Hey all,

I'm having problems where there is significant static when making SIP -> PSTN calls. SIP -> SIP and SIP -> VM calls are totally clear and fine.

Here's the setup:

Polycom 601,501, and ten 301s.
Digum 2400 TDM card w/echo cancelling, 12 FXO ports.
The TDM card is on IRQ 5 with nothing else on it.

Server Specs:
Asus P4P800E Deluxe
P4 3.0 Ghz
1 GB Ram
80 GB SATA HD

- There is no static when using a normal phone direct to the 66 block.
- The sound is also a bit low, and bumping the volume on the Polycom phones makes the static alot worse (obviously)

zapata.conf settings:
[channels]

language=en
context=from-pstn
signalling=fxs_ks

usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=128
echocancelwhenbridged=yes
echotraining=500
rxgain=6
txgain=3
group=0
callgroup=1
pickupgroup=1
immediate=yes
faxdetect=no


zaptel.conf settings:

fxsks=17-24
loadzone        = us
defaultzone     = us



- Running Fedora Core 4 - Kernel 2.6.14-1.1653_FC4smp
- USB is completely disabled.

cat /proc/interrupts:
CPU0 0: 3115334 XT-PIC timer
 1:          8          XT-PIC  i8042
 2:          0          XT-PIC  cascade
 5:   12453626          XT-PIC  wctdm24xxp
 8:          1          XT-PIC  rtc
10:      93751          XT-PIC  libata
11:     907892          XT-PIC  SysKonnect SK-98xx, eth1
15:     111542          XT-PIC  ide1
NMI:          0
LOC:    3115228
ERR:          0
MIS:          0





Any other information you need to help me figure this out, please let me know.

- Roman

_______________________________________________
--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

Reply via email to