[Asterisk-Users] Audio cuts out
Gary Richardson
gary.richardson at gmail.com
Mon Jun 12 10:05:45 MST 2006
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> 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> install. ztdummy is loaded.
> >
> > Does anyone have any insite into this problem?
> >
> > Thanks.
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