I think you have your line types mixed up - FXS is for phones, FXO is for lines.
An analogue passthorugh setup _is_ doable, just not overly recommended. PaulH Alex Samad wrote: > 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 > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
