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.
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
