Unlike Panasonic PBX systems, this isn’t actually a feature of SIP. You can send a stream of music to device, but it’s a call and makes the device “busy” and therefore it becomes a really expensive MP3 player.
Nope - unless you have more than one extension on a phone, it will only take one call at a time. Even then, if the phone is busy, it will always prefer the existing SIP connection. To accomplish this, you’d have to use an AMI script that sends a “hang up” to the phone to terminate the “noise” call so you can page to it. Even then, setting this up will require considerable AMI programming to log into the data stream, identify it, terminate the call, then play another file back.
Paging Pro has this feature, but the phones have to be configured to automatically play the recording if you are using it in a “paging” context.
If all you want is to send a file, you can do that using a Call File. The setup for call files is relatively straightforward, but there’s nothing in FreePBX that will allow you to manage the source file for your call. Note that some of this (scheduling the call, for example) is part of Paging Pro. You may even be able to set up the “playback” file from there, but there’s more flies in that ointment.
As with all automated calls, the “answer” signal causes the file to start, so steps may be required to get the audio processed correctly. For example, any extension that can receive FAXes will automatically answer the call (to listen to it) and will then ring if the call isn’t to a FAX. Some Cell carriers, as well, preemptively answer their calls before they are presented to the recipient, cutting off the start of the call.
Depending on your phone, it might be able to listen to more than one multicast address/port stream, Thusly asterisk would not be aware of a multicast you provided as a (possibly illegal ) spotify stream and as such it would not have any impact on your server above and beyond your ‘third party’ stream .
Asterisk/FreePBX could then independently also page/intercom using multicast , but depending on how your phones behaves, you might need to mute your spotify before before paging from Asterisk. (that would need a ‘hook’)
(Same problem with most radio stations, but thats beside the point)
No, Multicasts are not in any way phone calls, just network streams. so if the phone is subscribed to more than one multicast, it will either play them all or play the latest one, that is device dependent. But if you are the master of both, then turn th noisy one down while the important one is playing.
I see. So there is no way to assign priorities to multicast channels then? The receiver will not be a phone but a SIP board that I have access to programing. Is there any way the SIP terminal can identify with channel is wich?
I believe you’re trying to accomplish a similar setup as in this post:
Right now, I have several phones that play music. With the Sangoma and Yealink phones, you can enable barge, so a page will put the “Music” on hold. But essentially, I currently the music working as phone calls. I haven’t tested through the other thread listed above, but it would be nice to multicast a music stream.
Based on what I remember, I could get the music stream to multicast, but the main issue I had is once they paged or made a call, it would cancel the stream and the device would never resume, unless I stopped and started the music stream.