Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line
connected.
I am new in Linux and Asterisk, my steps are theese:
1. Install CentOS 4.4 (basic instalation).
2. Command line:
yum -y update
yum install gcc kernel-devel bison openssl-devel
yum install openssl-devel
3. Download the source:
wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
4. Uncompress:
tar xvfz asterisk-1.2.17.tar.gz
tar xvfz zaptel-1.2.16.tar.gz
5. Compile:
cd zaptel-1.2.16
make clean
make
make install
cd ..
cd asterisk-1.2.17
make clean
make
make install
make samples
make config
Mi configuration files:
zaptel.com
loadzone=es
defaultzone=es
fxsks=1
zapata.conf
[channels]
signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512(64ms),1024(128ms)
echocancel=yes
echotraining=yes
echocancelwhenbridged=no
rxgain=0
txgain=0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
;busydetect=yes
;busycount=10
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
polarityonanswerdelay=600
;callprogress=no
progzone=es
channel => 1
sip.conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[101]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
[102]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
extensions.conf
[general]
static=yes
writeprotect=yes
;autofallthrough=yes
;clearglobalvars=no
;priorityjumping=no
[SOME]
exten => 101,1,Dial(SIP/101,30,Ttm)
exten => 101,2,Hangup
exten => 102,1,Dial(SIP/102,30,Ttm)
exten => 102,2,Hangup
[incoming]
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,Dial(SIP/101,30,Ttm)
[outgoing]
exten =>_9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =>_9XXXXXXXX,2,Hangup()
exten =>_9XXXXXXXX,102,Hangup()
Command line:
modprobe zaptel
modprobe wcfxo
modprobe wctdm
Then I start Asterisk (asterisk -vvvc), and when I call to the analog line
number, the console shows that:
*CLI> -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 26 19:34:33 WARNING[3818]: pbx.c:2377 __ast_pbx_run: Channel
'Zap/1-1'
sent into invalid extension 's' in context 'default', but no
invalid handler
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Apr 26 19:34:38 NOTICE[3821]: chan_zap.c:6223 ss_thread: Got event 18 (Ring
Begin)...
Apr 26 19:34:40 NOTICE[3821]: chan_zap.c:6223 ss_thread: Got event 2
(Ring/Answered)...
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 26 19:34:40 WARNING[3821]: pbx.c:2377 __ast_pbx_run: Channel
'Zap/1-1'
sent into invalid extension 's' in context 'default', but no
invalid handler
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Apr 26 19:34:47 NOTICE[3824]: chan_zap.c:6223 ss_thread: Got event 18 (Ring
Begin)...
Apr 26 19:34:49 NOTICE[3824]: chan_zap.c:6223 ss_thread: Got event 2
(Ring/Answered)...
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 26 19:34:49 WARNING[3824]: pbx.c:2377 __ast_pbx_run: Channel
'Zap/1-1'
sent into invalid extension 's' in context 'default', but no
invalid handler
-- Hungup 'Zap/1-1'
The call doesn't ring, I want to redirect to extension 101.
Thank you very much for your time.
See you,
Josu Lazkano
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