shoretel softphone

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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

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.

Zimbra Two-time development related Materials

GetFreeBusy Get about free/busy information. Get information about free/busy Getoutgoingfilterrules Get outgoing (can be translated as out-of-office) filter rules. Get an out-of-Office filter Rule See:https://files.zimbra.com/docs/soap_api/8.0/soapapi-zimbra-doc/api-reference/index.html 650) this.width=650; "title=" QQ picture 20160420155911.png "Src=" http://s2.51cto.com/wyfs02/M02/7F/39/ Wkiol1cxn3gcqcq9aaf2vxoafvk3

2015 Mid-year summary

In the first half of this year, familiar with the Xwindow drawing mechanism, the implementation of XML layout xlinearlayout, Xframelayout, Xscrolllayout, xpreferencelayout layout of the implementation mechanism to master, To Xwindowsmanage management know about, follow up need can further follow up. Have a general understanding of shared memory and messaging mechanisms. The task of setting the module on the PC Softphone currently on hand can be fully

Asterisk 1.8 SIP protocol stack Analysis 2

", req, 0 ); [Dec 21 15:30:13] Debug [28437]: chan_sip.c: 21609 handle_request_invite: SIP/1501159973-0000000b: New call is still down... trying .. Sip/2.0 100 tryingVia: SIP/2.0/udp 10.10.10.84: 59584; branch = z9hG4bK-d87543-7c6375234a299e1c-1 -- d87543-; received = 10.10.10.84; rport = 59584From: "1501159973" To: "6969 (softphone)" Call-ID: zjriyjzkyzyzzdnjndrmmjhmmmnlnzdmode4ntyzzme.CSeq: 2 inviteServer: asterisk PBX 1.8.2-RC1Allow: Invite, ac

Some scattered VoIP concepts

telephone services and dial-up Internet services, it directly encapsulates user data and user line signaling in the IP package. Integrated Access Device (IAD) IAD provides both analog user lines and Ethernet interfaces for accessing common telephones and computer devices. It is applicable to users who use telephone services and computers to use data services respectively. The other type of IAD Only provides Ethernet interfaces for accessing computer devices. It is suitable for users who use bot

Libexosip2 (1-3) -- how-to send or update registrations.

protected]", "sip.antisip.com", "*", 1800, reg );Update a registration You just need to reuse the Registry identifier: Int I; Exosip_lock (CTX ); I = exosip_register_build_register (CTX, RID, 1800, reg ); If (I { Exosip_unlock (CTX ); Return-1; } Exosip_register_send_register (CTX, RID, Reg ); Exosip_unlock (CTX ); Note: The above Code also shows that the stack is sometimes able to build and send a default SIP messages with only one API callClosing the registration A

Deploying IP voice on a wide area network

distributed across IP networks, providing a distributed, virtual telephone network that greatly enhances the scalability and availability of enterprise IP telephony solutions.    Cisco CallManager supports user features such as automatic answering, call forwarding, hotline, call keeping, call waiting, calling line recognition, caller name identification, direct inward/outward dialing, telephone catalogue, distinguishing vibration ling, multi-party conference telephone, etc.    Cisco CallMan

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