[asterisk-users] problem with A400P01 OpenVox

[Josu Lazkano]

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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