asterisk users - May 2021 - HELP! AGI AUTOHANGUP does not seem to hangup the channel.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup

"Cause the channel to automatically hangup at time seconds in the
future"

SET AUTOHANGUP TIME

Looks great. Except... it doesn't. It just causes AGI to send
"HANGUP"
and any audio to stop playing.
It does NOT hangup the channel, or even send any SIP event. The line
just goes silent.

It's been an entire afternoon of profuse googling; I have tried adding
and removing hangup handlers, I have even tried setting the
AGIEXITONHANGUP flag to "1" as per
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

But this is all that happens (5 seconds is ridiculous, it's just to
test). This is with "pjsip set logger on"

AGI Rx << SET AUTOHANGUP 5
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE AGIEXITONHANGUP "1"
AGI Tx >> 200 result=1
AGI Tx >> HANGUP   <<
AGI Rx << HANGUP
AGI Tx >> 511 Command Not Permitted on a dead channel or intercept routine

basically, the next log event is whenever the next REGISTER would
normally happen.

And of course, if I try and respond to the AGI again, it tells me the
channel is dead. So WHY is the "hangup" event not sent to the phone?

I cannot use other methods like setting the absolute channel timeout
variable, or using a local "dial" with a timeout message because:

1: the system tests whether the next file will push the listener over
the 1 hour limit for included calls in UK packages, so the autohangup
value has to be dynamic
2: It has to be over-ridable so that the listener can continue past the hour
3: The message that plays out is dynamically generated

Everything about it works fine... except the fact that it doesn't
actually hangup.
Which leaves the possibility that an old person might fall asleep and
end up with a large bill, which we don't want!

I have also tried configuring pjsip wizard like this.

endpoint/rtp_timeout=5

And I see this shortly after the "hangup" command has been sent, so
that part is working:

[May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150
rtp_check_timeout: Disconnecting channel
'PJSIP/fromvoipfone-206-0000000b' for lack of audio RTP activity in 5
seconds

But, again, it doesn't disconnect. The line stays open. And yes, my
fallthrough after agi is

same => n, Hangup()

Also, apparently I now have a load of channels, which won't even hangup with

channel request hangup all

Requested Hangup on channel 'PJSIP/fromvoipfone-206-0000000b'
Requested Hangup on channel 'PJSIP/fromvoipfone-206-0000000a'
Requested Hangup on channel 'PJSIP/fromvoipfone-206-00000009'

...and wait.. and then...

Channel              Location             State   Application(Data)
PJSIP/fromvoipfone-2 s at test:2             Up      AGI(agi://localhost:3456)
PJSIP/fromvoipfone-2 s at test:2             Up      AGI(agi://localhost:3456)
PJSIP/fromvoipfone-2 s at test:2             Up      AGI(agi://localhost:3456)
3 active channels
3 active calls

So they just won't die.

Asterisk 18.4.0 - worth filing a bug?

On Wed, 26 May 2021 at 17:22, Jonathan H <lardconcepts at gmail.com>
wrote:
>
>
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup
>
> "Cause the channel to automatically hangup at time seconds in the
future"
>
> SET AUTOHANGUP TIME
>
> Looks great. Except... it doesn't. It just causes AGI to send
"HANGUP"
> and any audio to stop playing.
> It does NOT hangup the channel, or even send any SIP event. The line
> just goes silent.
>
> It's been an entire afternoon of profuse googling; I have tried adding
> and removing hangup handlers, I have even tried setting the
> AGIEXITONHANGUP flag to "1" as per
>
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
>
> But this is all that happens (5 seconds is ridiculous, it's just to
> test). This is with "pjsip set logger on"
>
> AGI Rx << SET AUTOHANGUP 5
> AGI Tx >> 200 result=0
> AGI Rx << SET VARIABLE AGIEXITONHANGUP "1"
> AGI Tx >> 200 result=1
> AGI Tx >> HANGUP   <<
> AGI Rx << HANGUP
> AGI Tx >> 511 Command Not Permitted on a dead channel or intercept
routine
>
> basically, the next log event is whenever the next REGISTER would
> normally happen.
>
> And of course, if I try and respond to the AGI again, it tells me the
> channel is dead. So WHY is the "hangup" event not sent to the
phone?
>
> I cannot use other methods like setting the absolute channel timeout
> variable, or using a local "dial" with a timeout message because:
>
> 1: the system tests whether the next file will push the listener over
> the 1 hour limit for included calls in UK packages, so the autohangup
> value has to be dynamic
> 2: It has to be over-ridable so that the listener can continue past the
hour
> 3: The message that plays out is dynamically generated
>
> Everything about it works fine... except the fact that it doesn't
> actually hangup.
> Which leaves the possibility that an old person might fall asleep and
> end up with a large bill, which we don't want!

On Wed, 26 May 2021, Jonathan H wrote:
> It just causes AGI to send "HANGUP" and any audio to stop
playing. It
> does NOT hangup the channel, or even send any SIP event. The line just 
> goes silent.
I wouldn't expect the AGI() application to send a SIP event. The AGI()
application does not care what technology you use.

Receiving 'HANGUP' as text from Asterisk appears to be a FastAGI thing
which kind of makes sense -- if your FastAGI server is not localhost,
how could Asterisk send it a signal?

Are you supposed to close your TCP connection and exit your AGI when you 
receive the HANGUP text?

When I set autohangup in a 'normal' AGI, it looks like this:

<SIP/poly-77a1-000002a2>AGI Tx >> agi_request: null-agi.php
<SIP/poly-77a1-000002a2>AGI Tx >> agi_channel:
SIP/poly-77a1-000002a2
<SIP/poly-77a1-000002a2>AGI Tx >> agi_language: en
<SIP/poly-77a1-000002a2>AGI Tx >> agi_type: SIP
<SIP/poly-77a1-000002a2>AGI Tx >> agi_uniqueid: 1622093977.1168
<SIP/poly-77a1-000002a2>AGI Tx >> agi_version: 13.14.1~dfsg-2+deb9u4
<SIP/poly-77a1-000002a2>AGI Tx >> agi_callerid: 5555555555
<SIP/poly-77a1-000002a2>AGI Tx >> agi_calleridname: Steve Edwards
<SIP/poly-77a1-000002a2>AGI Tx >> agi_callingpres: 0
<SIP/poly-77a1-000002a2>AGI Tx >> agi_callingani2: 0
<SIP/poly-77a1-000002a2>AGI Tx >> agi_callington: 0
<SIP/poly-77a1-000002a2>AGI Tx >> agi_callingtns: 0
<SIP/poly-77a1-000002a2>AGI Tx >> agi_dnid: *
<SIP/poly-77a1-000002a2>AGI Tx >> agi_rdnis: unknown
<SIP/poly-77a1-000002a2>AGI Tx >> agi_context: newline
<SIP/poly-77a1-000002a2>AGI Tx >> agi_extension: *
<SIP/poly-77a1-000002a2>AGI Tx >> agi_priority: 6
<SIP/poly-77a1-000002a2>AGI Tx >> agi_enhanced: 0.0
<SIP/poly-77a1-000002a2>AGI Tx >> agi_accountcode: 
<SIP/poly-77a1-000002a2>AGI Tx >> agi_threadid: 1945654064
<SIP/poly-77a1-000002a2>AGI Tx >> 
<SIP/poly-77a1-000002a2>AGI Rx << set autohangup 5
<SIP/poly-77a1-000002a2>AGI Tx >> 200 result=0
        > 0x73c3dba0 -- Strict RTP learning complete - Locking on source
address 192.168.0.139:2254
(and then after 5 seconds)
     -- <SIP/poly-77a1-000002a2>AGI Script null-agi.php completed,
returning 4

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281

On Wed, 26 May 2021, Jonathan H wrote:
> AGI Rx << SET AUTOHANGUP 5
> AGI Tx >> 200 result=0
> AGI Tx >> HANGUP   <<
This does raise a question in my mind...

The AGI protocol is: your AGI sends a request (the Rx line) and receives 
a response (the Tx line). 1 line out, 1 line in.

If the 'HANGUP' text can arrive asynchronously, how are you supposed to 
know it has arrived? Poll (or select) on the file pointer?
> I cannot use other methods like setting the absolute channel timeout 
> variable
I don't understand why you can't use the absolute channel timeout. 
Wherever you 'set autohangup x' just set
'TIMEOUT(absolute)=${EPOCH}+x.'

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281