I have a FreePBX system in a school. We have SIP speakers through the building. For certain emergency we need to be able to press a key on a phone and have a message be able to be played through the speaker system and then the page hang up.
I can get the message to play, but the line stays open. I need to be able to press a button, it plays and shuts the phone line down.
How are you doing that? what is the hardwae? using a page or a multicast?, with a dialplan or perhaps AMI?
You will need to provide a lot more data . . .
We do this with
Essentially the call is generated like a page. It may be worth looking over the code at
I am using multicast to my speakers. Besides that I have no plan on how to do this.
I did read up on dial plans a bit today and it looks like it might be the best way. But I have never created a dial plan so if you can offer a good place to look it would be great.
either stay in the channel until the the multicast is finished and then hangup (best), or set an absolute hangup time to the channel (easiest)
Does it have to be desk phone or can it be softphone?
Would have to be a desk phone.
Okay, and so how do I play a message using the “dial plan”. I have looked up examples But I am a bit confused on the format.
I would start with
Here’s the issue, I have a video phone at the front door. It calls a queue group (working). After 15 seconds, it is to play a sound file to a multicast address and hang up.
Here is the dial plan
exten => 4001,1,Answer()
same => n,Set(TIMEOUT(absolute)=10)
same => n,Set(MUTEAUDIO(in)=on)
same => n,Dial(MulticastRTP/basic/126.96.36.199:4001,gA(/var/lib/asterisk/sounds/en/custom/ding-dong))
same => n,Hangup()
It works when I call from a normal phone, but when I call from the video phone, it …
his last post (using ffmpeg) is what I now use as a jack of all trades announcer and occasionally so to listen to the BBC world service because it doesn’t need to get asterisk involved.
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