Hi All
I am occasionally hearing a slight pop or skip in an audio message playback
on one ALL SIP installation. There are other Audio problems with the
installation too (underwater Audio, 1 second 1 way audio delay (takes 1
second for Audio spoken by the customer to reach the agent, and Robotic
voices).
I have looked up many guides and memorized voip-info.org... I am starting to
wonder if the problem is in my timing. I was using dahdi_dummy, I have put
in a sangoma card and I am using it as a timing source (and only as a timing
source).
Here is the output of my timing source below.
*CLI> timing test 50
Attempting to test a timer with 50 ticks per second.
Using the 'DAHDI' timing module for this test.
It has been 1002 milliseconds, and we got 51 timer ticks
*CLI> timing test 100
Attempting to test a timer with 100 ticks per second.
Using the 'DAHDI' timing module for this test.
It has been 1002 milliseconds, and we got 102 timer ticks
*CLI> timing test 500
Attempting to test a timer with 500 ticks per second.
Using the 'DAHDI' timing module for this test.
It has been 1001 milliseconds, and we got 510 timer ticks
*CLI> timing test 10
Attempting to test a timer with 10 ticks per second.
Using the 'DAHDI' timing module for this test.
It has been 1080 milliseconds, and we got 11 timer ticks
in asterisk.conf, internal_timing = yes
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