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russell at digium.com
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 Posted: Thu Jan 17, 2008 3:52 pm Post subject: [asterisk-users] buffer-issue when piping live-streams into |
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Michael Kamleitner wrote:
| Quote: | 10:00 I'm calling the pbx, musiconhold starts correctly to play the
live-stream (almost live, with very small delay) - that's OK.
10:01 I hangup.
-- than I pause for 20 min --
10:20 I'm calling a second time. However moh now doesn't stream live, but
starts to continue playing the stream from 10:01. This goes on for about
30secs, then the replay stops for a second and continues at the correct
position (once again, rather "live"). along I get this message at the
console:
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<snip>
| Quote: | musiconhold.conf:
[default]
mode=custom
application=/etc/asterisk/mohstream.sh
mohstream.sh
#!/bin/bash
/usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o
raw:- --mono -R 8000 -a -12 -
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Most players don't work quite correctly with Asterisk MOH. For it to work the
way you expect, the player you are using must throw away the audio when Asterisk
isn't currently reading from the stream. There was a magic version of mpg123
(0.59r IIRC) that did that, and that is why it was the recommended version.
If you're reading from a raw TCP stream, then you can use the small streamplayer
utility included with Asterisk. Otherwise, I don't really have a good
suggestion for you right now. I suppose that you could use some sort of hack to
ensure that music on hold is always playing so that the stream is being serviced.
extensions.conf:
[moh_hack]
exten => hack,1,Answer
exten => hack,n,StartMusicOnHold(default)
exten => hack,n,While(1)
exten => hack,n,Wait(300)
exten => hack,n,EndWhile()
*CLI> originate Local/hack at moh_hack application Echo
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc. |
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russell at digium.com
Guest
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| Posted: Thu Jan 17, 2008 5:32 pm Post subject: [asterisk-users] buffer-issue when piping live-streams into |
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Michael Kamleitner wrote:
| Quote: | thx a lot russel...your hack actually works!! 
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Awesome.
Well, that option only exists in Asterisk trunk, and is only relevant when using
realtime for music on hold. I assume you're probably using one of the released
versions of Asterisk, so this wouldn't be available.
| Quote: | anyway, thx a lot for your suggestions
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You're quite welcome. I'm glad I could help out.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc. |
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