Not able to hear audios in between PJSIP Extensions

I setup Asterisk18 and PBX 15 inside a docker container and My Exposed Ports are like this
docker ps
But I am able to connect to extensions but not able to hear audio Both Ways.
CONTAINER ID IMAGE COMMAND CREATED STATUS PORTS

                  NAMES

d85e25c8b852 v1.0:latest “/bin/bash” 2 hours ago Up 2 hours 5038/tcp, 0.0.0.0:80->80/tcp, 0.0.0.0:5061->5061/tcp, 0.0.0.0:8000->8000/tcp, 0.0.0.0:5060->5060/udp, 0.0.0.0:8088->8088/tcp, 5060/tcp, 0.0.0.0:15000-15005->15000-15005/udp, 15006-15010/udp voip_server

a=sendrecv

    -- Channel PJSIP/300-0000000e joined 'simple_bridge' basic-bridge <8b0e60c3-f83c-4ba7-af09-35d67decd8a4>
    -- Channel PJSIP/100-0000000d joined 'simple_bridge' basic-bridge <8b0e60c3-f83c-4ba7-af09-35d67decd8a4>
<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[2024-06-24 18:14:49] NOTICE[36018]: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel 'PJSIP/300-0000000e' for lack of audio RTP activity in 32 seconds
    -- Channel PJSIP/300-0000000e left 'simple_bridge' basic-bridge <8b0e60c3-f83c-4ba7-af09-35d67decd8a4>
    -- Channel PJSIP/100-0000000d left 'simple_bridge' basic-bridge <8b0e60c3-f83c-4ba7-af09-35d67decd8a4>
  == Spawn extension (macro-dial-one, s, 54) exited non-zero on 'PJSIP/100-0000000d' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'PJSIP/100-0000000d' in macro 'exten-vm'
  == Spawn extension (from-internal, 300, 3) exited non-zero on 'PJSIP/100-0000000d'
    -- Executing [h@from-internal:1] Macro("PJSIP/100-0000000d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/100-0000000d", "1?theend") in new stack
[2024-06-24 18:14:49] NOTICE[36018]: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel 'PJSIP/100-0000000d' for lack of audio RTP activity in 32 seconds
<--- Transmitting SIP request (443 bytes) to UDP:172.17.0.1:56685 --->
BYE sip:300@172.17.0.1:56685;ob SIP/2.0
Via: SIP/2.0/UDP 103.248.120.114:5060;rport;branch=z9hG4bKPj715258a9-e32f-4cc5-b243-2d0c086ca76a
From: "100" <sip:100@172.17.0.2>;tag=c821045f-ccf0-4a34-a5f4-166f2754aafb
To: <sip:300@172.17.0.1;ob>;tag=345d4b7f518d48b6948d456622c1596f
Call-ID: 175b80be-567e-4927-897a-42e8fae2eb37
CSeq: 13705 BYE
Reason: Q.850;cause=44
Max-Forwards: 70
User-Agent: FPBX-15.0.23(18.23.1)
Content-Length:  0


    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/100-0000000d", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("PJSIP/100-0000000d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/100-0000000d' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-0000000d'
<--- Received SIP response (367 bytes) from UDP:172.17.0.1:56685 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.248.120.114:5060;rport=5060;received=192.168.1.7;branch=z9hG4bKPj715258a9-e32f-4cc5-b243-2d0c086ca76a
Call-ID: 175b80be-567e-4927-897a-42e8fae2eb37
From: "100" <sip:100@172.17.0.2>;tag=c821045f-ccf0-4a34-a5f4-166f2754aafb
To: <sip:300@172.17.0.1;ob>;tag=345d4b7f518d48b6948d456622c1596f
CSeq: 13705 BYE
Content-Length:  0


<--- Transmitting SIP response (978 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPj2f24df912d9340be9abdbdb05d8da99b
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
To: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
CSeq: 1895 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:103.248.120.114:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928261457 3928261459 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP request (407 bytes) to UDP:172.17.0.1:45164 --->
BYE sip:100@172.17.0.1:45164;ob SIP/2.0
Via: SIP/2.0/UDP 103.248.120.114:5060;rport;branch=z9hG4bKPj6be32c5f-9f20-44ff-895c-4b52960e14be
From: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
To: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
Call-ID: 74cd7c47acfd4811a7774e9107a73938
CSeq: 9123 BYE
Max-Forwards: 70
User-Agent: FPBX-15.0.23(18.23.1)
Content-Length:  0


<--- Received SIP response (355 bytes) from UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.248.120.114:5060;rport=5060;received=192.168.1.7;branch=z9hG4bKPj6be32c5f-9f20-44ff-895c-4b52960e14be
Call-ID: 74cd7c47acfd4811a7774e9107a73938
From: <sip:300@192.168.1.7>;tag=8d1e22b4-4217-4f0e-b6b8-6a5bd86612ee
To: <sip:100@192.168.1.7>;tag=458b58022dad41dc82c2f8ad7ddb5918
CSeq: 9123 BYE
Content-Length:  0


<--- Transmitting SIP request (425 bytes) to UDP:172.17.0.1:56685 --->
OPTIONS sip:300@172.17.0.1:56685;ob SIP/2.0
Via: SIP/2.0/UDP 103.248.120.114:5060;rport;branch=z9hG4bKPj05dfacfb-a3f3-44e0-a215-4fcde87ed63d
From: <sip:300@172.17.0.2>;tag=a6e77797-b551-45ea-b043-6d201341e5e7
To: <sip:300@172.17.0.1;ob>
Contact: <sip:300@103.248.120.114:5060>
Call-ID: 90dd4851-f349-45c2-b168-14c8bac16f88
CSeq: 23107 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.23(18.23.1)
Content-Length:  0


<--- Received SIP response (793 bytes) from UDP:172.17.0.1:56685 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.248.120.114:5060;rport=5060;received=192.168.1.7;branch=z9hG4bKPj05dfacfb-a3f3-44e0-a215-4fcde87ed63d
Call-ID: 90dd4851-f349-45c2-b168-14c8bac16f88
From: <sip:300@172.17.0.2>;tag=a6e77797-b551-45ea-b043-6d201341e5e7
To: <sip:300@172.17.0.1;ob>;tag=z9hG4bKPj05dfacfb-a3f3-44e0-a215-4fcde87ed63d
CSeq: 23107 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.21.3
Content-Length:  0

I think the problem is that your exposed ports are on 172.17.0.1 while your container is running on 172.17.0.2; your INVITE is requesting media on 172.17.0.2, but your FPBX contact is using 103.248.120.114

I would try adding on your transports PJSIP section :

external_media_address=103.248.120.114
external_signaling_address=103.248.120.114

hope this helps

I would suggest running wireshark on the docker host and capturing both the SIP and RTP packets to make sure the ports are as you think they are.

103.248.120.114 - This is my public IP and FreePBX Auto detected it , should I remove this.
Also , where to add external_media_address=103.248.120.114
external_signaling_address=103.248.120.114

I think the problem is that your exposed ports are on 172.17.0.1 while your container is running on 172.17.0.2; your INVITE is requesting media on 172.17.0.2
I also see this problem but How to resolve this?

external_media_address and external_signaling_address are transport parameters.

Is 103.248.120.114 assigned to your docker host or is the host natted behind something that has that address?

Does the container need to communicate with the docker host for anything or does all traffic just pass through? If you don’t need the container to connect to anything on the host, you may want to consider using a macvlan network which would give the container direct access to the same LAN the host is on. It would eliminate the docker natting.

So , I disabled NAT from PBX and assign static local ip - 192.168.1.7 , which is my local IP.

<--- Transmitting SIP response (973 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPjce80deaef9714571acd05547b6bae7d6
Call-ID: aebddb33eadf444b95f53c8516f560a2
From: <sip:100@192.168.1.7>;tag=79033ecda0114b20b1f9dc40b19f9214
To: <sip:300@192.168.1.7>;tag=3cbcca14-d4e0-4de3-af34-a6a06a0233df
CSeq: 3135 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:172.17.0.2:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928263857 3928263859 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15002 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP response (973 bytes) to UDP:172.17.0.1:45164 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:62344;rport=45164;received=172.17.0.1;branch=z9hG4bKPjce80deaef9714571acd05547b6bae7d6
Call-ID: aebddb33eadf444b95f53c8516f560a2
From: <sip:100@192.168.1.7>;tag=79033ecda0114b20b1f9dc40b19f9214
To: <sip:300@192.168.1.7>;tag=3cbcca14-d4e0-4de3-af34-a6a06a0233df
CSeq: 3135 INVITE
Server: FPBX-15.0.23(18.23.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:172.17.0.2:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: "300" <sip:300@192.168.1.7>
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 3928263857 3928263859 IN IP4 172.17.0.2
s=Asterisk
c=IN IP4 172.17.0.2
t=0 0
m=audio 15002 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP request (415 bytes) to UDP:172.17.0.1:56685 --->
OPTIONS sip:300@172.17.0.1:56685;ob SIP/2.0
Via: SIP/2.0/UDP 172.17.0.2:5060;rport;branch=z9hG4bKPj549a0b82-c16c-40c2-9b9f-51f13b83c507
From: <sip:300@172.17.0.2>;tag=b952a082-5dfb-4f97-adac-5b92d65b00b3
To: <sip:300@172.17.0.1;ob>
Contact: <sip:300@172.17.0.2:5060>
Call-ID: 2caffc7e-7dca-4275-b0ec-e3309149c50e
CSeq: 63157 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.23(18.23.1)
Content-Length:  0

These are the logs now.
Sip.conf

#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a 
;extension to work to for example, those go here.  So you have extension 
;1000 defined in your system you start by creating a line [1000](+) in this 
;file.  Then on the next line add the extra parameter that is needed.  
;When the sip.conf is loaded it will append your additions to the end of 
;that extension. 
;
#include sip_custom_post.con
[general]
externip=191.168.1.7
localnet=192.168.1.0/255.255.255.0 ; Replace with your local network
nat=no

Pjsip.conf

#include pjsip_custom.conf
#include pjsip.transports.conf
#include pjsip.transports_custom_post.conf
#include pjsip.endpoint.conf
#include pjsip.endpoint_custom_post.conf
#include pjsip.aor.conf
#include pjsip.aor_custom_post.conf
#include pjsip.auth.conf
#include pjsip.auth_custom_post.conf
#include pjsip.registration.conf
#include pjsip.registration_custom_post.conf
#include pjsip.identify.conf
#include pjsip.identify_custom_post.conf


[global]
type=global
user_agent=FPBX-15.0.23(18.23.1)
use_callerid_contact=no
debug=no
keep_alive_interval=90
endpoint_identifier_order=ip,username,anonymous,header,auth_username
taskprocessor_overload_trigger=pjsip_only
#include pjsip_custom_post.conf

Still No Audio

@logikanet @gjoseph Any Suggestions?

192.168.1.7 is the local IP of what? Where did you assign it? It looks like the container still has the docker-assigned bridge address. If you want to assign the container a specific ip address from your LAN, you have to create a docker macvlan network on the host, remove the container from the default host network, then add the container to the new macvlan network.

Please include the received SIP packets in your logs, not just the responses. A diagram of what your local network looks like would also help: router, host, container with their respective IP addresses. It would also help to know where the client is. On your local LAN or remote.

Thanks , I will try creatinf the macvlan on My Windows Machine.

Even When I did this - ChatGPT

Its not working

You are using FreePBX. People here cannot assumed to know about the internal of FreePBX.

Also, sip.conf, etc., on FreePBX have fixed contents, so they are of no value to someone who does understand FreePBX. The actual details are in the additional files, someone debugging this from a FreePBX point of view (which you won’t really find here) would want screen shots of the GUI, as they may well not know how to interpret the actual Asterisk configuration.