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 <[EMAIL PROTECTED]> 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 > > > > > │ > 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 > [EMAIL PROTECTED]: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 > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users