[asterisk-users] (no subject)

[C F]

The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P

As for your problem looks like you are trying to use the wrong span
for dial out.


On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya <bikrish at w2sindia.com> wrote:
>
>
> 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
>
>
>
>
> &#9474;
> Alarms
> Span
> &#9474;
>
> &#9474;
> RED
> T2XXP (PCI) Card 0 Span
> 1
>
>
> &#9474;
> OK
> T2XXP (PCI) Card 0 Span
> 2
>
>
> &#9474;
>
>
>
> 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
>
>
>
>
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