[asterisk-users] Asterisk 11.24.1 garbled audio

Max Grobecker max.grobecker at ml.grobecker.info
Tue Nov 15 17:11:18 CST 2016


Am 15.11.2016 um 17:52 schrieb Olivier:
> Hi,
> How can I double check which timer is currently is use in a running system ?
> core show settings doesn't tell anything, if I'm not mistaken.

To determine which timing module is currently in use, you can take a look at "module show like timing".
There should be only one module with "use count" 1 - that's the one that is currently used.
If there is no call running, you can unload any additional timing module you don't want to use to force Asterisk 
to use the only one left by simply doing "module unload res_....".

Also, please check in "core show settings" if internal timing is enabled or not. If it's not, please enable it in asterisk.conf.
The internal timing should be enabled by default, but if it's not Asterisk might not use any timing module at all if RTP is being bridged between two ends of a call.
Asterisk normally synchronises the RTP clocking to one end of the call. But if this RTP source is not realiable (jitter, packet loss, silence suppression...) you
can end up having audio problems.


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