Asterisk call disconnection after a time of 9s

Asterisk Asterisk SIP
1

Hi Team,

I am facing an issue in Asterisk, in which , when I am calling from my phone to asterisk,to login an agent using AgentLogin() application, via service provider.But the issue I am facing is disconnection of the call after 9s, after agent is logged in.
Please can anyone help me with the issue.

Please refer to the Sip output below-
<— Received SIP request (1310 bytes) from UDP:54.172.60.2:5060 —>
INVITE sip:+16508300379@35.233.190.15 SIP/2.0
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15
CSeq: 31282 INVITE
Max-Forwards: 62
P-Asserted-Identity: sip:+16617480240@4.55.11.163:5060
Privacy: id
Diversion: sip:+16508300379@public-vip.us1.twilio.com;reason=unconditional
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.2:5060;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Contact: “Anonymous” sip:Anonymous@172.18.0.207:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC5b8a51bee9fbfe6aa5da40bad31c4d13
Content-Type: application/sdp
X-Twilio-CallSid: CA6c95e6abee6261109ff5ab29645a2d81
Content-Length: 238

v=0
o=root 336781834 336781834 IN IP4 34.203.250.232
s=Twilio Media Gateway
c=IN IP4 34.203.250.232
t=0 0
m=audio 13204 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

== Setting global variable ‘SIPDOMAIN’ to ‘35.233.190.15’
<— Transmitting SIP response (662 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Content-Length: 0

-- Executing [+16508300379@from_outside:1] Verbose("PJSIP/Xo_main-0000007e", "+16508300379") in new stack

+16508300379
– Executing [+16508300379@from_outside:2] Macro(“PJSIP/Xo_main-0000007e”, “start-ivr”) in new stack
– Executing [s@macro-start-ivr:1] Answer(“PJSIP/Xo_main-0000007e”, “”) in new stack
> 0x7f2ae010a880 – Strict RTP learning after remote address set to: 34.203.250.232:13204
<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Executing [s@macro-start-ivr:2] Wait("PJSIP/Xo_main-0000007e", "1") in new stack

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Executing [s@macro-start-ivr:3] Goto("PJSIP/Xo_main-0000007e", "ivr,s,1") in new stack
-- Goto (ivr,s,1)

== Channel ‘PJSIP/Xo_main-0000007e’ jumping out of macro ‘start-ivr’
– Executing [s@ivr:1] BackGround(“PJSIP/Xo_main-0000007e”, “thank-you-for-calling”) in new stack
– <PJSIP/Xo_main-0000007e> Playing ‘thank-you-for-calling.ulaw’ (language ‘en’)
> 0x7f2ae010a880 – Strict RTP switching to RTP target address 34.203.250.232:13204 as source
– Executing [s@ivr:2] Verbose(“PJSIP/Xo_main-0000007e”, “Anonymous”) in new stack
Anonymous
– Executing [s@ivr:3] WaitExten(“PJSIP/Xo_main-0000007e”, “.65”) in new stack
– Timeout on PJSIP/Xo_main-0000007e, continuing…
– Executing [s@ivr:4] BackGround(“PJSIP/Xo_main-0000007e”, “press-1”) in new stack
– <PJSIP/Xo_main-0000007e> Playing ‘press-1.ulaw’ (language ‘en’)
<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Executing [s@ivr:5] WaitExten("PJSIP/Xo_main-0000007e", "15") in new stack
   > 0x7f2ae010a880 -- Strict RTP learning complete - Locking on source address 34.203.250.232:13204

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Executing [1@ivr:1] Set("PJSIP/Xo_main-0000007e", "USER_TRIES=0") in new stack
-- Executing [1@ivr:2] Set("PJSIP/Xo_main-0000007e", "USER_TRIES=1") in new stack
-- Executing [1@ivr:3] GotoIf("PJSIP/Xo_main-0000007e", "0?t,1") in new stack
-- Executing [1@ivr:4] Read("PJSIP/Xo_main-0000007e", "CASAGENT,"agent-user"") in new stack
-- <PJSIP/Xo_main-0000007e> Playing 'agent-user.gsm' (language 'en')

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- User entered '22444'
-- Executing [1@ivr:5] GotoIf("PJSIP/Xo_main-0000007e", "0?user-retry") in new stack
-- Executing [1@ivr:6] GotoIf("PJSIP/Xo_main-0000007e", "0?user-retry") in new stack
-- Executing [1@ivr:7] Authenticate("PJSIP/Xo_main-0000007e", "22444") in new stack
-- <PJSIP/Xo_main-0000007e> Playing 'agent-pass.gsm' (language 'en')

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- <PJSIP/Xo_main-0000007e> Playing 'auth-thankyou.gsm' (language 'en')
-- Executing [1@ivr:8] AgentLogin("PJSIP/Xo_main-0000007e", "22444") in new stack
-- <PJSIP/Xo_main-0000007e> Playing 'agent-loginok.gsm' (language 'en')

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

== Agent ‘22444’ logged in (format ulaw/ulaw)
– Channel PJSIP/Xo_main-0000007e joined ‘holding_bridge’ agent_hold-bridge
[2019-05-21 11:57:40.056] WARNING[22135][C-00000140]: res_musiconhold.c:918 _get_mohbyname: Music on Hold class ‘default’ not found in memory. Verify your configuration.
[2019-05-21 11:57:40.056] WARNING[22135][C-00000140]: res_musiconhold.c:918 _get_mohbyname: Music on Hold class ‘default’ not found in memory. Verify your configuration.
[2019-05-21 11:57:40.056] WARNING[22135][C-00000140]: bridge_holding.c:209 participant_entertainment_start: Failed to start moh, starting silence generator instead
<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (1149 bytes) to UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.2:5060;rport=5060;received=54.172.60.2;branch=z9hG4bKcc17.62fcaf77.0
Via: SIP/2.0/UDP 172.18.0.207:5060;rport=5060;received=172.18.0.207;branch=z9hG4bK1a803f4d-ea1c-4ab6-8996-799b43665e9a_6772d868_363-6417517713662558237
Record-Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
To: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
CSeq: 31282 INVITE
Server: Asterisk PBX certified/13.21-cert3
Contact: sip:10.0.1.2:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 336781834 336781836 IN IP4 10.0.1.2
s=Asterisk
c=IN IP4 10.0.1.2
t=0 0
m=audio 16302 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (595 bytes) to UDP:54.172.60.2:5060 —>
BYE sip:Anonymous@172.18.0.207:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.2:5060;rport;branch=z9hG4bKPj2022c400-a2c6-4033-858c-8d3d5959a677
From: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
To: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
CSeq: 15705 BYE
Route: sip:54.172.60.2:5060;lr;ftag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Max-Forwards: 70
User-Agent: Asterisk PBX certified/13.21-cert3
Content-Length: 0

<— Received SIP response (489 bytes) from UDP:54.172.60.2:5060 —>
SIP/2.0 200 OK
CSeq: 15705 BYE
Call-ID: 93c7c57fda9cec4871404b66269525fb@0.0.0.0
From: sip:+16508300379@35.233.190.15;tag=b42371ae-2a3a-4e7d-a50a-ae322bc00be7
To: “Anonymous” sip:Anonymous@aster.pstn.twilio.com;tag=07463216_6772d868_1a803f4d-ea1c-4ab6-8996-799b43665e9a
Via: SIP/2.0/UDP 10.0.1.2:5060;received=35.233.190.15;rport=5060;branch=z9hG4bKPj2022c400-a2c6-4033-858c-8d3d5959a677
Server: Twilio
X-Twilio-CallSid: CA6c95e6abee6261109ff5ab29645a2d81
Content-Length: 0

-- Channel PJSIP/Xo_main-0000007e left 'holding_bridge' agent_hold-bridge <c93dbf52-4c64-4ca3-8af7-aa8d966e65a9>

== Agent ‘22444’ logged out. Logged in for 9 seconds.
== Spawn extension (ivr, 1, 8) exited non-zero on ‘PJSIP/Xo_main-0000007e’
– Executing [h@ivr:1] GotoIf(“PJSIP/Xo_main-0000007e”, “0 & 1?casdevs”) in new stack
– Executing [h@ivr:2] Goto(“PJSIP/Xo_main-0000007e”, “agent-logoff,22444,1”) in new stack
– Goto (agent-logoff,22444,1)
– Executing [22444@agent-logoff:1] Verbose(“PJSIP/Xo_main-0000007e”, “Logging off Agnet No. 22444”) in new stack
Logging off Agnet No. 22444
– Executing [22444@agent-logoff:2] System(“PJSIP/Xo_main-0000007e”, “/usr/sbin/asterisk -rx “ConfBridge kick 22444 all””) in new stack
– Remote UNIX connection
– Remote UNIX connection disconnected
– Executing [22444@agent-logoff:3] System(“PJSIP/Xo_main-0000007e”, "/usr/sbin/asterisk -rx “agent logoff 22444"”) in new stack
– Remote UNIX connection
– Remote UNIX connection disconnected
– Executing [22444@agent-logoff:4] Hangup(“PJSIP/Xo_main-0000007e”, “”) in new stack
== Spawn extension (agent-logoff, 22444, 4) exited non-zero on ‘PJSIP/Xo_main-0000007e’

2

Asterisk is not receiving an ACK message to its 200 OK, presumably because you are telling the remote side to send the ACK to a private IP address and port:

Contact: sip:10.0.1.2:5060

You would need to alter the PJSIP configuration and set the local_net, external_signaling_address, and external_media_address options on the transport.

3

Clearly as @jcolp mentioned Asterisk is terminating the call due to the missing ACK reply

10.0.1.2 is not a reachable address

4

Hi Team,

I have followed the instructions suggested by @jcolp and @ambiorixg12, and everything seems to be working fine now.
Thank You.

5

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