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

William K. Volkman wkvsf at users.sourceforge.net
Tue Nov 29 03:34:06 MST 2005


Hello,
On Tue, 2005-11-29 at 02:25, Rich Adamson wrote:
> Well... I don't have an ADIT box around, so can't help on that.
> 
> Do take a close look at the channel assignment stuff, both in zaptel.conf
> and zapata.conf. Are you absolutely sure the ordering of the cards
> and channels are right (haven't moved any cards around or removed any)?
> Your statement "it wasn't until I changed the connection to span 2 that 
> it started allowing inbound calls to work" suggests the ordering of
> the channels might not be what you are expecting.
> 
> You have channels 25-27 defined in zapata.conf, but they are shown as
> unused in zaptel.conf. (I did not try to match up all the other ones.)

Sorry, I had also make the requisite changes in zaptel.conf:
span=1,0,0,esf,b8zs,yellow
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
fxsks=1-8
unused=9-16
unused=17-24
fxsks=25-48
unused=49-72,73-96
fxsks=97
fxoks=98-101
fxoks=102-105
loadzone = us
defaultzone=us

> 
> Take a close look at the "group=" definitions below. First set to
> group=1, then six lines below that its group=0. Are you calling out
> with an extensions.conf entry like Zap/g1? And, are all the channels
> that are included in "g1" actually connected/usable? (eg, be carefull
> with assumptions about what happens when a channel is included in the
> group definition but the associated ADIT port isn't connected to 
> anything.) Instead of using Zap/g1, prove to yourself things are
> configured correctly by sending calls to Zap/99 (or whichever channel
> you have connected to a real line), and do that for each fxo line
> that you think is wired/working.

Yes the calls out are/were to Zap/g1/xxxxxxx, changing them to
the specific Zap channels makes no difference.  I just now
tried adding "w" to the dial stream, no effect.   Discovered
that my new test-set shows DTMF digits, hooked it up and
I'm seeing only the first digit of the phone number being
sent on the outgoing line (the reason for the "Call didn't
go through" message).  Any ideas where next to look?

> Might look at 'zap show status' and 'zap show channels' to ensure
> what your expecting is what is defined.

Is "show status" a asterisk 1.2 command?
*CLI> zap show status
No such command 'zap show status' (type 'help' for help)
*CLI> zap show channels
   Chan Extension  Context         Language   MusicOnHold
 pseudo            internal                   default
     25            incoming                   default
     26            incoming                   default
     27            incoming                   default
     97            incoming                   default
     98            internal                   default
     99            internal                   default
    100            internal                   default
    101            internal                   default
    102            internal                   default
    103            internal                   default
    104            internal                   default
    105            internal                   default
*CLI>

> RED/NOP: RED generally means the T1 port is not seeing any timing
> signals (eg, nothing is connected to it). NOP generally mean 
> Not-OPerational.

When the cable is connected to span1 the RED goes away but it
stays in "NOP".

> Not sure why T1 port #1 on the card didn't work. Could be a bad port
> or the channel #'s aren't as you expect. You can test for a bad port
> by creating a T1 crossover cable, and send test calls out one T1 and 
> receive those calls on another T1 (on the same card).

I may try this tomorrow, I've got about another 1/2 hour before
I have to revert the system to original/working configuration.

> Last, any changes made to zapata.conf requires a complete restart of
> asterisk (not just a reload).

That I knew and have been doing...

>  And, any changes to zaptel.conf requires
> a reload of the zaptel drivers.
> 
I thought that running ztcfg was sufficient.  In any case I've
got scripts that rmmod the modules and modprobe them before
starting up asterisk.

> Rich
> 

Thanks btw. for that informative explanation of the loopstart
v.s. groundstart signalling, could I suggest that information
would be useful on the voip-info.org wiki (If not already there,
I found some useful information tucked in unrelated topics).
And I'm in the US using Qwest POTS lines so loopstart it is.

> ------------------------
> 
> > 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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