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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090514/ef2eea46/attachment.pgp
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
On Thu, May 14, 2009 at 07:46:26AM +0200, Marco Sambo wrote:> FXO channels shuld have FXS signalling, and FXS channels shuld have FXO > signalling, so: > > # FXO channels are 1,2,3 > fxsks=1,2,3 > # FXS channel is 4 > fxoks=4yep turned it around and tested it out, worked, had to fxs tune to get the fxs channel working.> > > > > > > > sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is > > > that a attached fxs presents internally as a fxo > > > > > > I have a pstn line attached to the FXO and I have my pabx attached to > > > 2 FXS ports, which signal as fxo into asterisk (I could be wrong about > > > that). > > > > >>> # cat /etc/zaptel.conf > > >>> fxsks=4 > > >>> fxoks=1,2,3 > >> _______________________________________________ > -- 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-- The misnaming of fields of study is so common as to lead to what might be general systems laws. For example, Frank Harary once suggested the law that any field that had the word "science" in its name was guaranteed thereby not to be a science. He would cite as examples Military Science, Library Science, Political Science, Homemaking Science, Social Science, and Computer Science. Discuss the generality of this law, and possible reasons for its predictive power. -- Gerald Weinberg, "An Introduction to General Systems Thinking" -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090515/e5d194a3/attachment.pgp