asterisk

问题 i have to implement webRTC solution wich allow phone call via browser based on asterisk and node.js (video and audio call are ok thanks to opensource project) any guide thanks 回答1: check this project https://github.com/js-platform/node-webrtc But i have inform you that webrtc is in early beta stage, it is experemental technology. So to do something in this field you have be sip and node.js guru and understand how to search info in web and deal with sockets. 回答2: I've made a prof of concept

问题 I am writing a java application to dial a number and get user's pin by DTMF. I am using asterisk-java connect to an asterisk VOIP server and dial the number but I don't know how to stream a file and read user's input as DTMF. Here's my code: OriginateAction originateAction = new OriginateAction(); originateAction.setChannel("SIP/1001"); originateAction.setContext("default"); originateAction.setCallerId("Server"); originateAction.setPriority(1); // connect to Asterisk and log in

问题 I’m using FreePBX with Asterisk’s Java API. For the moment, I’m able to display all my SIP peers with their respective states: public void onManagerEvent(ManagerEvent event) { // Look if the event is a IP phone (Peer entry) if(event instanceof PeerEntryEvent) { PeerEntryEvent ev = (PeerEntryEvent)event; // Get the user extension peer = ev.getObjectName(); // Add to the array peersName.add(peer); } } I’m able to display the phone number and name of both callers when a channel is open: private

问题 I am applying an auto dial in asterisk using .call file My a.call Channel: DAHDI/g0/09********* MaxRetries: 1 RetryTime: 600 WaitTime: 30 Context: outgoing Extension: 10 Priority: 1 My problem is that every time above number is called by same number means even if i change the dialled number(receiver number above) the caller number is same. How can i set the caller number in an outgoing call? Thanks in advance. 回答1: You can use Channel: DAHDI/g0/09********* MaxRetries: 1 RetryTime: 600

问题 I have a dialplan that contains the IVR flow for a number of applications, all on different extensions. I am now trying to clean it up by moving some of them to separate .conf files. Here is how the external .conf files are being #included into extensions.conf at present. All the separate .conf files are present within /etc/asterisk : In extensions.conf : #include "temp.conf" [globals] ... The IVR in temp.conf works OK, but none of the other applications work. While the incoming dispatcher

问题 Can I use SIP account from database? As I understand. I can config SIP account in sip.conf. But I want to support use many account. Can I get SIP account from database that I created for dial out instead of sip.conf? Or Other way I can do it. 回答1: What you are looking for is called 'RealTime' in Asterisk (yes, it doesn't make much sense ;) ). http://www.voip-info.org/wiki/view/Asterisk+RealTime http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip 来源： https://stackoverflow.com/questions

问题 I'm using a cloud-based Asterisk server as my PBX. At my current location, the Internet is rather shaky, but cell phones are reliable and commonplace. However, international cell calls are expensive, VOIP calls are much cheaper. So, I came up with a script in Asterisk which dials my local cell phone: exten => _abcd.,1,NoOp(-- Making outbound call to number ${EXTEN:4} --) same => n,Answer() same => n,Wait(1) same => n,Originate(SIP/+86[my_cell_no]@[voip_provider],exten,incoming_remote,##$

问题 As a beginner that just installed his first asterix server I came across a small issue. i want to check if a number that i dial is busy or not before actually dialing it. So for example I would call 0904 => number busy => redirect to 0905 However I don't want it to dial 0904 first if its busy but just instantly redirect it to 0905 instead. I have a very basic setup with just 3 users. 2 of them are in a queue "support" one of them is in the queue "admin" I have seen and read a little about

问题 I'm currently working on a project 'email to voice call'. Using python i'v extracted the email & converted it into speech and saved in a WAV file. Now using asterisk (I'v installed Asterisk 10.2.1 on my ubuntu 10.10 os) i want to generate call to the cell phone (say 919833000000 india's no.) of the user through my system. I have written a python code to connect to asterisk manager interface. Also i have configured the sip.conf and extensions.conf files as well as manager.conf. I have

问题 In relation to SIP registration, simply trying to add termination URI: Termination URI Configure a SIP Domain Name to uniquely identify your Termination SIP URI for this Trunk. This URI will be used by your communications infrastructure to direct SIP traffic towards Twilio. When you point your infrastructure toward this URI, Twilio uses a Geo DNS lookup to intelligently direct your traffic to our closest POP. Learn more about Termination Settings I can add <foo>.pstn.twilio.com fine, but get