asterisk

问题 [out] exten=>_X.,1,Answer() exten=>_X.,n,Background(hello) exten=>_X.,n,WaitExten(5) exten=>1,1,Goto(check,s,1) Audio menu are playing, but when I press 1, sound stop and after few second check command run. How to quickly respond to a user request? 回答1: You are doing begginer error which described in almost any asterisk book for beginer. I recommend you read "Asterisk The Future of Telephony" by O'Relly. This exact error is simple: in your dialplan you have extension _X. , which mean "any

问题 Assume line is: "Chicago Sun 01:52" . What does *a, b, c = line.split() do? In particular, what is the significance of the asterisk? Edit: Upon testing it, it seems like "Chicago" , "Sun" and "01:52" are all stored in a , b and c . The asterisk seems to lead to "Chicago" being stored in a as the first element of a list. So, we have a = ["Chicago"] , b = "Sun" and c = "01:52" . Could anyone point to material on the functionality of the asterisk operator in this situation? 回答1: Splitting that

问题 I once used pre-configured asterisk to make calls (using AMI). When I do that the response from originate used to have channel and unique id infos. Now I'm trying to build a new Asterisk. Everything is set but when I call originate only info that response has is "call successfully queued". Is there any option like "Show extra info on response" hidden somewhere? You can find the two different responses I get. this one is the old one, includes some valuable info. Response: Success ActionID:

问题 I have created a linux service that runs as a deamon (and gets started from /etc/init.d/X). I need to set some environment variables that can be accessed by the application. Here's the scenario. The application is a bunch of Perl AGI scripts that depend on (and therefore need to run as) asterisk user but asterisk doesn't have a shell. Ideally I'd just set this in /home/asterisk/.bashrc but that doesn't exist for asterisk. How can I set environment variables for my app in the asterisk user's

问题 I'm writing a rtp video streamer for android that reads h264 coded data from an Android local socket and packetize it. The thing is that I did it but I keep getting black frames in the client side (Voip). The communication goes like this: Android -> Asterisk -> Jitsi (Osx) (and reverse) There are a few things that I haven't understood yet: 1) Android's mediarecorder gives me a raw h264 stream, How can I know when a NAL starts / ends based on that stream? It doesn't have any 0x000001 pattern

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 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? 回答1: Twilio Seems to offer SIP Trunking now. 回答2: As

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

问题 i'm facing the following scenario: we have to local (REGISTERED) users (iOS apps pjSIP) which initiating local calls between each other. the problem arise when one of the users (let's say user B) is closing the application few minutes after he successfully REGISTERS. now, when user A tries to call user B we see that the INVITE is sent but we got no reply (e.g 180 ringing) from User B. Note: when we are sending an invite to User B he get's Push notification to his device what cases him to open