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)* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> 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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