Do not post any "free but not open" software here! SIP Proxies Net-SIP A Perl SIP framework that includes a stateless proxy sipd SIP Proxy SIP Express Router (SER): the SIP router/proxy/jack-in-all-trades from IP
Created by: system,Last modification on Fri 26 of Sep, 2008 [22:52 UTC] by spamblock Welcome to the VOIP Wiki - a reference guide to all things VOIP This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is prim
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