freepbx sip

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FreePBX SIP Trunk

FreePBX SIP TrunkDockingbackground: PBX1 is a virtual machine running FreePBX, whichnow needs to be connected via SIP TRUNK docking , PBX2, using PBX2 E1 The line calls out the phone. PBX1 192.168.100.1PBX2 192.168.100.2PBX1on the configurationOneConfigurationTrunkNew SIP

CentOS 5.8 asterisk-1.8.10.1 Installation II: Installation of FreePBX

/sbin/amportal start" >>/etc/rc.local# Chkconfig httpd on# Chkconfig Mysqld onOK Installation Complete # rebootchmod 777/var/www/html-rcd/var/lib/asterisk/chmod 777 BinChmod-r 777 bin/*chmod 777/var/lib/php/session/chmod 777-r/var/lib/php/session/*http://192.168.100.128 Default User name password is admin/admin First login admin interface, need apply configuration ============================================================Now you can create the SIP n

Freepbx configure asterisk video call

You can configure a SIP Phone as needed, but video functions cannot be implemented in the early stage, Now the video function has been successfully configured. Please share with us: The premise is to create an account through extensions and use X-lite as the client: 1. log on to freepbx. The default value is admin/admin. Click module admin and check for update online. 2. Click Check update and you w

FreePBX system recording menu Arbitrary File Upload Vulnerability

FreePBX system recording menu Arbitrary File Upload Vulnerability Release date: 2010-09-23Updated on: 2010-09-25 Affected Systems:FreePBX 2.8.0Description:--------------------------------------------------------------------------------Bugtraq id: 43454Cve id: CVE-2010-3490 Previously called Asterisk Management Portal, FreePBX is a standardized implementation of the IP telephone tool Asterisk and provides We

FreePBX 'usersnum' Parameter Remote Command Execution Vulnerability

Release date:Updated on: Affected Systems:FreePBX 2.xDescription:--------------------------------------------------------------------------------Bugtraq id: 65756 FreePBX is an open source Web PBX solution. FreePBX 2.x and other versions have the remote command execution vulnerability. Attackers can exploit this vulnerability to execute arbitrary commands in the context of the affected application. *> Test

Ubuntu freepbx-2.11.0.40 Installation

about The installation of FreePBX, I do not want to say anything, the internet piracy is rubbish, or look at the official bar, the harm I did not know the day of the problem. In the end I do not know where I was wrong, may be a privilege problem, there may be other reasons. Asterisk installation should be no problem, I installed less than three times, but this installation, in the./configure when the man ignored a warning, thought an overall picture,

Instances for communication between SIP and IAX Intranet and Internet and PSTN lines and mobile phones

= YesHidecallerid = NoCallwaiting = YesUsecallingpres = YesCallwaitingcallerid = YesThreewaycalling = YesTransfer = YesCancallforward = YesCallreturn = YesEchocancel = YesEchocancelwhenbridged = NoEchotraining = 800Rxgain = 0.0TXT gain = 0.0Group = 0Channel = 1Callgroup = 1Pickupgroup = 1Immediate = NoFaxdetect = incomingGroup = 1Channel = 2By now, the configuration file must be manually modified! Iv. Call TestThe following operations are all completed by the

[Sip] Sip series standard navigation announcement board

This page is maintained by Paul E. Jones. Core sip documents RFC 2543 SIP: Session Initiation Protocol (obsolete) RFC 3261 SIP: Session Initiation Protocol SDP Related Documents RFC 2327 Session Description Protocol (SDP) RFC 3264 An offer/Answer Model with

SIP traversal NAT SIP Traversal Firewall

Firewallnat In fact, the essence of the light-mouth board is equivalent to NAT device. To achieve NAT penetration, there is no problem with the SIP processing of the optical orifice plate. Firewall is a passive network security defense technology, located at the boundary of the network, executing the access control policy between the two networks, preventing the external network from illegally accessing the internal information resources, or preventi

Analysis of SIP route and record_route/SIP routing mechanisms

Analysis of SIP route and record_route/SIP Routing Mechanism from: [url] Routing Analysis of SIP Routing Mechanism (zz)We have already introduced sip-related knowledge about important SIP header domains, registration processes, and session processes. Now we will introduce t

Sipana a distributed sip analyzer (analysis) Open-Source SIP protocol analysis tool

This tool is a real-time analysis of the SIP Communication Protocol, which is the main function of the Communication Engineering of zhongzheng University. In addition, the distributed design can be used to analyze cross-domain sip packets, draw a complete signaling flowchart, and control the web interface... background in view of the fact that there is no free communication protocol analysis software for th

Network integration of video surveillance services based on Ngn/ims (SIP Protocol) communication Platform

I. Overview Review the video surveillance market, the monitoring platform system used by the signaling control protocol is not unified, resulting in the interconnection of the technical threshold. This is a bit similar to the 80 's China's switch market eight system of the situation. These reasons hinder the network era of large-scale carrier-level video surveillance, remote access, centralized management needs. The industry has explored two ways to solve the problem: From the monitoring industr

SIP (Session Initiation Protocol) Conversation Initiation Protocol

SIP (Session Initiation Protocol) Conversation Initiation Protocol is an application layer control (signaling) protocol for network telephony and conferencing. It is mainly a multimedia communication protocol based on IP network. It can realize the signaling functions are also using RTP as a media transmission protocol. Originally presented by the IETF Mmusic (multiparty multimedia session control) workgroup. SIP

Understanding sip-the most popular communication protocol today

Address: http://www.sipcenter.com/sip.nsf/html/Chinese+SIP+Overview Introduction Next-generation services Historical Review Advantages of SIP: Web-like scalable open communication SIP session Composition Introduction Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now we have

SIP's most popular communication protocols are mature

Introduction Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now we have the SIP Session Initiation Protocol. SIP was born less than a decade ago in the computer science laboratory. It is the first protocol suitable for multi-user sessions in various media content. Now it has become the specification of the Internet Engineering Task Gro

Learning notes-SIP message receiving and processing flow based on sipdroid SIP audio and video telephony

/success (actual Sipprovider callback), the Fail/sucess interface of the transactionclient is called Public classTransactionclientextendsTransaction {Transactionclientlistener transaction_listener; Publictransactionclient (Sipprovider sip_provider, Message req, Transactionclientlistener listener) {Super(Sip_provider); Request=NewMessage (req); Init (Listener, Request.gettransactionid ()); //This.transaction_listener = listener;} //The actual Sipprovider callback Public voidOnreceivedmessage (Si

SIP protocol (Chinese)-1

1. Introduction to the SIP protocol Many applications on the internet need to establish and manage a session. The session here refers to the exchange of data between participants. Considering the actual situation of participants, the implementation of these applications is often very complicated: Participants may move between agents, they may have multiple names, the communication between them may be based on different media (such as text, multimedia,

Basic SIP application solution

Basic SIP Application As one of the main VOIP communication protocols, the SIP protocol is simple, flexible, and open, and is gradually dominant in the VOIP communication field. The main methods used for SIP Communication include SIP terminals, proxy/targeted servers, location servers, and PSTN gateways. Currently, th

[SIP Protocol] learning notes for beginners

Author: gnuhpcSource: http://www.cnblogs.com/gnuhpc/ 1. How is the SIP coming and how is it constructed? In general, SIP is a lightweight signaling protocol, which can be used as a signaling for audio, video, and timely information. Speaking of how the SIP is coming out, we need to mention H.323, and the standard has to mention ITU-T, we will first talk about t

"Go" SIP preliminary

1 , what is SIPSIP (Session Initiation Protocol) belongs to the IP Application layer protocol, which is used to provide the user with session application on the IP network. A session refers to the communication of voice, video, and other media forms between two or more users, possibly IP telephony, conferencing, instant messaging, and so on.SIP is a signaling protocol that corresponds to a call signaling protocol (such as SS7 ISUP) in a traditional telephony network. Building a complete multimed

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