External calls fall after 60 seconds

In my system, external calls drop after 60 seconds if they are not answered. Below is the excerpt from extensions.conf:

exten => _002XXXXXXXX,1,Macro(prefixo,${CALLERID(number)})
exten => _002XXXXXXXX,2,Dial(${TRUNKTB01}/${EXTEN:2},tT)
exten => _002XXXXXXXX,3,Dial(${TRUNKTB02}/${EXTEN:2})
exten => _002XXXXXXXX,4,Dial(${TRUNKMIRIADES}/${EXTEN})
exten => _002XXXXXXXX,5,Dial(${TRUNKTEL}/${EXTEN})
exten => _002XXXXXXXX,6,Dial(${TRUNKPSTN}/${EXTEN:2})
exten => _002XXXXXXXX,7,Hangup()

The output of the trace:

Scheduling destruction of call ‘5cb86ec1-4e5a228e-83d324b3@10.228.0.179’ in 15000 ms
Destroying call ‘e1b8f847-889c132-9336c49@10.192.250.15’
govoip3*CLI>
<-- SIP read from 10.192.250.1:5060:
CANCEL sip:00212421344@10.192.231.252;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.192.250.1;branch=z9hG4bK809338bb6D03F957
From: “24061” sip:24061@10.192.231.252;tag=4F4EE817-400EFE2B
To: sip:00212421344@10.192.231.252;user=phone
CSeq: 2 CANCEL
Call-ID: c66f3b7-3247330b-bbbf1947@10.192.250.1
Contact: sip:24061@10.192.250.1;transport=udp
User-Agent: PolycomVVX-VVX_500-UA/5.1.2.1801
Proxy-Authorization: Digest username=“24061”, realm=“Jar209Reftel.refertelecom.pt”, nonce=“7d27f66e”, uri="sip:00212421344@10.192.231.252;user=phone;transport=udp", response=“1c9c15c5ed95abc4bd7e3b08a7eae7ff”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

I do not understand why the asterisk cancels the call after 60 seconds every time.

Even with this configuration:

exten => _002XXXXXXXX,2,Dial(${TRUNKTB01}/${EXTEN:2},120,tT)…

SIP.conf:

[general]
realm=Jar209Reftel.refertelecom.pt
context=default
allowoverlap=yes
srvlookup=yes
defaultexpiry=90
qualify=yes
videosupport=yes
notifyringing = yes
notifyhold = yes
callcounter = yes
counteronpeer = yes
alwaysauthreject = yes
;allowguest=no
disallow=all
allow = alaw
allow = ulaw
allow = g729
allow = gsm
allow=h261
allow=h263
allow=h263p
rtptimeout=300
nat=no
relaxdtmf=yes
rfc2833compensate=yes
udpbindaddr=0.0.0.0
bindport=5060
bindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
tos_sip=ef
tos_audio=ef
tos_video=af41
tos_text=af41
cos_sip=5;
cos_audio=5
cos_video=4
cos_text=3
sendrpid=yes
trustrpid=yes

Can someone help me to understand why asterisk cancels calls after 60 seconds?

Asterisk isn’t canceling the call, the Polycom is. I believe there’s a built in automatic termination if the call is unanswered.

Thanks for your answer.
I’ve several different types of polycom phones (VVX500, IP430, IP331…) working. How can I change this timeout please?

I have no experience with Polycoms so I can’t comment. Perhaps someone else knows.

Ok. Thanks. Maybe someone knows…

;exten => _99909*.,1,Progress() ; Indicates that the call is in progress
;exten => _99909*.,n,Ringing() ; Sends a ringback tone to the caller
;exten => _99909*.,n,AGI(agi-VDAD_ALL_inbound.agi) ; Execute your AGI script
;exten => _99909*.,n,Hangup() ; Hang up the call after AGI execution

Now the call is ringing for 60 seconds and then dropped. Therefore, the call should remain in the queue until it is answered by an agent.

how to do it