[asterisk-users] Problem with Asterisk + TDM410 FXO
Paul Hales
pdhales at optusnet.com.au
Thu May 14 00:18:28 CDT 2009
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.
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?
You could also buy some IP phones or put softphones around. That would
solve the problem (you said that a softphone worked OK)
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
More information about the asterisk-users
mailing list