[Asterisk-Users] Audio cuts out

[Andrei (MPI)]

Gary,

I would check echo cancelling parameters first. I've seen this to happen 
with one of the zaptel echo cancellers. Try to change the default echo 
algorithm in zconfig.h,  and recompile and install new zaptel. Also 
zapata.conf echo parameters may need to be changed either way.

Andrei

Gary Richardson wrote:
> 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 <http://1.2.7.1> install. ztdummy is loaded.
>
> Does anyone have any insite into this problem?
>
> Thanks.
> ------------------------------------------------------------------------
>
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