[asterisk-users] Problem with Asterisk + TDM410 FXO

[Alex Samad]

Hi

I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.

I borrowed a spa9000 to place between the PABX and the PSTN lines. I
have had this going for a while (>5 months) and it has been working fine
(some issues with echo and other minor things), which is why I am moving
to asterisk.

I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
and used just in case the internet connection is down.

I have tested the pstn line connection with a soft phone and it seems to
be working fine. I need some help on how to tell asterisk to ignore the
line for incoming !

when I connect the PABX to the FXO ports I ran into a problem.

It seems to register okay, I pick up the handset on the pabx and select
line 1 and i can hear a dial tone (same with line2) - this is the same
what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
use.

But I can't hear anything from the pabx - no dtmf tones and thus can't
dial!

when I try dialing in from the internet to asterisk then to ZAP/g1 the
pabx can see the ring and I can pick up the phone I can hear the other
end, but they can't hear me.

I don't believe its a firewall issue as I can't dial from the pabx

okay some print outs

# zaptel_hardware 
pci:0000:05:02.0     wctdm24xxp+  d161:8005 Wildcard TDM410P

# ztcfg -vv

Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels to configure.

# cat /etc/zaptel.conf 
fxsks=4
fxoks=1,2,3

loadzone=au
defaultzone=au

/etc/asterisk/zapata.conf
========================
# grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ks
rxwink=300		; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
Group=1
signalling=fxo_ks
context=in-pbx
channel=1-2
Group=2
echocancel=yes
signalling=fxs_ks
context=in-pstn
channel=4
Group=3
signalling=fxo_ks
context=in-spare
channel=3


the thing that has me beet is that it work with the spa9000 I would
expect it to just sort of work with the digium card.

the os is debian amd64 2.6.26
#dpkg -l asteri* | grep ^ii
ii  asterisk                                    1:1.4.21.2~dfsg-3
Open Source Private Branch Exchange (PBX)
ii  asterisk-barbarast.com                      0.0.0-1
asterisk setup for hme1.samad.com.au
ii  asterisk-doc                                1:1.4.21.2~dfsg-3
Source code documentation for Asterisk
ii  asterisk-sounds-extra                       1.4.7-1
Additional sound files for the Asterisk PBX
ii  asterisk-sounds-main                        1:1.4.21.2~dfsg-3
Core Sound files for Asterisk (English)

#dpkg -l zapt* | grep ^ii
ii  zaptel                                      1:1.4.11~dfsg-3
zapata telephony utilities
ii  zaptel-modules-2.6.22-2-amd64               1:1.4.11~dfsg-3+2.6.22-4
zaptel modules for Linux (kernel 2.6.22-2-am
ii  zaptel-modules-2.6.26-2-amd64
1:1.4.11~dfsg-3+2.6.26-15   zaptel modules for Linux (kernel 2.6.26-2-am
ii  zaptel-source


thanks
Alex

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