搜索资源列表
asterisk-1.2.0.tar
- 这个是Asterisk的源码。是一个IP PBX开源的项目，适合做IP电话等业务。支持sip，h323，mgcp等协议。-this is the source Asterisk. IP PBX is an open-source project, suitable for IP telephony services. Support sip, h323, Controller and other agreements.
asterisk.tar
- 一个非常美妙的proxy。功能强大。基于sip的协议。如果还要的话，我这里还有一个sip的协议测试工具。-a very wonderful proxy. Powerful. Based on the agreement sip. If they have, then I have here a sip of protocol test tools.
asterisk-sounds-1.0.1.tar
- linux的PBX源代码（sip）-PBX source code (sip)
asterisk-addons-1.0.1.tar
- linux下的PBX源码（SIP）-the source PBX (SIP)
asterisk
- linux下面,sip,h.323代理服务器c语言原码,要在linux环境下安装,可以构建voip系统服务器
csharpSip.rar
- c#sip proxy，绝对的100％的源代码，非常难找，asterisk的windows编程必备啊,c# sip proxy, absolute 100 of the source code, very difficult to find, asterisk of windows programming must ah
sip
- SIP协议讲座-Asterisk.ppt -Lecture-Asterisk.pptSIP SIP protocol agreement talks-Asterisk.ppt
sipPhone
- 基于sip的voip网络电话的客户端程序，服务器可用开源平台asterisk，实现了单呼、组呼、强插、强拆等功能。-The voip sip based VoIP client, the server is available open-source platform asterisk, to achieve a single call, group call, Override, demolitions and other functions.
asterisk-core-sounds-en-g729-current.tar
- SIP code oSIP协议栈(及eXoSIP,Ortp等)使用入门-SIP code oSIP protocol stack (and eXoSIP, Ortp etc.) Getting Started
fusionpbx-2.0.7.tar
- pbx based on open source voip application asterisk -- SIP
asterisk-1.8.3.2.tar
- 最新的asterisk源码，pbx服务器，sip网络电话服务器，可实现模拟电话和网络电话互通-Asterisk source, pbx server, sip network telephony server, analog phones and VoIP interoperability
obelisk-1.0.0
- This project contains a SIP stack and server applications built on top of the stack, examples of which are: Stateless Proxy, Registrar,NAT Keepalives,MWI Notifier server. The server applications have been designed to work in association with Asterisk
freepbx-2.4.0
- freepbx asterisk SIP
SIP-Asterisk
- 如何将sip网关注册到asterisk服务器简要介绍。-How to register to the asterisk server sip gateway briefly.
asterisk.tar
- asterisk 代码，构建自己的sip电话-asterisk source
asterisk-1.2.35.tar
- asterisk-linux下的安装文件-asterisk-sip voip
YouToonew9
- 基于sip协议，用asterisk作为服务器，实现voip软电话功能，实现了接听，呼叫，挂机，呼叫转移，以及通信录等功能。-Based on sip protocol, with the asterisk as a server, voip soft phone capabilities to achieve and realize the answer, call, hang up, call forwarding, and address book functions.
asterisk-0.1.0.tar
- 一个基于sip协议的桌面ip-pbx,配上语音卡，使你的pc轻松成为pbx-A desktop based on sip protocol ip-pbx, coupled with voice cards, make your pc easily become pbx
sip-enc_2009-07-28.tar
- asterisk 1.4.21.2的sip信令和rtp流加密文件，放入对应目录替换掉原来的文件，然后编译即可，加密端口为5064，绝对防封杀。-asterisk 1.4.21.2 sip_chan and rtp encode
chrome浏览器网页版SIP(290024)
- chrome 浏览器 网页SIP WEBRTC 开发 呼叫中心需要的工具 开发网页软电话 必备(chrome web sip webrtc asterisk callcenter)