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Share Asterisk realization of the analog ring tone form call

Simply speaking, the ring is called when the calling call, the main call will hear a piece of music or voice. Instead of a normal ringing tone. In the Asterisk dial () in the addition of lowercase m can be set to achieve playback of voice, that is, the realization of ring tones. The following example: Copy Code Exten = _x.,1,dial (sip/${exten},,m) so that when the call is called the user, the voice stored in the asterisk can be played, usually stored in the "/var/lib/asterisk/mohmp3/" directory.

FreeSwitch SIP (1): Linux under compile and install v1.4

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

Cisco uc-feature -01-phone-unplug the-cisco IP phone

, some people say that using a network cable is not slow, other can be all the phone to a VLAN, all the phones are connected to the back end of the Poe switch, in fact, the power supply is also provided by the switch, you only connectThe second type: use his own company's IP phone softphone as shown in. 650) this.width=650; "height=" 676 "title=" clip_image003 "style=" margin:0px;border:0px;padding-top:0px; Padding-right:0px;padding-left:0px;backgroun

sip:180 Ringing vs 183 Session Progress

If You know this phone is ringing (an ALERT q.931 message, for instance) you send a 180 ringing. If You receive a notification indicating this call is progressing, but you don't know for sure whether the user I s being alerted or not, your send a 183 session Progress message. Both can indicate early media with SDP. If There is no SDP, the end device (softphone/gateway/etc.) has to generate the ringback tone or progress tone. Usually you'll be 180 with

IP Centrex Service Based on Softswitch

multimedia software, IAD is used to provide users with POTS ports and Data ports. For branches with fewer employees, you can use multi-port IAD devices to provide users with POTS ports and Data ports for simultaneous access of multiple telephones. For institutions of a certain scale, the Access Gateway AG is used to provide hundreds of lines of telephone access. For new users, the new access mode under the softswitch architecture is used to access the service through soft terminals, IAD, IP pho

API Monitor---------------Using API Monitor to crack copy protected software

For this tutorial we'll be the using mirial softphone which is a HD video conferencing application. This tutorial are for educational purposes only, so please do not use this to create or distribute a cracked copy of the so Ftware.When you install the application, it prompts a license file. After installing the license, you have the evaluate the application. The expiration date is displayed on this screen; It is March 15, 2011.Step 1Trial applications

VoIP eavesdropping: reinforces network security to prevent VoIP risks

network interface box, which is sometimes located outside the house or office, then, a listening device is placed on the box to continuously monitor the telephone call content.Possible VoIP eavesdropping usually follows the same process, but different tools are used. The first requirement is to be able to access the media that carries voice calls. This may be achieved by intruding into a VoIP Phone, a workstation running a softphone device, or a VoIP

Snom IP Phone Web Interface & amp; lt; v8 multiple defects and repair

because there is no password thats protects the interface. # XSS Vulnerability: # The xss vulnerability found in the section Addres Book of Snom IP Phone software. # The vulnerability allows the attacker to inject javascript code to the field number. # To exploit the vulnerability we need to access to the Snom IP Phone by this url http: // address/adr.htm. # Then we can write any javascript code that we want and send the form. by the next refreshing of the page the javascript code will run. # I

Android Network phone source code

1. sipdroid \ SRC \ org \ zoolu is the implementation of the SIP protocol stack. 2. sipdroid \ SRC \ org \ sipdroid is the implementation of softphone 3. Implementation related to stun in sipdroid \ SRC \ com 4. sipdroid the default encoding format is G711-A rate. 5. The program compiled directly using ant debug supports only the-rate and U-Rate Audio Encoding formats. Other programs must be imported using the ndk method before they can be used. 6. If

Thirteen methods to Ensure VoIP network security

choice. It places the voice service and other servers in a separate domain and restricts access to it. 9. Minimize the use of softphone VoIP soft terminal phones are vulnerable to computer hacking, even after the company's firewall, because they are used together with common PCs, VoIP software, and a pair of headphones. Moreover, soft terminal phones do not separate voice and data, so they are vulnerable to viruses and worms. 10. Conduct regular secu

Avaya Unified Communication product with high quality and low price

reduces the cost of Unified Communication by up to 40%. Enhancing collaboration tools and system capacity helps SME meet challenges With Avaya one-X Portal for IP Office, you can use a browser-based easy-to-use interface to combine VPN and PC/phone) to manage communications in your Office, Home Office, or on the road. Avaya IP Office 6.0 also provides the instant communication function fully integrated with the Embedded Voice call function, and the online status function of displaying idle col

Elastix2.5 & vtigercrm5.2.1 configure incoming call screen and click call

1. Set user name and password permissions for vtiger to access the AMI interaction of elastix PBX PBX> Tools> asterisk file editor Edit manager_custom.conf: (this file does not exist by default)[Vtiger]Secret = vtiger Permit = 0.0.0.0/255.255.255.0. Of course, you can set 127.0.0.1.Read = system, call, log, verbose, command, agent, user, dialplanWrite = system, call, log, verbose, command, agent, user, originate Ii. Restart Asterisk Su-Asterisk-RManager reloadManager show user vtiger Iii. vtig

Analysis of SIP packages

time for mediation to see if there is any abnormal status. when the last party hangs up, the server initiates another mediation and sends a request to the other Party: request: Bye SIP: 015B0081587310007000D0000C578@192.168.2.40 and the other side returns a response: Status: 200 OK the call is complete! All these SIP messages are attached to the UDP packet for transmission. The ethereal packet capture tool shows the status in detail.The following is an example of a request to connect to a

Detailed introduction to the general structure of Softswitch Technology

to users who use telephone services and data services on computers at the same time. In this case, a dedicated "softphone software" needs to be installed on users' computer devices ". Media Servers Access Gateway (MNS) is used to complete information about various multimedia data sources and adapt Multimedia Streams mixed with video and audio to IP packets. H.323 Gateway: used to connect an IP address Telephone Network Gateway using the H.323 protoco

Minisipserver simple and not simple, easy landing, the realization of computer-to-fixed, mobile communication

;SIP:Then click on stun:, as configuredTo the main interface and then click on the extension: for the extension function can follow the instructions of the label to do the configuration, now I only do the login use function.So far, the free call function on the intranet has been configured. Now go to configure the PC-side clientI use this software eyeBeam.exe, the network has a lot of SIP protocol soft phone, we can go to searchThe configuration is as follows:After the configuration, congratulat

sip:180 ringing vs 183 Session Progress

If You know this phone is ringing (an ALERT q.931 message, for instance) you send a ringing. If You receive a notification indicating then the call was progressing, but don't know for sure whether the user I s being alerted or not, you send a 183 Session Progress message. Both can indicate early media with SDP. If There is no SDP, the end device (softphone/gateway/etc.) have to generate the ringback tone or progress tone. Usually you'll see the withou

XMPP Integration with Asterisk

Http://gnu-linux.org/xmpp-integration-with-asterisk.htmlXMPP stands for extensible Messaging and Presence Protocol, its a widely used communication Protocol. In this blog I'll use the OpenFire an opensource XMPP server.Asterisk is opensource telephony switching Exchange service for Linux. In this blog I am using the FreePBX Install on CentOS 6.5.In this blog I assume that user have already install OpenFire and asterisk server on Linux machines. So I'l

The entire process of lync 2013 and elastix 2.40 voice communication settings (2)

. 650) This. width = 650; "Title =" image (93) "style =" border-top: 0px; border-Right: 0px; Background-image: none; border-bottom: 0px; padding-top: 0px; padding-left: 0px; border-left: 0px; padding-Right: 0px; "border =" 0 "alt =" image (93) "src =" http://img1.51cto.com/attachment/201410/15/705126_1413365833VOuh.png "Height =" 58 "/> Restart the service: 650) This. width = 650; "Title =" image (94) "style =" border-top: 0px; border-Right: 0px; Background-image: none; border-bottom: 0px; paddi

Detailed functions of NGN User Access Technology

method is the development trend in the future, the user's terminal costs will be greatly reduced, and only one terminal can provide integrated access based on HPC. Wireless Network Transmission is achieved through AP supporting WLAN 802.11 protocol, the local AP is connected to an Ethernet switch with up to M ports, the user-side AP is connected to IAD downstream, or other terminal devices with up to 5 categories, such as Ephone and IAD, to achieve wireless access for end users, and access the

Introduction to self-contained programs in the operating system of UbuntuLinux

Article Title: Introduction to programs in the operating system of UbuntuLinux. Linux is a technology channel of the IT lab in China. Includes basic categories such as desktop applications, Linux system management, kernel research, embedded systems, and open source. Ubuntu already comes with a lot of great programs. Unlike Xandros, most "must be installed" programs, such as OpenOffice.org, Gimp, and other programs, have been installed by default. Therefore, you do not need to install it again.

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