Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] problem in making call pc to phone & vice versa


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
bikrish at w2sindia.com
Guest





PostPosted: Thu Jul 03, 2008 7:51 am    Post subject: [asterisk-users] problem in making call pc to phone & v Reply with quote

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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services