asteriskami

问题 I am wondering if there is an easy way to hangup a channel created with Originate? What I do is following: Call in dialplan triggers Agi; Agi starts AMI: Originate with Channel "SIP/201". Now the extension rings. At answer I can bridge the channels. But, if the calling party leaves the call before the call is answered, I would like to stop the outgoing call. When I send AMI Hangup with Channel "SIP/201", It can't hangup originated call. I can stop that call using 'hangup request channelname'

问题 I am wondering if there is an easy way to hangup a channel created with Originate? What I do is following: Call in dialplan triggers Agi; Agi starts AMI: Originate with Channel "SIP/201". Now the extension rings. At answer I can bridge the channels. But, if the calling party leaves the call before the call is answered, I would like to stop the outgoing call. When I send AMI Hangup with Channel "SIP/201", It can't hangup originated call. I can stop that call using 'hangup request channelname'

问题 I have been building a WinForms desktop application using C# that interfaces with Asterisk using Aster.NET (formerly/forked from Asterisk.NET). We're having real trouble reliably identifying and tracking calls that are related to an individual extension/user. The problem we're having is due to the unpredictable/fuzzy nature of the events fired/triggered by Asterisk, with them being massively variable depending on how the call is routed before it hits an extension. For example, the event

问题 I have been building a WinForms desktop application using C# that interfaces with Asterisk using Aster.NET (formerly/forked from Asterisk.NET). We're having real trouble reliably identifying and tracking calls that are related to an individual extension/user. The problem we're having is due to the unpredictable/fuzzy nature of the events fired/triggered by Asterisk, with them being massively variable depending on how the call is routed before it hits an extension. For example, the event

问题 I am doing something about ConfBridge these days. I used "confbridge kick all" to end the conferences when admin user exit, but the others in the conference will hear "you have been kick out from this conference.". it is not a good way to end a conference. Is there any configure or command which can set that conference will end up when the admin user exit the conference? 回答1: Just set endmarked=yes on the user profile you're using (found in confbridge.conf). The docs say: ;end_marked=yes ;

问题 I have followed the instructions in this thread: Asterisk AMI - pickup call. However, I am still unable to answer calls via AMI. I can make the call to the extension, but corresponding phone for that extension doesn't ring. I can then run the AMI command to answer that call, it does answer, but obviously there isn't any actual response. Dialplan (testing with extension 116): exten => 116,1,AGI(agi:async) Any ideas what I am doing wrong here? 回答1: Use exten => 116,1,Answer exten => 116,2,AGI

问题 In extconfig.conf they have mentioned that "However, note that using dynamic realtime extensions is not recommended anymore as a best practice; instead, you should consider writing a static dialplan with proper data abstraction via a tool like func_odbc." 1) Why asterisk is not recommending dynamic realtime extensions? 2) How to do static dialplan with data abstraction using tool liek func_odbc? My requirement is having have more extensions (in this case mobile number) coming up, how can I

问题 I am trying to create an Application for Asterisk Interface, the configuration setup has been done properly as I have double checked it. It seems the reason why I can't access Config files from Asterisk server has to do with the connection. I need to use a cookie for connecting to the same instance of the login session. But so far I am unable to use Cookie Properly need a little help. I am supposed to login by pressing button1 (bt1) and get config file by pressing button2 (btn2). public class

问题 I have a completely standard installation of Trixbox with 2 SIP extensions set up on it. Both extensions are Snom 370 SIP phones. I can originate a call from one extension to the other using the following CLI command: originate sip/101 extension 102 This causes the phone on 101 to ring, then when that phone is picked up it dials 102. What I would like is for the phone on 101 to automatically call 102, without 101 waiting to be picked up. Is this something that can be done, or do the SIP

问题 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