Setup X100P card for outgoing calls

[lexer]

Hi

I'm newbie on asterisk, I installed a Asterisk 11.8.1 and a X100P Card OEM, I'm trying to configure card for PSTN outgoing calls (Incoming calls are working), The card was detected by asterisk, here the configuration:

dahdi_genconf -v

Default parameters from /etc/dahdi/genconf_parameters
Generating /etc/dahdi/assigned-spans.conf
Generating /etc/dahdi/system.conf
Generating /etc/asterisk/dahdi-channels.conf

nano /etc/dahdi/system.conf

# Autogenerated by /usr/sbin/dahdi_genconf on Wed Mar 12 16:27:08 2014
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
fxsks=1
echocanceller=mg2,1

# Global data

loadzone = us
defaultzone = us

nano /etc/asterisk/dahdi-channels.conf

; Autogenerated by /usr/sbin/dahdi_genconf on Wed Mar 12 16:27:08 2014
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
;;; line="1 WCFXO/0/0 FXSKS (In use) (EC: MG2 - INACTIVE)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default


dahdi show status

*CLI> dahdi show status
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
Generic Clone Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)

*CLI>dahdi show channel 1
Channel: 1
Description:
File Descriptor: 9
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID:
Calling TON: 0
Caller ID subaddress:
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Offhook

When I try to dial an external number from the sip extension 1000, I receive the following error:

== Using SIP RTP CoS mark 5
[Mar 12 16:34:21] NOTICE[2435][C-0000000c]: chan_sip.c:25632 handle_request_invite: Call from '1000' (192.20.20.148:56019) to extension '92226958' rejected because extension not found in context 'from-sip'.

This is the code that added to the end of the file extensions.conf:

[from-sip]
exten => 1000,1,Dial(SIP/1000,15,Trt)
exten => 1001,1,Dial(SIP/1001,15,Trt)

[internos]
exten=> 1000,1,dial(dahdi/1,20,Trt)
exten=> 1000,1,hangup

[outgoing]

exten => _9.,1,Dial(DAHDI/1/${EXTEN:1},45)

exten => _9.,n,Hangup

[from-pstn]

exten => s,1,Dial(SIP/1000,45)

exten => s,n,Hangup

Please, any ideas?

 

[atcom]

Simple: install FreePBX to manage your asterisk system!

Or add this to from-sip:
include => outgoing

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Calgary Telephone Systems, Avaya LG Asterisk (FreePBX) VOIP & TDM