Call Hangup Detect

yuyoungtaik · August 6, 2024, 5:51am

I use asterisk Version : 18.23.1

[hdlr_Hangup]
exten => s,1,NoOp(hdlr_Hangup context is load…)
same => n, hangup()

[ivr-parkgolf]
include => hdlr_Hangup

exten => s,1,Wait(1)
same => n, Answer()
same => n, Set(CHANNEL(hangup_handler_push)=hdlr_Hangup,s,1)
same=> n, playback(h1)
…

When I make a call with a softphone and hang up, a hangup event occurs immediately.
When I received an IP phone number from the telecommunications company and connected,
After the call, even if I hang up, the hangup event does not occur and the flow continues.
I need help figuring out where to check.

I’ve just started using Asterisk, so I don’t know where to set it up.
I searched and found that it might be correct, but
it says to set chan_dahdi.conf’s
busydetect=yes, but even after setting it,
there is no response.
Since I only received the phone number from the telecommunications company’s IP-PBX (it is not physically installed as hardware), is it correct to edit the dahdi?

p.s: I tested it and it seems like I get user disconnected 30 seconds after I hang up the phone.

jcolp · August 6, 2024, 8:27am

DAHDI is only for physical telephony cards. If using SIP to connect upstream, then you would receive a BYE upon hangup. If you aren’t receiving that for a period of time, then the problem is upstream.

yuyoungtaik · August 6, 2024, 8:38am

Hang up the phone and wait 30 seconds
-User disconnected

It seems like my calls are disconnected after 30 seconds of hanging up. How can I make this time go faster?

-- User disconnected
-- Executing [h@ivr-parkgolf:1] NoOp("PJSIP/sjtelcom-00000007", "h extension .......") in new stack
-- PJSIP/sjtelcom-00000007 Internal Gosub(hdlr_Hangup,s,1) start

jcolp · August 6, 2024, 8:41am

Provide the output of “pjsip set logger on” from start to finish for a call.

yuyoungtaik · August 6, 2024, 9:05am

<— Received SIP request (1069 bytes) from UDP:203.203.100.40:5060 —>
INVITE sip:07012341234@170.100.100.100:5060;transport=UDP;user=phone SIP/2.0
f: sip:010234523345@203.203.100.40:5060;user=phone;tag=-65014-35b0cd3-692b5ebf-35b0cd3
t: sip:07012341234@170.100.100.100:5060;user=phone
i: 9d833008486f0cb13c435b0cd31baa18d638fa891e6ea82480-0007-8638
CSeq: 1 INVITE
User-agent: CS2000_NGSS/9.0
X-Nortel-Profile: DEFAULT
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
v: SIP/2.0/UDP 203.203.100.40:5060;branch=z9hG4bK-35b0cd3-1baa18d6-65f2c8a5-7fffa57cd660
P-Asserted-Identity: sip:010234523345@203.203.100.40;user=phone
Max-Forwards: 140
m: sip:203.203.100.40:5060;transport=UDP
k: 100rel,resource-priority
c: application/sdp
l: 360

v=0
o=genband 269012992 1722934812 IN IP4 203.240.134.38
s=-
c=IN IP4 203.240.134.38
t=0 0
m=audio 43040 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:30
a=sqn: 0
a=cdsc:1 image udptl t38

<— Transmitting SIP response (419 bytes) to UDP:203.203.100.40:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.203.100.40:5060;rport=5060;received=203.203.100.40;branch=z9hG4bK-35b0cd3-1baa18d6-65f2c8a5-7fffa57cd660
Call-ID: 9d833008486f0cb13c435b0cd31baa18d638fa891e6ea82480-0007-8638
From: sip:010234523345@203.203.100.40;user=phone;tag=-65014-35b0cd3-692b5ebf-35b0cd3
To: sip:07012341234@170.100.100.100;user=phone
CSeq: 1 INVITE
Server: Asterisk PBX 18.23.1
Content-Length: 0

-- Executing [07012341234@ivr-parkgolf:1] Goto("PJSIP/sjtelcom-00000009", "ivr-parkgolf,s,1") in new stack
-- Goto (ivr-parkgolf,s,1)
-- Executing [s@ivr-parkgolf:1] Wait("PJSIP/sjtelcom-00000009", "1") in new stack
-- Executing [s@ivr-parkgolf:2] Answer("PJSIP/sjtelcom-00000009", "") in new stack
   > 0x7f286c01ff10 -- Strict RTP learning after remote address set to: 203.240.134.38:43040

<— Transmitting SIP response (946 bytes) to UDP:203.203.100.40:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.203.100.40:5060;rport=5060;received=203.203.100.40;branch=z9hG4bK-35b0cd3-1baa18d6-65f2c8a5-7fffa57cd660
Call-ID: 9d833008486f0cb13c435b0cd31baa18d638fa891e6ea82480-0007-8638
From: sip:010234523345@203.203.100.40;user=phone;tag=-65014-35b0cd3-692b5ebf-35b0cd3
To: sip:07012341234@170.100.100.100;user=phone;tag=44de60d4-ef34-424c-aa07-c466facea4fd
CSeq: 1 INVITE
Server: Asterisk PBX 18.23.1
Contact: sip:172.30.1.22:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 258