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tzafrir.cohen at xorco...
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 Posted: Thu Jul 03, 2008 8:02 am Post subject: [asterisk-users] problem in making call pc to phone & vi |
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Hi
On Thu, Jul 03, 2008 at 06:21:27PM +0530, Bikrish Amatya wrote:
| 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
|
You have two ports. Which of those is connected?
| Quote: |
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
|
Only the channels of ports 1 are configured in Asterisk?
| Quote: | 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 |
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lyle at lcrcomputer.net
Guest
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| Posted: Thu Jul 03, 2008 8:20 am Post subject: [asterisk-users] problem in making call pc to phone & vi |
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|
Your E1 links are down. (red alarm) Your card does not like or see your
providers E1.
Lyle
Bikrish Amatya wrote:
| 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
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