[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
[EMAIL PROTECTED]:1]
Dial("SIP/bikrish-09b21980",
"Zap/g1/600833") 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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[asterisk-users] (no subject)
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/g1/600833") 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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Register Now: http://www.astricon.net
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[asterisk-users] connectivity with oracle database and astreisk
Hi all In my company there is oracle database which has the information about the client. Now my requirement is... when my clients calls to our company .. they should be able to get information about them when they call to our pbx. I mean how can reterive information from oracle database and play it to clients . Thanks in advance Bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg recording of calls
Hi all I appreciate the help that you have given me on call recording. I would like to share how i achieve the way i wanted. I used monitor and soxmix for this. First i used monitor to record the calls and made use of system command to create directory of each extension and inside each directory of respecitve extension created a sound file name with soxmix to create sound file in data-time format and i gave ftp access to the person of the recording directory. Now he can go have ftp access and go to each directory access the sound file by data and time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reg call recording
Hi all I am using asterisk as pbx for my company. My company has requirement that all the incoming and outgoing calls should be recorded for all the extensions and should be able to play recorded call on extensions basis, that is , say 123 extension has made what call on the particular date and should be able to play and listen to it. What is the better way to achieve this? Any kind of suggestion is truly appreciated. Bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
