Is it possible that someone calls skype number with extension and call is forwarded to another number based on extension? I'd like customers to reach my vendors but with me knowing about this and them without seeing the destination number.
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I'm working on a project that has server and client roles. I would like to have servers and clients automatically detecting each other. At a first glance, zeroconf seems to be the best solution. But it would add a dependency, Bonjour, to the project. I use Qt for the GUI and Qt has native support of broadcast and multicast. So I'm wondering if it's feasible to just use those features to replace zeroconf?
Here is a decent post about how zeroconf works.
I don't think I need the features of obtaining an IP Address and obtaining a Hostname from zeroconf. All I want is let one instance be aware of other instance's existence.
My current plan is combining broadcast and multicast. Each server chooses a unique multicast group and broadcast this group to the others. When a client wants to connect a to specific server, it joins the corresponding group.
Some people mention that it's normal that routers blocking local broadcast. If this is true, my plan would not be feasible.
Is there any standard way to implement this rather than using zeroconf?
NET experts,
I have a scenario where a 4-port PSTN card is installed in a server and I have installed Freebpx on that server as per suggestion by someone. When a call comes on any of the PSTN line, it is forwarded to one of the operators on his hard phone.
Each operator is also having a computer screen at his table, powered by an individual CPU. This runs our CRM software to be handled by operator. When a call arrives to an operator hard phone, say operator 2, we want that the Caller number should also be displayed in the CRM software. Based on this caller number, operator can enter some information related to the Caller and save it in database via our CRM software. Also, when operator disconnect the call, we should receive call stop time for statistics later on.
Thus, we need caller number and call start time when a call is picked by an operator on his hard phone. and then we need call end time when a call is finished.
Can someone help us how we can achieve this? Do we have to capture the SIP packets and parse them or their is some other way to do so? Our CRM database is totally separate from the Freepbx and resides on another server.
If you want to get these events realtime, you should look at AMI (Asterisk Manager Interface - port 5038 TCP by default) and it's configuration manager.conf (note: FreePBX uses the, see manager_custom.conf).
If you want the archived version, you should set up a database server, and point the CDR (Call Detail Records) module to it. PostgreSQL or MySQL/MariaDB works just fine. Asterisk will simply ignore additional fields in the CDR as long as they can be NULL or has a DEFAULT value. This can be used to store custom data.
I was following THIS tutorial in order to implement video calling using WebRTC. The example allows people of the same group to communicate using video or audio and it works well. I was wondering if you could Video call people who are on different pages of the same website without having to be on the specific 'video call page' only?
What is the way to allow incoming calls even when you are not joined in a group? I believe it must be a group request (Sending a request to join from Person A to Person B). Is this possible using WebRTC?
WebRtc is not based on "who is connected to what page".
The connections only happen through your signalling server logic. This means that any webrtc peerconnection can connect to anyother peerconnection as long as your signalling server relays the connection startup logic(SDPs and Ice candidates).
The webpage is only a way to display the media that you have, the javascript you wrote will run on any page you want.
I need to stream several "channels" (by channels I'm thinking of radio channels, so playlists might be more appropriate) of queued MP3 files to around 200 clients over the internet from a Windows 2008 R2 / IIS 7 web server. Encryption of the stream is not a requirement.
I need some way to ensure each client can only stream one channel at a time. I was thinking of restricting by IP address, and would welcome any suggestions on how I might go about this, or if there may be a better way.
For my clients I assume I'd need a player on the client end that could facilitate "logging in". Ideally I would be able to stream to windows xp/7/8, mac and android clients.
I would need to be able to log in remotely and control each channel / playlist from a finite list of MP3 files hosted on the web server.
I'm wondering if there any off the shelf products that I could use to do this. If not, I'm stuck on what would be the best way to go about this.
I've briefly read up on shoutcast, auto dj, streamcast and ice cast, but I don't have any experience with these solutions, and can't find any information about how to implement the security requirements I have (limiting access to one stream per client / ip address).
IP addresses do not uniquely identify users. There are plenty of situations where NAT comes into play, and you can have hundreds of users all behind the same public IP address.
What you need to do is have another method for identifying users. Assuming you don't want to require accounts, you can use a session ID.
Basically, you assign an ID with a cookie to a browser. When the user clicks a link to launch their audio player, the session ID is passed in the URL to the stream. With this method, it doesn't matter if the browser itself or a separate audio player is used to play the stream. It is up to the streaming server to then accept or reject the request.
It is common to accept any new request and disconnect the old when a new stream starts. Icecast doesn't support this sort of thing natively, but it does provide an API of sorts with the admin interface that you can use from your own scripts to get this behavior.
Alternatively, I have written a server called AudioPump that provides similar functionality. It isn't generally available yet, but please contact me at brad#audiopump.co if you're interested.
I am trying to build an application where I am required to record and playback simultaneously. The application needs to go live on an asterisk telephony server. My problem is:
A user calls the asterisk server and starts to speak.
The voice packets being sent by the user are recorded in a wav file on the system.
A copy of the voice packets are sent as feedback simultaneously.
I have taken a look at ChanSpy, but it will not work if one is using Record.
My questions:
1. If a user calls an asterisk server, does that channel become a simplex or half-duplex channel?
2. Are there any commands etc. that allow us to do the above?
3. If not, does that mean I need to go into C programming for asterisk (agi-bin)?
P.S: Please let me know in case more information is needed.
Why not try MixMonitor? That allows you to record the call, and doesn't interfere with ChanSpy usage. ChanSpy IS the best way to do this, by the way.
You are wanting to feed to voice channel from the caller back to the same caller correct?
Have you tried the Echo command?