hi i had installed asterisk, i am using Sangoma USB FXO (u100) . i had made a very simple dial plan. i want that calls coming to pstn should be received by extensions which are made in sip and extentions.conf. heres the configuration files
1 /etc/zaptel.conf
2 /etc/asterisk/zapata.conf
3 /etc/asterisk/zapata.confc
4 /etc/asterisk/sip.conf
5 /etc/asterisk/extensions.conf
#######################################################
vi /etc/zaptel.conf
#autogenerated by /usr/sbin/wancfg_zaptel do not hand edit
#autogenrated on 2010-02-09
#Zaptel Channels Configurations
#For detailed Zaptel options, view /etc/zaptel.conf.bak
loadzone=us
defaultzone=us
#Sangoma USB U100 [bus:2-3 span:1]
fxsks=1
fxsks=2
######################################################
vi /etc/asterisk/zapata.conf
;autogenerated by /usr/sbin/wancfg_zaptel do not hand edit
;autogenrated on 2010-02-09
;Zaptel Channels Configurations
;For detailed Zaptel options, view /etc/asterisk/zapata.conf.bak
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma AU100 [slot:0 bus: span:1]
context=from-zaptel
group=0
signalling = fxs_ks
channel => 1
context=from-zaptel
group=0
signalling = fxs_ks
channel => 2
#######################################################
vi /etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
[2001]
type=friend
context=my-phones
secret=1234
host=dynamic
######################################################
vi /etc/asterisk/extensions.conf
[others]
[from-zaptel]
exten => _X.,1,Dial(SIP/2000)
exten => _X.,n,Playback(thank-you)
exten => _X.,n,Hangup
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => _X.,1,Dial(Zap/1-1/${EXTEN})
exten => _X.,n,Hangup
######################################################
here is the message when i call to PSTN from mobile
[root@localhost ~]# asterisk -vvvvvr
Asterisk 1.4.29, Copyright © 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
== Parsing ‘/etc/asterisk/asterisk.conf’: Found
== Parsing ‘/etc/asterisk/extconfig.conf’: Found
Connected to Asterisk 1.4.29 currently running on localhost (pid = 2312)
Verbosity is at least 5
– Starting simple switch on ‘Zap/1-1’
[Jan 2 06:18:01] NOTICE[12066]: chan_dahdi.c:6850 ss_thread: Got event 18 (Ring Begin)…
[Jan 2 06:18:01] NOTICE[12066]: chan_dahdi.c:6850 ss_thread: Got event 17 (Polarity Reversal)…
[Jan 2 06:18:01] NOTICE[12066]: chan_dahdi.c:6850 ss_thread: Got event 18 (Ring Begin)…
== Starting Zap/1-1 at from-zaptel,s,1 failed so falling back to exten ‘s’
== Starting Zap/1-1 at from-zaptel,s,1 still failed so falling back to context ‘default’
[Jan 2 06:18:01] WARNING[12066]: pbx.c:2454 __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’
hope some 1 will help me out n tell where i had done mistake.