we have notice an annoying problem that manifests itself sometimes (but not all of the time - it is hard to repro)
when on a call, the audio will drop out for 8-10 seconds, then the call will resume as if nothing happened. Its not long enough for PSTN calls to hangup (unless the guy on the other end decides that the call is dead and hangs up).
This is on a SIP phone -> asterisk -> VOIP Provider -> PSTN setup.
We thought originally it was our asterisk -> voip provider problems, but we have found that it is on the local asterisk box !
If I make a dialplan extension like:
and dial it, I can get that same 10 second dropout that we get on the PSTN calls.
Doing a ‘top -d1’ on the asterisk box, yes, the box is actually ‘freezing’ for a second, at least, from the ssh perspective, when the box is ‘tied up’ asterisk is hogging the CPU time for that 10 seconds.
Has anyone else seen this ? This is on 1.2.17.