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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