# Asterisk MusicOnHold Notes (Streaming and Classic Hold)
The way custom applications work with Asterisk, from my very very basic functional perspective is it expects 8khz PCM on stdin. So if you had a binary called "audioplayer" that played audio to stdout as raw 8khz PCM samples; then you'd just have to call that binary as a custom application in your musiconhold configuration:
The way custom applications work with Asterisk, from my very very basic functional perspective is it expects 8khz PCM on stdin. So if you had a binary called "audioplayer" that played audio to stdout as raw 8khz PCM samples; then you'd just have to call that binary as a custom application in your musiconhold configuration:
That's it. Provided it only literally sent data to stdout; this would work. But as I alluded to in the example; such a thing doesn't exist. Thankfully shell scripts count as applications and as long as we ultimately get some audio to stdout; it will work.
That's it. Provided it only literally sent data to stdout; this would work. But as I alluded to in the example; such a thing doesn't exist. Thankfully shell scripts count as applications and as long as we ultimately get some audio to stdout; it will work.
Icecast/Shoutcast Source Streams
### Icecast/Shoutcast Source Streams
This would be a lot easier if mplayer supported stdout; but it doesn't. So in order to make this work we need to make a named pipe, write to that pipe, and then somehow pipe the pipe to stdout.
This would be a lot easier if mplayer supported stdout; **but it doesn't**. So in order to make this work we need to make a named pipe, write to that pipe, and then somehow pipe the pipe to stdout.
Define the path and name of our pipe using a pre-determined prefix and random number suffix.
- Define the path and name of our pipe using a pre-determined prefix and random number suffix.
Create this named pipe with mknod.
- Create this named pipe with mknod.
Play back the stream using mplayer; outputting PCM to the named pipe, resampled to 8khz, and downmixed.
- Play back the stream using mplayer; outputting PCM to the named pipe, resampled to 8khz, and downmixed.
CAT this pipe to stdout.
- CAT this pipe to stdout.
Delete the named pipe.
- Delete the named pipe.
Originally this script had things to find existing pipes and delete them. Storing them in /tmp means they'll disappear on reboot. If things go right you won't need to constantly start and stop anything. You can however always look at running processes and see which pipes are being used.
Originally this script had things to find existing pipes and delete them. Storing them in /tmp means they'll disappear on reboot. *If things go right* you won't need to constantly start and stop anything. You can however always look at running processes and see which pipes are being used.
The original version of my Classic Hold used this, in combination with icecast2 and ices2.
The original version of my Classic Hold used this, in combination with icecast2 and ices2.
But surely we can do this without getting icecast ivolved....directly with applications.
But surely we can do this without getting icecast ivolved....directly with applications.
Oh...sure...we can...but holy cow.
Oh...sure...we can...***but holy cow***.
Classic Hold Version 2: Less Complex, More Headache
## Classic Hold Version 2: Less Complex, ***More Headache***
Let me start by saying this: getting this to work started out as medium-hard; before going full fledged pull-my-hair-out. This is because there's a number of additional steps we have to do that are much more involved than setting up and configuring icecast and ices2. Most of these steps were documented and I had to piece them together. This was on top of the fact that debugging on a headless machine for sound problems posed limitations. I had to test things on a local VM before attempting to implement them on the remote machine.
Let me start by saying this: getting this to work started out as medium-hard; before going full fledged pull-my-hair-out. This is because there's a *number* of additional steps we have to do that are much more involved than setting up and configuring icecast and ices2. Most of these steps weren't documented and I had to piece them together. This was on top of the fact that debugging on a headless machine for sound problems posed limitations. I had to test things on a local VM before attempting to implement them on the remote machine.
The concept sounds simple; just play some files from a playlist to a pipe; pipe it around; pipe it in to Asterisk. Okay, hold up. It sounds easy; but is it? There's a little issue in that if you tell most players you want to play back to a file and not a sound device; they decode, not play. We don't want, need, or can handle this. We need to simulate real-time playback. The only way to get this done effectively...was pulseaudio. I tried ALSA but this broke on ubuntu; they don't actually include the stuff to configure it. So you'll have a working alsa system; install alsa-utils, and lose it all.
The concept sounds simple; just play some files from a playlist to a pipe; pipe it around; pipe it in to Asterisk. Okay, hold up. It sounds easy; but is it? There's a little issue in that if you tell most players you want to play back to a file and not a sound device; they decode, not play. We don't want, need, or can handle this. We need to simulate real-time playback. The only way to get this done effectively...was pulseaudio. I tried ALSA but this broke on ubuntu; they don't actually include the stuff to configure it. So you'll have a working alsa system; install alsa-utils, and lose it all.
But while we can create null sinks in PulseAudio stupid easily; there's a major drawback. It usually runs at the userlevel. Why does this matter? The asterisk user never actually logs in to the system...and therefore pulseaudio server never gets executed. So log in and activate? I tried that. So what you have to do...is run pulseaudio as system-wide and then configure it so the non-authenticated asterisk account can actually use it.
But while we can create null sinks in PulseAudio stupid easily; there's a *major drawback*. It usually runs at the userlevel. Why does this matter? The asterisk user **never actually logs in** to the system...and therefore pulseaudio server never gets executed. So log in and activate? I tried that. So what you have to do...is run pulseaudio as system-wide and then configure it so the non-authenticated asterisk account can actually use it.
