[Asterisk-Users] outgoing call routing
Jose Vicente Ortega
jvortega at txwes.edu
Mon Jun 20 11:58:17 MST 2005
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: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] En nombre de Jose
>Vicente Ortega
>Enviado el: domingo, 19 de junio de 2005 19:26
>Para: asterisk-users at 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|817XXXXXX") 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=20050619-101044-200-817XXXXXX") in new stack
> -- Executing Monitor("Zap/1-1",
>"wav|20050619-101044-200-817XXXXXX|mb") in new stack
> -- Executing GotoIf("Zap/1-1", "0?4") in new stack
> -- Executing SetCallerID("Zap/1-1", "817XXXXXX") 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=817XXXXXX") 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/g0/817XXXXXX") in new stack
> -- Called g0/817XXXXXX
> -- Hungup 'Zap/4-1'
>
>
> -- Executing Macro("SIP/302-ffef", "dialout-trunk|1|817XXXXXX") 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=20050619-101314-302-817XXXXXX") in new stack
> -- Executing Monitor("SIP/302-ffef",
>"wav|20050619-101314-302-817XXXXXX|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=817XXXXXX") in new
>stack
> -- Executing SetVar("SIP/302-ffef", "DIAL_TRUNK=1") in new stack
> -- Executing AGI("SIP/302-ffef", "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("SIP/302-ffef", "ZAP/g0/817XXXXXX") in new stack
> -- Called g0/817XXXXXX
> -- Zap/1-1 is ringing
> -- Hungup 'Zap/1-1'
>
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