WebRTC SIP Phone with Click2Dial
大小:4.29MiB版本:v 0.1.6更新时间:2021-12-21
This extension is for calling by clicking tel numbers at any web page. Indispensable for call centers and any other business.
WebRTC SIP Phone with Click2Dial 的使用方法详解,最全面的教程
WebRTC SIP Phone with Click2Dial 描述:
用户数:123
分类:社交通讯插件
扩展大小:4.29 MiB
最后更新时间:2021-12-21
版本:v 0.1.6
WebRTC SIP Phone with Click2Dial 插件简介:
这是来自Chrome商店的 WebRTC SIP Phone with Click2Dial 浏览器插件,您可以在当前页面下载它的最新版本安装文件,并安装在Chrome、Edge等浏览器上。
WebRTC SIP Phone with Click2Dial插件下载方法/流程:
点击下载按钮,关注“扩展迷Extfans”公众号并获取验证码,在网页弹窗中输入验证码,即可下载最新安装文件。
WebRTC SIP Phone with Click2Dial插件安装教程/方法:
(1)将扩展迷上下载的安装包文件(.zip)解压为文件夹,其中类型为“crx”的文件就是接下来需要用到的安装文件
(2) 从设置->更多工具->扩展程序 打开扩展程序页面,或者地址栏输入 Chrome://extensions/ 按下回车打开扩展程序页面
(3) 打开扩展程序页面的“开发者模式”
(4) 将crx文件拖拽到扩展程序页面,
完成安装如有其它安装问题,
请扫描网站底部二维码与客服联系如有疑问请参考:
https://www.extfans.com/installation/WebRTC SIP Phone is extension for send and receive calls using headset and work with any SIP operator who support WebRTC technology. (https://customer.sipde.net/AsteriskWebRTC)
Features:
☆ You can connect to any service provider
☆ Making and receiving phone calls
☆ Shown call rate before making (https://customer.sipde.net/APICheckers)
☆ Shown own account balance directly in Extension (https://customer.sipde.net/APICheckers)
☆ Click to call phone numbers on any web page (available soon)
☆ Receive alerts for incoming phone calls: know instantly who is calling, answer or decline the call
☆ Forward your phone calls to your voicemail or any phone number when you are unavailable or out of place
☆ Call History
☆ Blind Transfer
☆ Attended Transfer
☆ Multi-Party Conference
Updates:
v0.1.4
- MWI and VoiceMail control added
- Initiating of SIP calls is fast now
v0.1.5
- Web services added (custom balance checker and custom rate checker)
- Sending custom http message on answer (https://customer.sipde.net/how-to-CRM)
- Attended transfer works now
v0.1.6
- More convenient and useful settings with prompts
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NOTES
* The notice you see about having access to your history and website data is automatically generated because WebRTC SIP Phone runs on every tab. But it does NOT actually monitor your browsing history or require your personal information to work properly.