Here is is.
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
;Include genzaptelconf configs
#include zapata-auto.conf
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1
signalling=fxo_ks
; Note: this is an extension. Create a ZAP extension in AMP for Channel 1
channel = 1
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn
channel = 4
At 10:10 AM 6/20/2005, Sergio Serrano wrote:
Please,
send us zapata.conf. It's possible that you don't have well
configure zapata.conf, because in your trace you try to dial through g0
group and your Zap/4(I understand is your Zap connected to PSTN) must be
into the 0 group.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jose
Vicente Ortega
Enviado el: domingo, 19 de junio de 2005 19:26
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] outgoing call routing
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip
extensions and a regular phone connected to the box. All routing works
fine
from the regular phone connected to the box, whether its going to FWD,
broadvoice or the PSTN. The problem I am experiencing comes from making
calls from the sip phones. They get routed correctly to the sip and iax
trunks but when making calls that are routed to the zap channel they
ring
the regular phone and do not get routed to the PSTN.
Below are examples of the verbose from asterisk for calls from internal
zap
and internal sip channels to the PSTN.
-- Starting simple switch on 'Zap/1-1'
-- Executing Macro(Zap/1-1, dialout-trunk|1|817XX) in new
stack
-- Executing Macro(Zap/1-1, record-on|200) in new stack
-- Executing AGI(Zap/1-1, set-timestamp.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi
-- AGI Script set-timestamp.agi completed, returning 0
-- Executing SetVar(Zap/1-1,
CALLFILENAME=20050619-101044-200-817XX) in new stack
-- Executing Monitor(Zap/1-1,
wav|20050619-101044-200-817XX|mb) in new stack
-- Executing GotoIf(Zap/1-1, 0?4) in new stack
-- Executing SetCallerID(Zap/1-1, 817XX) in new stack
-- Executing Goto(Zap/1-1, 6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetCallerID(Zap/1-1, ) in new stack
-- Executing SetGroup(Zap/1-1, OUT_1) in new stack
-- Executing CheckGroup(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, DIAL_NUMBER=817XX) in new
stack
-- Executing SetVar(Zap/1-1, DIAL_TRUNK=1) in new stack
-- Executing AGI(Zap/1-1, fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Dial(Zap/1-1, ZAP/g0/817XX) in new stack
-- Called g0/817XX
-- Hungup 'Zap/4-1'
-- Executing Macro(SIP/302-ffef, dialout-trunk|1|817XX) in new
stack
-- Executing Macro(SIP/302-ffef, record-on|302) in new stack
-- Executing AGI(SIP/302-ffef, set-timestamp.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi
-- AGI Script set-timestamp.agi completed, returning 0
-- Executing SetVar(SIP/302-ffef,
CALLFILENAME=20050619-101314-302-817XX) in new stack
-- Executing Monitor(SIP/302-ffef,
wav|20050619-101314-302-817XX|mb) in new stack
-- Executing GotoIf(SIP/302-ffef, 1?4) in new stack
-- Goto (macro-dialout-trunk,s,4)
-- Executing Goto(SIP/302-ffef, 6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetCallerID(SIP/302-ffef, ) in new stack
-- Executing SetGroup(SIP/302-ffef, OUT_1) in new stack
-- Executing CheckGroup(SIP/302-ffef, ) in new stack
-- Executing SetVar(SIP/302-ffef, DIAL_NUMBER=817XX) in new
stack
-- Executing SetVar(SIP/302-ffef, DIAL_TRUNK=1) in new stack
