No audio on Incoming call but audio is working on Outgoing call

Asterisk Asterisk SIP
1

My pbx is in nat configuration.there is two lan connected to the server ,one is from firewall for internet connection and other is for sip trunk from Jio Operator. When i remove externip and localnet audio works on incoming call but not in outgoing call

eno5 -- inet 10.10.1.20  netmask 255.255.255.0
eno6 - inet 100.65.150.52  netmask 255.255.255.240

SIP CONF
externip=210.212.232.251
localnet=10.10.1.0/255.255.255.0

[jio]
type=peer
;nat=force_rport,comedia
;disallow=all
;allow=all
;allow=gsm
allow=ulaw
;allow=alaw
;allow=g729
host=100.64.216.4
dtmfmode=rfc2833
qualify=yes
insecure=very
canreinvite=no
context=inbound

SIP LOG

<--- SIP read from UDP:100.64.216.4:5060 --->
INVITE sip:+914843503402@100.65.150.52 SIP/2.0
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632
Max-Forwards: 67
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Contact: <sip:100.64.216.4:5060>
Supported: 100rel,timer,replaces,histinfo,resource-priority,tdialog,sdp-anat
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,PRACK,MESSAGE,REFER,NOTIFY,INFO,OPTIONS
Session-Expires: 1800;refresher=uac
Min-SE:90
User-Agent: KZKDCASCAHPNBC11AC/v.7.20A.204.549
P-Called-Party-ID: <sip:+914843503402@kl.wln.ims.jio.com;user=phone>
Accept:application/sdp,application/3gpp-ims+xml
Content-Type: application/sdp
Content-Length: 407
d:no-fork
a:*;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
P-Visited-Network-ID:ims.mnc862.mcc405.3gppnetwork.org

v=0
o=- 879334127 1938591927 IN IP4 100.64.216.5
s=QC VOIP
c=IN IP4 100.64.216.5
t=0 0
m=audio 47752 RTP/AVP 0 8 18 105 101
c=IN IP4 100.64.216.5
b=RS:601
b=RR:2001
a=rtpmap:105 telephone-event/16000
a=fmtp:105 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:240
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
<------------->
--- (20 headers 20 lines) ---
Sending to 100.64.216.4:5060 (no NAT)
Sending to 100.64.216.4:5060 (no NAT)
Using INVITE request as basis request - 18317566691722022162114@100.64.216.4
Found peer 'jio' for '+919895909009' from 100.64.216.4:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 105
Found RTP audio format 101
Found unknown media description format telephone-event for ID 105
Found audio description format telephone-event for ID 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7ff2a0073d60 -- Strict RTP learning after remote address set to: 100.64.216.5:47752
Peer audio RTP is at port 100.64.216.5:47752
Looking for +914843503402 in inbound (domain 100.65.150.52)
sip_route_dump: route/path hop: <sip:100.64.216.4:5060>

<--- Transmitting (no NAT) to 100.64.216.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Length: 0


<------------>
    -- Executing [+914843503402@inbound:1] NoOp("SIP/jio-0000001f", "+914843503402,1") in new stack
    -- Executing [+914843503402@inbound:2] NoOp("SIP/jio-0000001f", "+919895909009,1") in new stack
    -- Executing [+914843503402@inbound:3] NoOp("SIP/jio-0000001f", "14843503402") in new stack
    -- Executing [+914843503402@inbound:4] NoOp("SIP/jio-0000001f", "") in new stack
    -- Executing [+914843503402@inbound:5] Set("SIP/jio-0000001f", "CALLERID(dnid)=4843503402") in new stack
    -- Executing [+914843503402@inbound:6] Set("SIP/jio-0000001f", "CALLERID(num)=+919895909009") in new stack
    -- Executing [+914843503402@inbound:7] NoOp("SIP/jio-0000001f", "+914843503402") in new stack
    -- Executing [+914843503402@inbound:8] Ringing("SIP/jio-0000001f", "") in new stack

<--- Transmitting (no NAT) to 100.64.216.4:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Length: 0


<------------>
    -- Executing [+914843503402@inbound:9] Set("SIP/jio-0000001f", "__TRANSFER_CONTEXT=call-transfer") in new stack
    -- Executing [+914843503402@inbound:10] Goto("SIP/jio-0000001f", "start,+914843503402,1") in new stack
    -- Goto (start,+914843503402,1)
    -- Executing [+914843503402@start:1] NoOp("SIP/jio-0000001f", "4843503402") in new stack
    -- Executing [+914843503402@start:2] NoOp("SIP/jio-0000001f", "") in new stack
    -- Executing [+914843503402@start:3] NoOp("SIP/jio-0000001f", "+914843503402") in new stack
    -- Executing [+914843503402@start:4] Set("SIP/jio-0000001f", "GROUP()=4843503402") in new stack
    -- Executing [+914843503402@start:5] NoOp("SIP/jio-0000001f", "1") in new stack
    -- Executing [+914843503402@start:6] Set("SIP/jio-0000001f", "GCOUNT=1") in new stack
    -- Executing [+914843503402@start:7] NoOp("SIP/jio-0000001f", "") in new stack
    -- Executing [+914843503402@start:8] AGI("SIP/jio-0000001f", "lookup") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/lookup
    -- lookup: DATABASE SUCCESS("CONNECTION", " SUCCESS")
    -- lookup: ENTER APPLICATION("FROM CALLER : +919895909009", " APP NAME : EXTENSION, APP VALUE : 1 ")
    -- lookup: INBOUND ROUTING("FROM CALLER : +919895909009", " DID : 4843503402 APP NAME : EXTENSION, APP VALUE : 1  ")
    -- lookup: INBOUND PATTERN MATCH FAILED("FROM CALLER : +919895909009", " DID : 4843503402 APP NAME : EXTENSION, APP VALUE : 1  ")
Audio is at 19598
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 100.64.216.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

<------------>
Retransmitting #1 (no NAT) to 100.64.216.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

---
Retransmitting #2 (no NAT) to 100.64.216.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

---
    -- lookup: EXTENSION APPLICATION("FROM CALLER : +919895909009", " EXTENSION : 787")
    -- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/records/1645095075.89-4843503402-9895909009-20220217-162115.gsm,b)
    -- AGI Script Executing Application: (Dial) Options: (SIP/787,40,tTM(attendcall))
  == Begin MixMonitor Recording SIP/jio-0000001f
  == Using SIP RTP CoS mark 5
Audio is at 16360
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 157.46.147.236:44544:
INVITE sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb SIP/2.0
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK2343d3f9;rport
Max-Forwards: 70
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>
Contact: <sip:+919895909009@210.212.232.251:5060>
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX certified/16.8-cert11
Date: Thu, 17 Feb 2022 10:51:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 1709518656 1709518656 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 16360 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/787
Retransmitting #3 (no NAT) to 100.64.216.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:157.46.147.236:44544 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK2343d3f9;rport=5060
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Retransmitting #4 (no NAT) to 100.64.216.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:157.46.147.236:44544 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK2343d3f9;rport=5060
Contact: <sip:787@157.46.147.236:44544;transport=UDP>
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>;tag=07e08a42
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.8 v2.10.17.2
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:787@157.46.147.236:44544;transport=UDP>
    -- SIP/787-00000020 is ringing
Retransmitting #5 (no NAT) to 100.64.216.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:157.46.147.236:44544 --->


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

<--- SIP read from UDP:157.46.147.236:44544 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK2343d3f9;rport=5060
Contact: <sip:787@157.46.147.236:44544;transport=UDP>
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>;tag=07e08a42
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.8 v2.10.17.2
Allow-Events: presence, kpml, talk
Content-Length: 325

v=0
o=Z 0 2 IN IP4 157.46.147.236
s=Z
c=IN IP4 157.46.147.236
t=0 0
m=audio 32805 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7ff37800c450 -- Strict RTP learning after remote address set to: 157.46.147.236:32805
Peer audio RTP is at port 157.46.147.236:32805
sip_route_dump: route/path hop: <sip:787@157.46.147.236:44544;transport=UDP>
Transmitting (NAT) to 157.46.147.236:44544:
ACK sip:787@157.46.147.236:44544;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK7e14fe29;rport
Max-Forwards: 70
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>;tag=07e08a42
Contact: <sip:+919895909009@210.212.232.251:5060>
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX certified/16.8-cert11
Content-Length: 0


---
    -- SIP/787-00000020 answered SIP/jio-0000001f
    -- Executing [s@macro-attendcall:1] NoOp("SIP/787-00000020", ""Call Answered 2022-02-17 16:21:19"") in new stack
    -- Executing [s@macro-attendcall:2] AGI("SIP/787-00000020", "answerlog") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/answerlog
       > 0x7ff37800c450 -- Strict RTP switching to RTP target address 157.46.147.236:32805 as source
    -- answerlog: DATABASE SUCCESS("CONNECTION", " SUCCESS")
    -- answerlog: ----------1645095075.89=================================
    -- <SIP/787-00000020>AGI Script answerlog completed, returning 0
    -- Channel SIP/787-00000020 joined 'simple_bridge' basic-bridge <d192f7b9-83d2-49be-8194-83e5ca983bc3>
    -- Channel SIP/jio-0000001f joined 'simple_bridge' basic-bridge <d192f7b9-83d2-49be-8194-83e5ca983bc3>
Really destroying SIP dialog 'cLo-mMrPHpPCa62rHhTEww..' Method: REGISTER
Retransmitting #6 (no NAT) to 100.64.216.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.64.216.4:5060;branch=z9hG4bKac149875632;received=100.64.216.4
From: <sip:+919895909009@100.64.216.4>;tag=1c344227972
To: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 1 INVITE
Server: Asterisk PBX certified/16.8-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+914843503402@210.212.232.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 109187263 109187263 IN IP4 210.212.232.251
s=Asterisk PBX certified/16.8-cert11
c=IN IP4 210.212.232.251
t=0 0
m=audio 19598 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=maxptime:150
a=sendrecv

---
[Feb 17 16:21:22] WARNING[160505]: chan_sip.c:4127 retrans_pkt: Retransmission timeout reached on transmission 18317566691722022162114@100.64.216.4 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Feb 17 16:21:22] WARNING[160505]: chan_sip.c:4151 retrans_pkt: Hanging up call 18317566691722022162114@100.64.216.4 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/jio-0000001f left 'simple_bridge' basic-bridge <d192f7b9-83d2-49be-8194-83e5ca983bc3>
    -- Channel SIP/787-00000020 left 'simple_bridge' basic-bridge <d192f7b9-83d2-49be-8194-83e5ca983bc3>
Scheduling destruction of SIP dialog '0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 157.46.147.236:44544:
BYE sip:787@157.46.147.236:44544;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK27a5fd05;rport
Max-Forwards: 70
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>;tag=07e08a42
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX certified/16.8-cert11
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
    -- <SIP/jio-0000001f>AGI Script lookup completed, returning 4
  == Spawn extension (start, +914843503402, 8) exited non-zero on 'SIP/jio-0000001f'
    -- Executing [h@start:1] NoOp("SIP/jio-0000001f", "ANSWERED") in new stack
    -- Executing [h@start:2] NoOp("SIP/jio-0000001f", "6") in new stack
    -- Executing [h@start:3] NoOp("SIP/jio-0000001f", "2022-02-17 16:21:15") in new stack
    -- Executing [h@start:4] NoOp("SIP/jio-0000001f", "1645095075.89") in new stack
    -- Executing [h@start:5] AGI("SIP/jio-0000001f", "calllog,6,ANSWERED,1645095075.89,H_START") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/calllog
    -- calllog,6,ANSWERED,1645095075.89,H_START: DATABASE SUCCESS("CONNECTION", " SUCCESS")
    -- calllog,6,ANSWERED,1645095075.89,H_START: H_START
    -- <SIP/jio-0000001f>AGI Script calllog completed, returning 0
Scheduling destruction of SIP dialog '18317566691722022162114@100.64.216.4' in 6400 ms (Method: INVITE)
  == MixMonitor close filestream (mixed)
set_destination: Parsing <sip:100.64.216.4:5060> for address/port to send to
set_destination: set destination to 100.64.216.4:5060
Reliably Transmitting (no NAT) to 100.64.216.4:5060:
BYE sip:100.64.216.4:5060 SIP/2.0
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK1a52fa02
Max-Forwards: 70
From: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
To: <sip:+919895909009@100.64.216.4>;tag=1c344227972
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 102 BYE
User-Agent: Asterisk PBX certified/16.8-cert11
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
  == End MixMonitor Recording SIP/jio-0000001f

<--- SIP read from UDP:157.46.147.236:44544 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK27a5fd05;rport=5060
Contact: <sip:787@157.46.147.236:44544;transport=UDP>
To: <sip:787@157.46.147.236:44544;transport=UDP;rinstance=f96643ab4729f5fb>;tag=07e08a42
From: <sip:+919895909009@210.212.232.251>;tag=as47c9693d
Call-ID: 0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060
CSeq: 103 BYE
User-Agent: Z 5.5.8 v2.10.17.2
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '0565b0fb109cc0af764ebce919cb1d30@210.212.232.251:5060' Method: INVITE
Retransmitting #1 (no NAT) to 100.64.216.4:5060:
BYE sip:100.64.216.4:5060 SIP/2.0
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK1a52fa02
Max-Forwards: 70
From: <sip:+914843503402@100.65.150.52>;tag=as64f077ea
To: <sip:+919895909009@100.64.216.4>;tag=1c344227972
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 102 BYE
User-Agent: Asterisk PBX certified/16.8-cert11
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:100.64.216.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 210.212.232.251:5060;branch=z9hG4bK1a52fa02
Max-Forwards:70
From: <sip:+914843503402@100.64.216.4>;tag=as64f077ea
To: <sip:+919895909009@100.65.150.52>;tag=1c344227972
Call-ID: 18317566691722022162114@100.64.216.4
CSeq: 102 BYE
Contact: <sip:100.64.216.4:5060>
Server: KZKDCASCAHPNBC11AC/v.7.20A.204.549
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
2

More details of your very non-internet network configuration is necessary, e.g. I assume that devices on 100… cannot route to devices on 10… Also 100… isn’t really local, it is “shared”.

You need to urgently plan to move to chan_pjsip as chan_sip is deprecated, and schedule for removal next year.

insecure=very has been deprecated for a long time, and, looking at the source code for the leading edge chan_sip is no longer recognized, which begs the question as to which version of Asterisk you are using. There are other deprecated options but I haven’t checked whether they are still recognized. Looking at your logs, you do not need to reduce the security, with any value of this option.

I see you are using a certified version of Asterisk. In that case you should be using the commercial support contract for which these versions are intended. Also, as far as I know, no current certified versions cover chan_sip, so you will definitely need to move to chan_pjsip. If you don’t have a support contract, please move to Asterisk 16.24.0 (or 18.10) before continuing.

Your immediate problem is that you haven’t listed 100.65.150.48/20 as a localnet. Note, as this is almost certainly disjoint should make sure that everything uses directmedia=no (which is the correct name for canreinvite=no).

Incidentally, your problem is worse than no audio. The bad Contact header means the call will get shutdown in no more than about 30 seconds regardless of any audio problem.

3

i added , but still there is no audio

localnet=100.65.150.48/20

WAN IP Address with subnet mask - 100.65.150.52/255.255.255.240
Gateway IP Address - 100.65.150.49
SBC IP - 100.64.216.4

Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
0.0.0.0         10.10.1.1       0.0.0.0         UG    0      0        0 eno5
10.10.1.0       0.0.0.0         255.255.255.0   U     0      0        0 eno5
100.64.216.4    100.65.150.49   255.255.255.255 UGH   0      0        0 eno6
100.64.216.5    100.65.150.49   255.255.255.255 UGH   0      0        0 eno6
100.65.150.48   0.0.0.0         255.255.255.240 U     0      0        0 eno6
172.17.0.0      0.0.0.0         255.255.0.0     U     0      0        0 docker0
172.20.0.0      0.0.0.0         255.255.0.0     U     0      0        0 br-f3f30ba712ac
4

100.64.216.0/20 also needs to be a local net. Possibly the whole of 100.64.0.0/10

5

Now its working , thank you david

6

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