[Asterisk-Users] Problem with ADIT 600 and FXO configuration

C F shmaltz at gmail.com
Mon Nov 28 11:21:09 MST 2005


What does the TE406 leds indicate?

On 11/28/05, William K. Volkman <wkvsf at users.sourceforge.net> wrote:
> I've looked through the archives of the mailing list for the
> last year and although informative I've not been successful
> at get this to work.  We had a working Asterisk PBX system
> with 3 Digium X101P FXO lines and two TDM400P FXS cards.
> I've setup an ADIT 600 with an 8 port FXO card (and an
> 8 port FXS card not currently installed).  We are going
> to be adding a T1 for incoming calls this week. I removed
> two of the X101P cards and installed a TE406P.  I'm using
> Asterisk 1.0.9 (and matching zaptel, libpri) from tar files.
>
> /etc/zaptel.conf has this configuration:
> span=1,1,0,esf,b8zs,yellow
> span=2,0,0,esf,b8zs
> span=3,0,0,esf,b8zs
> span=4,0,0,esf,b8zs
> #Modular unit, first card is FXO
> fxsks=1-3
> unused=4-8
> #Modular unit, 1 FXS cards
> unused=9-16
> unused=17-24
> unused=25-48,49-72,73-96
> fxsks=97
> fxoks=98-101
> fxoks=102-105
>
> /etc/asterisk/zapata.conf has this:
> group = 0
> signalling=fxs_ks
> context = incoming
> busydetect = yes
> overlapdial = no
> channel => 1-3
>
> signalling=fxs_ks
> channel => 97  ;X100P
>
> group = 1
> signalling = fxo_ks
> context = internal
> ;TDM400P
> callerid = "Available" <200>
> channel => 98-100
> callerid = "xxxxx"
> channel => 101
> ;TDM400P
> callerid = "xxxxx"
> channel => 102
> callerid = "xxxxx"
> channel => 103
>
> Parts of my adit configuration:
> -Setting slot a.
>
> set a:1 up
> set a:1 fdl none
> set a:1 lbo 4
> set a:1 framing esf
> set a:1 id "Inbound"
> set a:1 linecode b8zs
> set a:1 loopdetect csu
> set a:1:1-24 side drop
> set a:1:1-24 type voice
> set a:1:1-24 signal ls
> set a:2 up
> set a:2 fdl none
> set a:2 lbo 1
> set a:2 framing esf
> set a:2 id "Outbound PBX"
> set a:2 linecode b8zs
> set a:2 loopdetect csu
> set a:2:1-24 side drop
> set a:2:1-24 type voice
> set a:2:1-24 signal ls
>
> -Setting slot 1.
>
> set 1:1-8 signal lscpd
> set 1:1-8 txgain -3
> set 1:1-8 rxgain -6
>
> -Setting primary and secondary clock sources.
>
> set clock1 a:1
> set clock2 internal
>
> -Setting the system idle pattern for DS0s.
>
> set idle 0xff
>
> -Making connections.
>
> connect a:2:1-3 1:1-3
>
> Inbound calls just ring and ring (the leds on the ADIT change
> state) however asterisk doesn't respond.  Attempts to make
> outgoing calls get:
>     -- Executing Dial("SIP/202-ba07", "Zap/g0/5551212") in new stack
> Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create
> channel of type 'Zap'
>   == Everyone is busy/congested at this time
>     -- Executing Congestion("SIP/202-ba07", "") in new stack
>   == Spawn extension (from-sip, 95942060, 3) exited non-zero on
> 'SIP/202-ba07'
>     -- Executing Hangup("SIP/202-ba07", "") in new stack
>
> I've tried just about all combinations of gs/ls/ks for the
> signalling to no avail.  Here is the output of status:
>
> > status a:2:1-3
>
> DS0         Rx AB  Tx AB  Signal  T1                 TP
> ---         -----  -----  ------  -----------------  --
> a:2:1         01     01      LS   Traffic            N
> a:2:2         01     01      LS   Traffic            N
> a:2:3         01     01      LS   Traffic            N
>
> > status 1:1-3
>
> FXO    Rx AB  Tx AB  Signal=>T1 Sig  T1                 TP
> ---    -----  -----  --------------  -----------------  --
> 1:1      01     01   LSCPD => LS     Traffic             N
> 1:2      01     01   LSCPD => LS     Traffic             N
> 1:3      01     01   LSCPD => LS     Traffic             N
>
> > show connect a:2:1-3
>     From            Desc                    Desc            To
>  -----------  ------------------      -----------------  ---------
>   A:02:01        LS VOICE   DS0 <--> FXO    VOICE LSCPD   1:01
>   A:02:02        LS VOICE   DS0 <--> FXO    VOICE LSCPD   1:02
>   A:02:03        LS VOICE   DS0 <--> FXO    VOICE LSCPD   1:03
>
> Can anyone spot what I've got wrong?  Any suggestions or hints
> welcome.
>
> Thanks,
> William.
>
>
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