[asterisk-users] (no subject)

[Bikrish Amatya]

Hello everybody


I have configures asterisk server
and i
am using TE220P digium card.  Here is the content of
the
/etc/zaptel.conf file 
###########################
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47


loadzone        = in
defaultzone     = in

############################

the content of
/etc/asterisk/zapata.conf is as follow

############################
[channels]
context=incoming
switchtype=national
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
immediate=no
callprogress=no
callerid=asreceived
group=1
channel=>1-15,17-31
#############################

output of zttool is as follow

                                                                    

                               
│    
Alarms         
Span                                              
│
                               
│    
RED            
T2XXP (PCI) Card 0 Span
1                     

                               
│    
OK             
T2XXP (PCI) Card 0 Span
2                      

                               
│                                                                 
                               


Output of  cat /prox/zaptel/1 is as follow


    Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span
1"
HDB3/CCS RED

           1
TE2/0/1/1
Clear (In use) RED
           2
TE2/0/1/2
Clear (In use) RED
           3
TE2/0/1/3
Clear (In use) RED
           4
TE2/0/1/4
Clear (In use) RED
           5
TE2/0/1/5
Clear (In use) RED
           6
TE2/0/1/6
Clear (In use) RED
           7
TE2/0/1/7
Clear (In use) RED
           8
TE2/0/1/8
Clear (In use) RED
           9
TE2/0/1/9
Clear (In use) RED
          10 TE2/0/1/10
Clear (In use) RED
          11 TE2/0/1/11
Clear (In use) RED
          12 TE2/0/1/12
Clear (In use) RED
          13 TE2/0/1/13
Clear (In use) RED
          14 TE2/0/1/14
Clear (In use) RED
          15 TE2/0/1/15
Clear (In use) RED
          16 TE2/0/1/16
HDLCFCS (In use) RED
          17 TE2/0/1/17
Clear (In use) RED
          18 TE2/0/1/18
Clear (In use) RED
          19 TE2/0/1/19
Clear (In use) RED
          20 TE2/0/1/20
Clear (In use) RED
          21 TE2/0/1/21
Clear (In use) RED
          22 TE2/0/1/22
Clear (In use) RED
          23 TE2/0/1/23
Clear (In use) RED
          24 TE2/0/1/24
Clear (In use) RED
          25 TE2/0/1/25
Clear (In use) RED
          26 TE2/0/1/26
Clear (In use) RED
          27 TE2/0/1/27
Clear (In use) RED
          28 TE2/0/1/28
Clear (In use) RED
          29 TE2/0/1/29
Clear (In use) RED
          30 TE2/0/1/30
Clear (In use) RED
          31 TE2/0/1/31
Clear (In use) RED
       
I
am
new to asterisk and googled around , configured the asterisk
server. Now
when i make a call from outside , it give me busy
tone..  and when i
call from softphone .. it shows me as show
below


       -- Executing
[9999600833 at incoming:1]
Dial("SIP/bikrish-09b21980",
"Zap/g1/9999600833") in
new stack
[Jul  3
19:14:34] WARNING[6018]: app_dial.c:1183
dial_exec_full: Unable to
create channel of type 'Zap' (cause 34 -
Circuit/channel
congestion)
  == Everyone is busy/congested at
this time
(1:0/1/0)
  == Auto fallthrough, channel
'SIP/bikrish-09b21980' status is 'CONGESTION'

I am not able
to
figure out the problem. Any kind of help would be appericiated.

Thanking you

bikrish