asterisk

在css中插入以下代码即可。 .el-form-item.is-required:not(.is-no-asterisk) .el-form-item__label-wrap>.el-form-item__label:before, .el-form-item.is-required:not(.is-no-asterisk)>.el-form-item__label:before { content: ''; } 如果觉得这样不美观，可以单独创建一个css文件存放以下代码，并在需要使用的页面中引入外部css文件实现同样的功能。 来源： https://my.oschina.net/u/4115842/blog/3113143

I am learning asterisk. After I installed asterisk, I tried to connect with it using asterisk -rvvvvc. But it gave me the following error message: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) How can I solve this issue? It's probably because asterisk is not running on your server. Try to run it with this command : asterisk -vvvvvvc You'll enter into the Asterisk CLI and if something goes wrong you'll see it. After that you can quit the CLI by entering the exit command. Then you can reconnect to the CLI by typing asterisk -r . All this commands assumed you're the

I have a Twilio account which has a number (let's say 8881231234), and I have Asterisk box. I'd like to use Twilio as an Asterisk trunk to be able to make calls at their rates and receive calls from that number on my Asterisk. I haven't found any specs to interconnect Asterisk with Twilio. Is it possible to set up Asterisk so that every outgoing call is routed through Twilio and have the calls on my 8881231234 number ring on my SIP phone? user2152758 Twilio Seems to offer SIP Trunking now. g3rv4 As it was well pointed out below, Twilio now DOES WORK as a SIP trunk, you can visit https://www

My objective: I want to use softphone(3CX phone) register with asterisk server, and make call to the server and asterisk act as a server to automatically response something, like play a song. How i did: I installed asteriskNow using virtualbox, and registered the softphone by setting exntension for my SIP device (extension 333). And i write a dialplan in etc/asterisk/extensions.conf . The dialplan is : [incoming] exten =>s,1,Answer() exten =>s,n,Playback(dir-intro-oper) exten =>s,n,Hangup() I want any incoming call to server, the server will automatically answer, and play a pre-defined voice

I want to pickup call in Asterisk using AMI. I can originate call, but totally don't know, how to answer the phone... Script for calling: #login sock = socket.socket(af, socktype, proto) sock.connect(sockaddr) sock.send('Action: login\r\n') sock.send('Events: off\r\n') sock.send('Username: '+str(ast_server.login)+'\r\n') sock.send('Secret: '+str(ast_server.password)+'\r\n\r\n') #originate call sock.send('Action: originate\r\n') sock.send('Channel: ' + str(user.asterisk_chan_type) + '/' + str(user.internal_number)+'\r\n') sock.send('Timeout: '+str(ast_server.wait_time*1000)+'\r\n') sock.send(