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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