Unable to establish call

Can anyone help me to establish the IVR call? Following are the details in debug log.

[root@appserver1 ~]# asterisk -r
Asterisk 13.23.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.23.1 currently running on appserver1 (pid = 32531)
appserver1*CLI> sip set debug on 
SIP Debugging re-enabled
appserver1*CLI> 
appserver1*CLI> 

<--- SIP read from UDP:192.168.151.39:5060 --->
INVITE sip:16279@192.168.151.45:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.151.39:5060;rport;branch=z9hG4bK-524287-1---65eb95a344c327f
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Contact: <sip:1958416690@192.168.151.39:5060>
Route: <sip:192.168.151.45:5060;lr>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, INFO
Content-Type: application/sdp
User-Agent: TBS Solution V1.0 (Nov 11 2019_14:23:34)
Content-Length: 127

v=0
o=tbs 1 2 IN IP4 192.168.151.42
s=tbs1
c=IN IP4 192.168.151.42
t=3 4
m=audio 33177 RTP/AVP 8
a=ptime:20
a=sendrecv


<------------->
--- (13 headers 9 lines) ---
Sending to 192.168.151.39:5060 (NAT)
Sending to 192.168.151.39:5060 (NAT)
Using INVITE request as basis request - 728a7a4b52402bc1
No matching peer for '1958416690' from '192.168.151.39:5060'
Found RTP audio format 8
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.151.42:33177
Looking for 16279 in public (domain 192.168.151.45)
sip_route_dump: route/path hop: <sip:1958416690@192.168.151.39:5060>

<--- Transmitting (NAT) to 192.168.151.39:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:16279@192.168.151.45:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.151.39:5060 --->
CANCEL sip:16279@192.168.151.45:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.151.39:5060;rport;branch=z9hG4bK-524287-1---65eb95a344c327f
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>
Call-ID: 728a7a4b52402bc1
CSeq: 1 CANCEL
Route: <sip:192.168.151.45:5060;lr>
Max-Forwards: 70
User-Agent: TBS Solution V1.0 (Nov 11 2019_14:23:34)/r/nReason: SIP ;cause=503 ;text="503 Service Unavailable"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.151.39:5060 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.151.39:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.151.39:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 CANCEL
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Retransmitting #1 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #5 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #6 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #7 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #8 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #9 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #10 (NAT) to 192.168.151.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.151.39:5060;branch=z9hG4bK-524287-1---65eb95a344c327f;received=192.168.151.39;rport=5060
From: <sip:1958416690@192.168.151.39:5060>;tag=1e0311706e6734c7
To: <sip:16279@192.168.151.45:5060>;tag=as77a92b84
Call-ID: 728a7a4b52402bc1
CSeq: 1 INVITE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 19 16:33:59] WARNING[32584]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission 728a7a4b52402bc1 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog '728a7a4b52402bc1' Method: CANCEL
appserver1*CLI> 
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@appserver1 ~]# 

The caller cancelled, so it was they that objected to the call. The caller has a s protocol error in that they failed to ACK the OK from the call. My guess is that there is no network path back from Asterisk to the caller and they are not seeing any of Asterisk’s responses.

The caller appears to be being treated as unknown by Asterisk, but with allowguest set on.

The verbosity level is too low for the logs to show how the call was handled, Use at least 5 when submitting diagnostic logs.

You appear to have screen scraped the logs, rather than used the log files, so no timing information is available.