[Asterisk-Users] Audio cuts out

[Steve Totaro]

Recording many simultaneous calls can cause this behavior too.

Thanks,
Steve Totaro

Gary Richardson wrote:
> We're not using any zaptel hardware though. I didn't think the echo 
> cancellers would be doing anything? We're digital and sip from end to 
> end. Do I need to disable echo cancellation in some way?
>
> Thanks.
>
> On 6/12/06, *Andrei (MPI)* <asterisk at markovprocesses.com 
> <mailto:asterisk at markovprocesses.com>> wrote:
>
>     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> <http://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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