[Asterisk-Users] Audio cuts out

[Gary Richardson]

Hey All,

I've been experiencing a problem for a bit. During a call to the PSTN, audio
will cut out for 2-5 seconds. It's completely random and may or may not
happen during a call.

Our setup is 79XX phones -> asterisk -> 2811 router -> PRI to the PSTN.
Everything is talking SIP. The asterisk box is a dual core system.
/proc/interrupts looks like:

 cat /proc/interrupts
           CPU0       CPU1
  0:  733669449  732813122    IO-APIC-edge  timer
  8:          1          0    IO-APIC-edge  rtc
  9:          0          0   IO-APIC-level  acpi
 14:    6598410    6589174    IO-APIC-edge  ide0
169:          0          0   IO-APIC-level  uhci_hcd
185:          0          0   IO-APIC-level  ehci_hcd, uhci_hcd
193:          0          0   IO-APIC-level  uhci_hcd
201:          0          0   IO-APIC-level  uhci_hcd
209:   11404158   10762030   IO-APIC-level  3w-9xxx
225:  100440701        136         PCI-MSI  eth0
233:         14   10512166         PCI-MSI  eth1
NMI:          0          0
LOC: 1466464719 1466464718
ERR:          0
MIS:          0

Can-Reinvite is enabled, but I do have it configured to allow call recording
on outbound calls, so I think the audio streams all go through asterisk.
There are no G.729 licenses involved and everything should be talking G.711.


Oh, and this is an 1.2.7.1 install. ztdummy is loaded.

Does anyone have any insite into this problem?

Thanks.
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