freeswitch voip

The VoIP-a reference guide to all things VoIP This wiki covers everything related to VoIP, software, hardware, service providers, reviews, deployments, standards, Tips tricks and everything else related to voice over IP networks, IP telephony and Internet telephony.''Welcome to voip-info.org! Please e let me know at s

News This section is for news, ie news reports, press releases, product release announcements etc. Research: peer-to-peer Internet telephony using SIP PDF Iconv application module for character conversion. Version 0.9.2 of ldapget application module released. bugfix. Over 5 million VoIP subscribers worldwide-dmeurope story Interviews with BKW, twisted and David Mandelstam Interview with drumkilla, the manager of the stable branc

: ', which is CID Type=noneInbound DestinationBy default, incoming calls on this gateway would go to the username.If you want FreeSWITCH to respect the ${sip_to_user}, set the value to "Auto_to_user". Be sure the contextRegistrationIf This gateway was only for outbound calling, then there's rarely a need to maintain a registration ahead of time.VariablesIn addition to the parameters you can optionally set variables to set on either incoming or outgoin

This article comes from CSDN LIDP HTTP://BLOG.CSDN.NET/PERFECTPDL, reprint indicates the source, thanks .  I built a Freeswitch core research Exchange Group, 45211986, Welcome to join, in addition, the provision of SIP based communication server and client solutions, Clients registered to FreeSWITCH can dial each other, but what happens when the client wants to call through the

Compared with asterisk and freeswitch, there are a lot of slots. As for how to choose it, the benevolent sees benevolence, and the wise sees wisdom. In my experience, I think the asterisk configuration is relatively simple (I really hate the xml configuration), but freeswitch is relatively high in sound quality and performance. The performance of asterisk1.8 is much better than before, but I'm disappointed

This article, the original connection: http://blog.csdn.net/freewebsys/article/details/46546205, reprint please indicate the source!1, about FreeSWITCHFreeSWITCH is a soft-switching solution for telephony, including a softphone and soft switch to provide voice and chat product drivers. The FreeSWITCH can be used as a switch engine, PBX, multimedia gateway, and multimedia server.FreeSWITCH supports a variety of communication technology standards, inclu

1. Install dependenciesApt-get install autoconf automake devscripts gawk g ++ git-core 'libjpeg-dev | libjpeg62-turbo-dev 'libncurses5-dev 'libtool-bin | libtool' make python-dev gawk pkg-config libtiff5-dev libperl- dev libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-dev libspeex-dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns-dev libpq-dev./Bootstrap. sh-jVim modules. conf./Configure -- with-python -- enable-core-odbc-support -- enable-core-pgsql-s

Too many VoIP service providers want to sell you their "full solution", from the phone number on your desk, from different sites to the WAN and public exchange Telephone Network (PSTN). However, as I have seen, Unless users and suppliers have full experience and thoroughly checked every detail, the so-called "full set of VoIP systems" will certainly make some mistakes. Enterprises that have trouble with

In the current network communication, the Email service is no longer the preferred communication method. More instant messaging and voice services are emerging on the network. Now let's talk about the technical principles of VoIP for IP phones.Basic transmission process The traditional VoIP telephone network transmits voice in a circuit exchange mode. The required transmission bandwidth is 64 kbit/s. The so

This article is from csdnHttp://blog.csdn.net/voipmakerReprinted to indicate the source. Thank you. I have created a freeswitch learning exchange group, 45211986. welcome to join. Did stands for direct dial-in. It is a concept proposed by the operator. With this number, the operator can call to the VoIP system and configure a gateway by calling freeswitch,

I recently bought a new office building and the telephone system of the new Office, including Softswitch, digital relay gateway, E1 and VoIP, which should be configured by me. Multi-function programmer :) The following are some records during freeswitch configuration. I will share them with you here. For more details, see the official FS and Chinese official website. 1.

Every organization that is considering deploying a VoIP Phone System has heard the same terrible warning: "routing voice calls over the data network will expose the call content to eavesdroppers ".Although in fact, any phone call is at risk of being eavesdropped to some extent, is the VoIP call system itself at a high risk? In this article, we will explore the ins and outs of

This article is from csdn lidp http://blog.csdn.net/perfectpdl. please refer to this Article. Thank you. I have created a freeswitch kernel research and exchange group, 45211986. Welcome to join. In addition, we provide the SIP-based communication server and client solution, Developed for SIP/IMS video clients, supports access to sip Softswitch, IMS core network, andVoice, video, and instant messaging functions. The video formats support h263, h264, a

Official installation Address: Https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6Buy a cloud service from Ali Clouds, install the CentOS 64-bit system, and install the command:Yum Install Lrzsz-y1. Add RPM Source RPM-IVH http://pkgs.repoforge.org/rpmforge-release/rpmforge-release-0.5.3-1.el6.rf.x86_64.rpm or RPM-IVH http://mirror.cedia.org.ec/fedora-epel/6/x86_64/epel-release-6-8.noarch.rpm 2, the insta

Create the/etc/init. d/freeswitch script. The script content is as follows: #! /bin/sh## freeswitch: Starts the freeswitch Daemon## chkconfig: 345 96 02# processname: freeswitch# description: Freeswitch fedora init script # config:# Author: gled# Source function librar

System: centos 6.4 64-bit; Freeswitch version: 1.5 For details about the installation process, refer to the freeswitch wiki on the official website (or refer to my blog "Install freeswitch centos6"). Start with the SQLite problem encountered during freeswitch installation./configure: Checking for sqlite3> = 3.6.20... P

How can I test the VoIP function with an existing PBX or key-press system? There are multiple ways to use the existing PBX system or key-press system to test the VoIP function. How to test the function depends on your purpose. If there are two sites connected with PBX connection lines, but you want to use VoIP so that you can send calls between internal network

From the FreeSWITCH installation process./configure encountered the problem of SQLite start:Checking for sqlite3 >= 3.6.20 ... Package sqlite3 is not found in the Pkg-config search path. Perhaps should add the directory containing ' sqlite3.pc ' to the PKG_CONFIG_PATH environment variable No package ' Sqlit E3′foundConfigure:error:Library requirements (sqlite3 >= 3.6.20) not met; Consider adjusting the PKG_CONFIG_PATH environment variable if your libr

1, added SIP Provider, add the configuration file in Conf/sip_profiles/external This SIP Provider requires REGISTER, and since FreeSWITCH is accessed through NAT, it is set to send pings for 30 seconds. 2, the did mapped by this SIP Provider to the corresponding extension SIP Profile External.xml Sets the context default to public, so we need to edit conf/dialplan/public.xml Note: You can also configure this gateway in your directory, the diffe

At present, the domestic use of open source system development call center mostly uses asterisk system. Asterisk entered earlier, the relevant information and open source code, communication technology personnel use the document can be set up a simple call center. With the development of the FreeSWITCH system by asterisk core technicians, some innate defects in the original asterisk also attract people's attention, such as the disadvantage of call co