Call Disconnect after 100 sec

Hi ,
I have successfully connected call . I am using G 729 but it is disconnecting after 100 sec .Please check below log .I am receiving BYE packet . I am not understand what is the issue with this .

Please help

:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj9b146480-1e67-485d-b8b5-6501a8f02c3f
From: sip:1033@172.31.4.130;tag=ce63a5f6-ebf9-4879-ae6d-c86055c8f04b
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 21630929-b501-449e-b8c6-3be4944f6064
CSeq: 31598 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj415547b1-ba4e-4e9e-b210-64dd109b1733
From: sip:1033@172.31.4.130;tag=a23dee82-2e12-4d1c-82e6-5f65bc331d37
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 92fd5f22-2f85-4d35-9d46-14987b7b9aec
CSeq: 22633 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPje6724036-9d94-468c-8696-f31dac4fbb00
From: sip:1033@172.31.4.130;tag=a03d9b4f-6eff-4ecc-ac86-63ec634398b9
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 86dd0aac-5d62-4497-be0c-b14f439fdc44
CSeq: 25835 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (422 bytes) to UDP:45.127.194.91:46224 —>
OPTIONS sip:1050@45.127.194.91:46224 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjed9128ec-3524-401d-8bd4-15b4d2c7be65
From: sip:1050@172.31.4.130;tag=c9e93ca5-1f6c-4e76-96db-2fc7f9e81928
To: sip:1050@45.127.194.91
Contact: sip:1050@16.200.90.76:8080
Call-ID: e4cfb43b-4081-4aab-a990-52c1ab4cade2
CSeq: 3423 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:56413 —>
OPTIONS sip:1033@157.38.136.156:56413 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj25b33b17-4b6b-4079-b8d7-f24ac39ee2df
From: sip:1033@172.31.4.130;tag=2dd10230-76fe-4678-b423-18e6cc67a1f7
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 98eb86fc-784e-4089-ae8f-4c5282441a80
CSeq: 35797 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Received SIP response (294 bytes) from UDP:157.38.136.156:56413 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj25b33b17-4b6b-4079-b8d7-f24ac39ee2df
From: sip:1033@172.31.4.130;tag=2dd10230-76fe-4678-b423-18e6cc67a1f7
To: sip:1033@157.38.136.156;tag=sSxa~
Call-ID: 98eb86fc-784e-4089-ae8f-4c5282441a80
CSeq: 35797 OPTIONS

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj2ee88f0d-1fc0-4408-aee2-908d3bff2bbe
From: sip:1033@172.31.4.130;tag=d1a3e8d3-5c1a-451a-8880-5757052a99f1
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: dcc25661-0ea0-4489-9bae-814647ecd7ad
CSeq: 48840 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj9b146480-1e67-485d-b8b5-6501a8f02c3f
From: sip:1033@172.31.4.130;tag=ce63a5f6-ebf9-4879-ae6d-c86055c8f04b
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 21630929-b501-449e-b8c6-3be4944f6064
CSeq: 31598 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj415547b1-ba4e-4e9e-b210-64dd109b1733
From: sip:1033@172.31.4.130;tag=a23dee82-2e12-4d1c-82e6-5f65bc331d37
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 92fd5f22-2f85-4d35-9d46-14987b7b9aec
CSeq: 22633 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPje6724036-9d94-468c-8696-f31dac4fbb00
From: sip:1033@172.31.4.130;tag=a03d9b4f-6eff-4ecc-ac86-63ec634398b9
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 86dd0aac-5d62-4497-be0c-b14f439fdc44
CSeq: 25835 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (422 bytes) to UDP:45.127.194.91:46224 —>
OPTIONS sip:1050@45.127.194.91:46224 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjed9128ec-3524-401d-8bd4-15b4d2c7be65
From: sip:1050@172.31.4.130;tag=c9e93ca5-1f6c-4e76-96db-2fc7f9e81928
To: sip:1050@45.127.194.91
Contact: sip:1050@16.200.90.76:8080
Call-ID: e4cfb43b-4081-4aab-a990-52c1ab4cade2
CSeq: 3423 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj2ee88f0d-1fc0-4408-aee2-908d3bff2bbe
From: sip:1033@172.31.4.130;tag=d1a3e8d3-5c1a-451a-8880-5757052a99f1
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: dcc25661-0ea0-4489-9bae-814647ecd7ad
CSeq: 48840 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (424 bytes) to UDP:157.38.136.156:56413 —>
OPTIONS sip:1033@157.38.136.156:56413 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjc8be4b31-34c7-4cd5-b610-396fc9bcc6d5
From: sip:1033@172.31.4.130;tag=da886c3d-fb6d-4d3f-ac5d-2ca0df1d0b8e
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 06d2b222-ffee-4b79-a741-6ac3348760b6
CSeq: 7171 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Received SIP response (293 bytes) from UDP:157.38.136.156:56413 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjc8be4b31-34c7-4cd5-b610-396fc9bcc6d5
From: sip:1033@172.31.4.130;tag=da886c3d-fb6d-4d3f-ac5d-2ca0df1d0b8e
To: sip:1033@157.38.136.156;tag=G0XQf
Call-ID: 06d2b222-ffee-4b79-a741-6ac3348760b6
CSeq: 7171 OPTIONS

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj415547b1-ba4e-4e9e-b210-64dd109b1733
From: sip:1033@172.31.4.130;tag=a23dee82-2e12-4d1c-82e6-5f65bc331d37
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 92fd5f22-2f85-4d35-9d46-14987b7b9aec
CSeq: 22633 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

-- Channel PJSIP/1050-00000049 left 'native_rtp' basic-bridge <cf5afe1a-2457-400a-b04e-3742c85da81f>
-- Channel PJSIP/1049-0000004a left 'native_rtp' basic-bridge <cf5afe1a-2457-400a-b04e-3742c85da81f>

<— Transmitting SIP request (412 bytes) to UDP:45.127.194.91:54570 —>
BYE sip:1049@45.127.194.91:54570 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjf42f2dfc-25d4-42e4-ae22-a09b003244a0
From: sip:1050@172.31.4.130;tag=2e616629-d81f-457c-8191-2cc8fbd086da
To: sip:1049@45.127.194.91;tag=~2JZlKz
Call-ID: 37c7359f-7789-4e98-bd1f-e433f6b3e749
CSeq: 23637 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

-- <PJSIP/1050-00000049>AGI Script agi://127.0.0.1:4575/fcdial.agi completed, returning 0
-- Auto fallthrough, channel 'PJSIP/1050-00000049' status is 'ANSWER'

<— Transmitting SIP request (387 bytes) to UDP:45.127.194.91:36251 —>
BYE sip:1050@45.127.194.91:36251 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjb90cdbf9-0df1-4fd9-a000-c470a5626606
From: sip:1049@16.200.90.76;tag=6cd0e42f-5a4a-4428-8276-f6464356130c
To: sip:1050@16.200.90.76;tag=pB3~myFRq
Call-ID: g61ArQ0730
CSeq: 15961 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPje6724036-9d94-468c-8696-f31dac4fbb00
From: sip:1033@172.31.4.130;tag=a03d9b4f-6eff-4ecc-ac86-63ec634398b9
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: 86dd0aac-5d62-4497-be0c-b14f439fdc44
CSeq: 25835 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Received SIP response (381 bytes) from UDP:45.127.194.91:36251 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjb90cdbf9-0df1-4fd9-a000-c470a5626606
From: sip:1049@16.200.90.76;tag=6cd0e42f-5a4a-4428-8276-f6464356130c
To: sip:1050@16.200.90.76;tag=pB3~myFRq
Call-ID: g61ArQ0730
CSeq: 15961 BYE
User-Agent: LinphoneAndroid/4.2.3 (Redmi 6) LinphoneSDK/4.3.3 (tags/4.3.3^0)
Supported: replaces, outbound, gruu

<— Transmitting SIP request (412 bytes) to UDP:45.127.194.91:54570 —>
BYE sip:1049@45.127.194.91:54570 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjf42f2dfc-25d4-42e4-ae22-a09b003244a0
From: sip:1050@172.31.4.130;tag=2e616629-d81f-457c-8191-2cc8fbd086da
To: sip:1049@45.127.194.91;tag=~2JZlKz
Call-ID: 37c7359f-7789-4e98-bd1f-e433f6b3e749
CSeq: 23637 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (425 bytes) to UDP:157.38.136.156:52916 —>
OPTIONS sip:1033@157.38.136.156:52916 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPj2ee88f0d-1fc0-4408-aee2-908d3bff2bbe
From: sip:1033@172.31.4.130;tag=d1a3e8d3-5c1a-451a-8880-5757052a99f1
To: sip:1033@157.38.136.156
Contact: sip:1033@16.200.90.76:8080
Call-ID: dcc25661-0ea0-4489-9bae-814647ecd7ad
CSeq: 48840 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

<— Transmitting SIP request (412 bytes) to UDP:45.127.194.91:54570 —>
BYE sip:1049@45.127.194.91:54570 SIP/2.0
Via: SIP/2.0/UDP 16.200.90.76:8080;rport;branch=z9hG4bKPjf42f2dfc-25d4-42e4-ae22-a09b003244a0
From: sip:1050@172.31.4.130;tag=2e616629-d81f-457c-8191-2cc8fbd086da
To: sip:1049@45.127.194.91;tag=~2JZlKz
Call-ID: 37c7359f-7789-4e98-bd1f-e433f6b3e749
CSeq: 23637 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0

Asterisk is disconnecting the call, is this call sent to an AGI script , what it is the call flow ?

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