Since you don't mention a timing source, you might not
be using one - you probably need zaptel or ztdummy, there
doesn't seem to be any way to do this without kernel
modules (if there is, I'd be really interested to know).
be using one - you probably need zaptel or ztdummy, there
doesn't seem to be any way to do this without kernel
modules (if there is, I'd be really interested to know).
------------
Apparently only some of the audio glitches disappeared after disabling the "Voice Activity Detection" functionality of the Thomson ip-phones.
e.g. I get perfect MP3 playback on extension 139 -
exten => 139,1,MP3Player(/usr/local/share/asterisk/sounds/queen.mp3).
But when putting a call on hold (musiconhold.conf), there are still periodical (every 2-3 seconds) audio hickups. I do have a HFC ISDN adapter installed, but the audio hickups also happen on SIP to SIP calls. I don't have a timing source installed as far as I know.
How can I do this on my install (Asterisk 1.2.9.1, FreeBSD 6.1 )? Or do you have hints for the musiconhold configuration file, because I tried the different included examples - but none helped.
Thanks,
Frank
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