Hi,
selecting and playing music is already taken care of - the shop runs a sound station (basically a big hard disk with a pc), and every workstation can play music via realplayer, mgp123 or whatever; the challenge is to turn an existing phone call into a conference and join the music source to that conference with as few buttons as possible
Musicman
playing again with the manager interface, I noticed that the “status” command can be sent without a channel. So for the moment I am using a script that gets status, searches for a call specifying my exten, and redirect that to a meetme room. Next it will originate a call from local/ to mp3player.
It remains unclear which call to use on a two-line phone, but the intended usage scenario would not have two simultaneous calls
Musicman
[quote=“musicman”]Hi,
selecting and playing music is already taken care of - the shop runs a sound station (basically a big hard disk with a pc), and every workstation can play music via realplayer, mgp123 or whatever; the challenge is to turn an existing phone call into a conference and join the music source to that conference with as few buttons as possible
Musicman[/quote]
Have you even taken a look at what the Autohybrid has to offer? 1) It acts like a normal phone, 2) it auto answers. 3) It could be used to get your music into a conference room very cleanly without having to fight to get mp3player to work. Think of it as just another person calling in.
Hi,
my impression was that it is a device without a phone, so one would perhaps connect a phone through.
I hope to get IP phones for this system, so this would have to go on some kind of ata, and get the “setup conference with this phone” in the same way as the mp3 player does.
Also, I would probably need 3 or 4 of those units, so everyone would have one connected to their pc speakers. This seems to get almost as expensive as the basic phone system (an asterisk server with a BRI card and 6-8 IP phones)
Musicman
Server, bout 2 GHz, 512 MB RAM and a single 40 GB Harddisc is more then nuff for such system. 400-700,- Euro (depends where you buy, new/used etc).
SNOM 360 bout 190,- Euro
Bri Card (Longshine HCF): Mediamarkt 19,-
(6 X 190) + 700 = not even 2000,- €
You wont get anything in the same Zipcode like Asterisk (function rich etc) for that money.
SipSoftphones (free) for “injecting” Music are free.
Asterisk does do streaming music on hold it should not be that hard to use that as the starting point.
each workstation is running what PowerCat??? Tune Tracker??? Sam3???
Some radio broadcast is built for streaming
Is the station realtime??? or cannned shows… do you stream to the inet now??? if so what format???
[quote=“bubba”]Asterisk does do streaming music on hold it should not be that hard to use that as the starting point.
each workstation is running what PowerCat??? Tune Tracker??? Sam3???
Some radio broadcast is built for streaming
Is the station realtime??? or cannned shows… do you stream to the inet now??? if so what format???[/quote]
MOHstreaming isnt the point here. The issue is about selective musicinjection into established channels without breaking the RTP streams for involved partys.
Well I have a radio station with music hold via some OLD hacked up patch boards and we can cue up any song we wish from any of five stations using streaming on hold on any line, so if you can play music in the background then, so yes it applies
Hi,
thanks for the many responses:
as for hardware: I am looking into either Snom phones or some dect stuff; I hope that there will be new dect or wlan offers within the next two or 3 months. This project sort of competes with an 8-line isdn to analog pbx box (I got a used eumex 208 for 50 euros…) plus cheap analog dect phones. It is very clear to me, and to the office stuff, that the extra features will pay off soon - but I need to convince the boss that cheap off-the-shelf pbx boxen from the local office cash&carry do not necessarily integrate well
as for playing music: music will be specifically chosen for some callers (we usually try to direct them to our website, but some have no internet access at all, or no internet access during business hours, or are not able to configure their browser). Playing music on the individual workstations in the office works well - I had trouble finding an easy interconnect between the computers and the phone
musicman
I think you could do something like this
nerdvittles.com/index.php?p=92
If you created two or more context pointing to the "live streams off the workstations
tie it into the system using ext
would need to have a local shoutcast / nicecast server system And as I think you said winamp it would be shoutcast.
“Keep in mind that anything above a 24K mono stream is wasted on a telephone call so there’s no need to choose a 128K stereo audio stream unless you just want to eat up your bandwidth”
tie it into the system using ext
Yeah, but HOW you want to do this?
You CANT use MOH since MOH is (like the name says) something becoming active ON HOLD.
Again: Read his initial posting, he needs to “inject” music into an established line. There is no MOH when you want to keeptalking over the music !
You can only accomplish this by having a conference and on of the conference party is a softphone which is using the soundcard wave-out as the source. And THERE you have winamp running.
I found this while researching something a bit different…
Go to the features.conf config file, and add this to the end:
[applicationmap]
testfeature_e => #9,callee,playback,some-wav-file
testfeature_r => #9,caller,playback, some-wav-file
go back into the asterisk console and type:
reload res_features.so
Read up on the Calle/Caller to determine what to set.
In your “application extension”
you’ll need to add:
exten => s,n,set(DYNAMIC_FEATURES=testfeature_e#testfeature_r)
now during the call, you can press #9 and it should play back the file…
Problems: You can’t use any variables here, so if you want to change the files, you have to edit the features.conf file and reload the res_features.so
Hi,
I was toying with applicationmap as well, but it did not seem to work with an agi.
As far as replacing the soundfile into the setup - as long as only one person at a time uses the feature, some script could symlink the actual soundfile to a default name.
musicman
Application mapping isnt working here, since it only knows:
CALLER
and
CALLEE
You cant play the stream to both partys since it doesnt know the parameter “BOTH” (which is EXTREMELY annoying…COUGH!!!)