Bikrish Amatya
2008-Jul-03 12:51 UTC
[asterisk-users] problem in making call pc to phone & vice versa
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
Tzafrir Cohen
2008-Jul-03 13:02 UTC
[asterisk-users] problem in making call pc to phone & vice versa
Hi On Thu, Jul 03, 2008 at 06:21:27PM +0530, Bikrish Amatya 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=47You have two ports. Which of those is connected?> > > 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=>1-15,17-31Only the channels of ports 1 are configured in Asterisk?> 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[snip] It is actually in use by Asterisk. It is also in RED alarm. That is: no layer 1 connection to the remote side. One possible reason for thaat is that there's nothing connected to that port. No point trying to call through this port. What do you have in /proc/zaptel/2 ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
Lyle Giese
2008-Jul-03 13:20 UTC
[asterisk-users] problem in making call pc to phone & vice versa
Your E1 links are down. (red alarm) Your card does not like or see your providers E1. Lyle Bikrish Amatya 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 > [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 > > > > > _______________________________________________ > -- 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 >