[Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
William K. Volkman
wkvsf at users.sourceforge.net
Tue Nov 29 01:21:52 MST 2005
Hello,
OK, some things I've found out so far. The ground connection
to the ADIT chassis wasn't really to ground (fixed that, it
made FXS card happy when connected).
Taking a cue from another post I also reduced the number of
options specified in zapata.conf to:
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
group=1
callgroup=1
pickupgroup=1-2
immediate=no
musiconhold=default
group = 0
signalling=fxs_ks
context = incoming
busydetect = no
overlapdial = no
channel => 25-27
signalling=fxs_ks
channel => 97 ;X100P
group = 1
signalling = fxo_ks
context = internal
channel => 98-100
channel => 101-105
Using zttool I tried to loopback the TE406P span 1 which
switched the ADIT a:2 port into loop back, setting the line
down and back up didn't clear the configuration (I had to
find the "set a:2 line loopdown" command). Moving the link
to span 2 on the TE406P I now can receive incoming calls
(yea!), trying to place an outbound call results in
dead air with the eventual message that the call didn't
go through :-(
Note that both the ADIT and the TE406P were showing
green on the T1 connection however it wasn't until
I changed the connection to span 2 that it started
allowing inbound calls to work, zap show channel 1
showed InAlarm: 1 although I didn't spot any other
error messages.
zztool currently shows:
RED/NOP T4XXP (PCI) Card 0 Span 1
OK T4XXP (PCI) Card 0 Span 2
RED T4XXP (PCI) Card 0 Span 3
RED T4XXP (PCI) Card 0 Span 4
RED Wildcard X101P Board 1
OK Wildcard TDM400P REV E/F Board 1
OK Wildcard TDM400P REV E/F Board 2
The "NOP" on Span 1 appears to mean "Not Opened" however
I don't know what that means.
I've got one more day/night to get this working so any
suggestions are welcome.
Thank you,
William.
On Mon, 2005-11-28 at 03:28, William K. Volkman 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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