PSTN Line->DAHDI->Asterisk->AGI->Dial SIP/201, extension…
=============Logs==================
== Starting DTMF CID detection on channel 2
– Starting simple switch on ‘DAHDI/2-1’
– Executing [s@from-pstn:1] NoOp(“DAHDI/2-1”, “”" <919544678732>") in new stack
– Executing [s@from-pstn:2] NoOp(“DAHDI/2-1”, “”) in new stack
– Executing [s@from-pstn:3] Monitor(“DAHDI/2-1”, “wav,myfile1”) in new stack
– Executing [s@from-pstn:4] AGI(“DAHDI/2-1”, “incoming_manager.pl”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/incoming_manager.pl
– AGI Script Executing Application: (NoOp) Options: (Callerd==919544678732==SessioInd==1383221271.16==DialedNumber==)
– AGI Script Executing Application: (Answer) Options: ()
– <DAHDI/2-1> Playing ‘demo-thanks.gsm’ (language ‘en’)
– AGI Script Executing Application: (NoOp) Options: (============================================================201)
– AGI Script Executing Application: (Dial) Options: (SIP/201,tTwWir)
== Using SIP RTP CoS mark 5
– Called SIP/201
[Oct 31 17:38:01] WARNING[4999][C-0000000a]: app_dial.c:2642 dial_exec_full: Invalid timeout specified: ‘tTwWir’. Setting timeout to infinite
– SIP/201-00000006 is ringing
– AGI Script Executing Application: (Hangup) Options: (16)
– <DAHDI/2-1>AGI Script incoming_manager.pl completed, returning 4
== Spawn extension (from-pstn, s, 4) exited non-zero on ‘DAHDI/2-1’
– Hanging up on ‘DAHDI/2-1’
– Hungup ‘DAHDI/2-1’
===============chan_dahdi.conf==============================
[general]
[trunkgroups]
[channels]
#include dahdi-channels.conf
;;[901]
;FXO Modules
usecallerid=yes
cidsignalling=dtmf
cidstart=dtmf
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
canpark=yes
cancallforward=yes
txgain=0.0
busydetect=yes
busycount=6
immediate=yes
callprogress=yes
toneduration=300
;relxdtmf=yes
;progzone=in
rxwink=300
INPUT_USER=300
faxdetect=OFF
musiconhold=default
switchtype=national
hanguponpolarityswitch=yes
ringtimeout=1000
polarityonanswerdelay=300
group=0
signalling=fxs_ks
context=from-pstn
channel=>2
group=5
echocancel=yes
signalling=fxo_ks
context=from-local
channel=>4
======dahdi-channel.conf=============
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default
;;; line="2 WCTDM/0/1 FXSKS (In use) (EC: MG2 - INACTIVE)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default
===========================extension.conf======================
[from-pstn]
include =>transfer
exten => s,1,NoOp(${CALLERID(all)})
exten => s,2,NoOp(${SESSIONID})
exten => s,3,Monitor(wav,myfile1)
exten => s,4,AGI(incoming_manager.pl)
[transfer]
exten => *1,1,Monitor(wav,myfile)
=====================================sip.conf=====================
[101]
username=101
secret=1234
type=Friend
callerid=101
host=Dynamic
nat=no
auth=Md5
dtmfmode=auto
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
Callgroup=1
pickupgroup=1-9,13
context=from-internal
[201]
username=201
secret=1234
type=Friend
callerid=201
host=Dynamic
nat=no
auth=Md5
dtmfmode=auto
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
Callgroup=1
pickupgroup=1-9,13
context=from-internal
===================================feature.conf======================
[general]
parkext=> 700
;parkext_exclusive=yes ; Specify that the parkext created for this parking lot
; will only access this parking lot. (default is no)
parkpos=> 101-102 ; What extensions to park calls on. (defafult parking lot)
[featuremap]
blindxfer=> #1 ; Blind transfer (default is #) – Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect=> *0 ; Disconnect (default is *) – Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon=> *6 ; One Touch Record a.k.a. Touch Monitor – Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer=> *8 ; Attended transfer – Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall=> #72 ; Park call (one step parking) – Make sure to set the K and/or k option in the Dial() app call!
automixmon=> *3 ; One Touch Record a.k.a. Touch MixMonitor – Make sure to set the X and/or x option in the Dial() or Queue() app call!
[applicationmap]
======================ISSues===Facing=============
Dear David…
if made a call from mobile–to–pstn–call–reached–in dahdi–and–hit asterisk–context–
then–after playing a prompt—dia the extension SIP/201—
1.If i disconnected the call from side ie from my mobile call still ringing the extension for two three rings …
2.I can’t use the pbx features like blindtransfer,atxtransfer,automonitor…if i am puching some
dtmf it not taking the asterisk…not saying pbx transfer and all…
.3…But all these features are working if am making a call from sip to sip extension…
So How to solve this issues and how can i use this features please help me David…
Becasue I want to use blindtransfer and automonitor for incoming calls from Pstn line…i have a land connection…
4.But its happening if am put the on hold in hard phone…its taking the dtmf…
but automonito will never work for that if am pressing the *1 to start monitoring …it will suddenly created the file without monitoring the call AS I released the call from Hold