PSTN call wont hangup

Asterisk Asterisk Support
1

hello

anyone know how to hangup call from/to pstn phone.
this is the scenario…

  1. The pstn phone call the asterisk server and directed to zap phone or sip phone extensions. after the conversation the pstn phone has been hangup prior to zap/sip phone. since zap/sip is not yet hangup i noticed that the channel still established connection though the pstn phone has been hangup.

Likewise, when i call pstn phone from zap/sip phone and after conversation
where pstn phone has been hanggup first the zap/sip channel stiil established
connection.

why is this happening?

here is my extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1 ;PSTN = Channel 1-8
RECEPTIONIST=Zap/9 ;Receptionist = Channel 9
PHONE_10=Zap/10
PHONE_11=Zap/11
PHONE_12=Zap/12
PHONE_13=Zap/13
PHONE_14=Zap/14
PHONE_15=Zap/15
PHONE_16=Zap/16
PHONE_17=Zap/17
PHONE_18=Zap/18
PHONE_19=Zap/19
PHONE_20=Zap/20
PHONE_21=Zap/21
PHONE_22=Zap/22
PHONE_23=Zap/23
PHONE_24=Zap/24

SIP_1=SIP/kerberos
SIP_2=SIP/anton

[incoming]
include => internal

exten => s,1,Wait(3)
exten => s,2,Answer()
exten => s,3,Background(silence/1)
exten => s,4,BackgroundDetect(enter-ext-of-person)
exten => 1,1,Goto(internal,271,1)
exten => 1,2,Goto(incoming,s,1)
exten => 2,1,Goto(internal,251,1)
exten => 2,2,Goto(incoming,s,1)
exten => 3,1,Goto(internal,231,1)
exten => 3,2,Goto(incoming,s,1)
exten => 4,1,Goto(internal,302,1)
exten => 4,2,Goto(incoming,s,1)
exten => 5,1,Goto(internal,331,1)
exten => 5,2,Goto(incoming,s,1)
exten => 6,1,Goto(internal,351,1)
exten => 6,2,Goto(incoming,s,1)
exten => 7,1,Goto(internal,921,1)
exten => 7,2,Goto(incoming,s,1)
exten => 8,1,Goto(internal,941,1)
exten => 8,2,Goto(incoming,s,1)
exten => 0,1,Goto(internal,0,1)
exten => 0,2,Goto(incoming,s,1)
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(incoming,s,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()

[internal]
include => macro-voicemail
ignorepat => 9

exten => 0,1,Macro(voicemail,${RECEPTIONIST})
exten => 201,1,Macro(voicemail,${PHONE_10})
exten => 202,1,Macro(voicemail,${PHONE_11})
exten => 203,1,Macro(voicemail,${PHONE_12})
exten => 231,1,Macro(voicemail,${PHONE_13})
exten => 251,1,Macro(voicemail,${PHONE_14})
exten => 271,1,Macro(voicemail,${PHONE_15})
exten => 301,1,Macro(voicemail,${PHONE_16})
exten => 302,1,Macro(voicemail,${PHONE_17})
exten => 331,1,Macro(voicemail,${PHONE_18})
exten => 333,1,Macro(voicemail,${PHONE_19})
exten => 351,1,Macro(voicemail,${PHONE_20})
exten => 921,1,Macro(voicemail,${PHONE_21})
exten => 941,1,Macro(voicemail,${PHONE_22})

exten => 801,1,Answer()
exten => 801,2,Dial(${PHONE_23},30)
exten => 801,3,Playback(vm-nobodyavail)
exten => 801,4,Hangup()

exten => 123,1,Answer()
exten => 123,2,Dial(${PHONE_24},30)
exten => 123,3,Playback(vm-nobodyavail)
exten => 123,4,Hangup()

exten => 334,1,Macro(voicemail,${SIP_1})
exten => 321,1,Macro(voicemail,${SIP_2})

exten => 888,1,Answer()
exten => 888,2,VoiceMailMain(@our_voicemail)
exten => 888,3,Hangup()

exten => 8000,1,Goto(conf,1)
exten => conf,1,Set(MEETME_RECORDINGFILE=/tmp/Tutorial-$ {TIMESTAMP})
exten => conf,2,Meetme(8000|sr)
exten => conf,3,Hangup()

exten => 1337,1,Goto(count,1)
exten => count,1,MeetMecount(1234|USERS)
exten => count,2,NoOp(Total: ${USERS} users)

exten => 555,1,Directory(our_voicemail,internal,f)
exten => 444,1,Directory(our_voicemail,internal)

exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1})

[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}@our_voicemail)
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}@our_voicemail)
exten => s-BUSY,2,Goto(incoming,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)

here is my zapata.conf

[channels]
group => 1
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=8.2
txgain=1.0
context => incoming
signalling => fxs_ks
amaflags => documentation
channel => 1-8

group => 2
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=8.2
txgain=1.0
immediate=no
context => internal
signalling => fxo_ls

mailbox => 0000
callerid => “Receptionist”<number e.g 2314141>
channel => 9

mailbox => 201
callerid => “Bill” <number 201>
channel => 10

mailbox => 202
callerid => “Handel” <number 202>
channel => 11

mailbox => 203
callerid => “Gifford” <number 203>
channel => 12

mailbox => 231
callerid => “Bobot” <number 204>
channel => 13

mailbox => 251
callerid => “Fedila” <number 205>
channel => 14

mailbox => 271
callerid => "Fritz <number 206>"
channel => 15

mailbox => 301
callerid => “Ayek” <number 301>
channel => 16

mailbox => 302
callerid => “Karen” <number 302>
channel => 17

mailbox => 331
callerid => “Henry” <number 304>
channel => 18

mailbox => 333
callerid => “Kerberos” <number 305>
channel => 19

mailbox => 351
callerid => “Cris” <number 303>
channel => 20

mailbox => 801
callerid => “GYM” <number 801>
channel => 21

mailbox => 100
callerid => “Guardhouse” <number 100>
channel => 22

here is my zaptel.conf

pan=1,0,0,esf,b8zs
fxsks=1-8
fxols=9-24
loadzone = us
defaultzone=us

Please help Thanks

2

you need to configure zaptel so that it knows when to hangup. what hardware are you using ? what disconnect supervision does your telco operate ? what settings for hanguponpolarityswitch, busydetect and busycount do you have in zapata.conf (if any) ?

search the forum for “disconnect supervision” … there’s lots of people have been through it before.

3

i dont know what disconnect supervision our telco here in philippines using.
ok search for that

4

look in groups.yahoo.com/group/asterisk-ph/ for disconnect supervision on PH telco.

5

i found out one solution in my channel bank srvp i set it to REVPOL as reverse porlity.

however, still i want to know the disconnect supervision in pldt thus
i go to yahho.asterisk phils.

6

PLDT uses POLARITY REVERSAL disconnect supervision.

7

i have tried polarity reversal disconnect setting in my channel bank
but it act according to its term definition as "reverse"
as when i call and then after conversation if the endpoint hangup first iam not disconnected.
however, when i received calls and after conversation if the endpoint hangup first i am disconnected.

i dont know what is going on.

8

when u are the called party (B-party) and A-party hangs up first, your’re disconnected. it means polarity reversal is working.

in the previous case, the B-party hangs up first but the connection remains.

this is the main reason: by PLDT standards - only the A-party can teardown the connection. you can test it. let the B-party hangup momentarily hangup the phone. surprise! he’s still there!

in other words, only the B-party receives polarity reversal when A-party hangs up. A-party never receives polarity reversal disconnect signal.

yes, this is kinda weird protocal becuz any party should be able to teardown the connection. this is no problem if humans initiate the call. but it’s a big one if Asterisk applications initiates the call.

9

k thanks for additional info but i still believe someday it can be done.
as of now there are many things to work such as connecting to database setted in windows (it sound more complicated) but as long as there is asterisk
courage will remain no matter how hard it is.

uhh Linux and Asterisk why did you exist? the more i know you the more you get harder as you get bigger and bigger. uh hu hu.
but i promised, though im at your two steps behind i’ll grow with you and i’l get big with you. someday you’ll follow me no matter how hard you are!!

…just relaxing my friends:)
happy weekend