On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote: > Alex Samad wrote: > > On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote: > > > >> I think you have your line types mixed up - FXS is for phones, FXO is > >> for lines. > >> > > > > 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). > > > > > > > By reading your configs below, you could be right - ports 1,2 and 3 are > FXS, while 4 is FXO.
fingers crossed I am. Pretty sure I am, like I said I have been able to make out going calls on the pstn line (although coming to think about it I haven't actually tried talking ..... > > What happens if you make a call in from the old fax line and send that > over to the old PABX? Does that work OK? not sure what you are asking here. I have checked an incoming call through the FXO(PSTN) through to a FXS port (pabx) > > You could also buy some IP phones or put softphones around. That would > solve the problem (you said that a softphone worked OK) I have bought a snom to trial, but I don't want to make tooooo many changes in one time. The annoying thing is the spa9000 can talk to it with its fxs ports :( Alex > > PaulH > > > > > >> 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 > >> > >> > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- 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 > -- "They didn't think we were a nation that could conceivably sacrifice for something greater than our self; that we were soft, that we were so self-absorbed and so materialistic that we wouldn't defend anything we believed in. My, were they wrong. They just were reading the wrong magazine or watching the wrong Springer show." - George W. Bush 03/12/2002 Washington, DC
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