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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