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

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

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.

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

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

Now its working , thank you david

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