Asterisk is trying to generate a call but not dialing

Trying for 30 secs, and then the call gets declined. Nobody picked up for 30000ms. This is the error in the console.

You’d have to actually provide logging and more information. If SIP is involved, then the SIP trace.

<— SIP read from WS:192.168.2.8:52194 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK9846333
To: sip:231017@bp.erss.in
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
CSeq: 16 REGISTER
Call-ID: 9k46842rl8q7le1h69ps
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231017”, realm=“asterisk”, nonce=“21cd12b1”, uri=“sip:bp.erss.in”, response=“63fec522f11841cb821a4c73c1f86f37”
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52194 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK9846333;received=192.168.2.8;rport=52194
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
To: sip:231017@bp.erss.in;tag=as6e6068c1
Call-ID: 9k46842rl8q7le1h69ps
CSeq: 16 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5ad8605b”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9k46842rl8q7le1h69ps’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:192.168.2.8:52194 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK2683030
To: sip:231017@bp.erss.in
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
CSeq: 17 REGISTER
Call-ID: 9k46842rl8q7le1h69ps
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231017”, realm=“asterisk”, nonce=“5ad8605b”, uri=“sip:bp.erss.in”, response=“3982993e7033859fed5267f08e2ed30e”
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52194 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK2683030;received=192.168.2.8;rport=52194
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
To: sip:231017@bp.erss.in;tag=as6e6068c1
Call-ID: 9k46842rl8q7le1h69ps
CSeq: 17 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=300
Date: Mon, 19 Sep 2022 08:31:47 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9k46842rl8q7le1h69ps’ in 32000 ms (Method: REGISTER)
– Nobody picked up in 30000 ms
Dial End message for SIP/231017-0000000b, SIP/231025-0000000c: 1663576310.00782994
0x7f3ff4000fc0 - Processing Dial End message for channel SIP/231017-0000000b, peer SIP/231025-0000000c
Scheduling destruction of SIP dialog ‘6e790b8041046364775a94ae1104df2e@192.168.2.14:5060’ in 32000 ms (Method: INVITE)
0x7f3ff4000fc0 - Transitioning CDR for SIP/231017-0000000b from state Dial to Finalized
0x7f3ff4001c50 - Transitioning CDR for SIP/231025-0000000c from state Single to Finalized
0x7f3ff4001c50 - Beginning finalize/dispatch for SIP/231025-0000000c
0x7f3ff4001c50 - Dispatching CDR for Party A SIP/231025-0000000c, Party B
– Executing [231025@test:3] Hangup(“SIP/231017-0000000b”, “”) in new stack
== Spawn extension (test, 231025, 3) exited non-zero on ‘SIP/231017-0000000b’
Scheduling destruction of SIP dialog ‘9k468bbn0vnp3d36icvj’ in 32000 ms (Method: INVITE)

<— Reliably Transmitting (NAT) to 192.168.2.8:52194 —>
SIP/2.0 603 Declined
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK4970342;received=192.168.2.8;rport=52194
From: “prathibha” sip:231017@bp.erss.in;tag=c7uh9jmc2j
To: sip:231025@bp.erss.in;tag=as63481c20
Call-ID: 9k468bbn0vnp3d36icvj
CSeq: 2 INVITE
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
0x7f3ff4001c50 - Created CDR for channel SIP/231017-0000000b
0x7f3ff4001c50 - Transitioning CDR for SIP/231017-0000000b from state NONE to Single
0x7f3ff4001c50 - Transitioning CDR for SIP/231017-0000000b from state Single to Finalized
0x7f3ff4000fc0 - Beginning finalize/dispatch for SIP/231017-0000000b
0x7f3ff4000fc0 - Dispatching CDR for Party A SIP/231017-0000000b, Party B SIP/231025-0000000c
[Sep 19 14:01:50] ERROR[12112]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle. CDR failed.

<— SIP read from WS:192.168.2.8:52194 —>
ACK sip:231025@bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK4970342
To: sip:231025@bp.erss.in;tag=as63481c20
From: “prathibha” sip:231017@bp.erss.in;tag=c7uh9jmc2j
Call-ID: 9k468bbn0vnp3d36icvj
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from WS:192.168.2.8:52195 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK3329133
To: sip:231025@bp.erss.in
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
CSeq: 16 REGISTER
Call-ID: 0vjbdj56g3gmda1pnsmu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231025”, realm=“asterisk”, nonce=“03b8beb2”, uri=“sip:bp.erss.in”, response=“071335d29acb2f15d2886071591c6567”
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52195 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK3329133;received=192.168.2.8;rport=52195
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
To: sip:231025@bp.erss.in;tag=as177f29f7
Call-ID: 0vjbdj56g3gmda1pnsmu
CSeq: 16 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“6e96b715”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0vjbdj56g3gmda1pnsmu’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:192.168.2.8:52195 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK6432181
To: sip:231025@bp.erss.in
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
CSeq: 17 REGISTER
Call-ID: 0vjbdj56g3gmda1pnsmu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231025”, realm=“asterisk”, nonce=“6e96b715”, uri=“sip:bp.erss.in”, response=“0b32d9bffcd5b85de584ddde2d255754”
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52195 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK6432181;received=192.168.2.8;rport=52195
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
To: sip:231025@bp.erss.in;tag=as177f29f7
Call-ID: 0vjbdj56g3gmda1pnsmu
CSeq: 17 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Date: Mon, 19 Sep 2022 08:32:16 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0vjbdj56g3gmda1pnsmu’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘9k46842rl8q7le1h69ps’ Method: REGISTER
Really destroying SIP dialog ‘6e790b8041046364775a94ae1104df2e@192.168.2.14:5060’ Method: INVITE
Really destroying SIP dialog ‘9k468bbn0vnp3d36icvj’ Method: ACK
Really destroying SIP dialog ‘0vjbdj56g3gmda1pnsmu’ Method: REGISTER

<— SIP read from WS:192.168.2.8:52194 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK7383588
To: sip:231017@bp.erss.in
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
CSeq: 18 REGISTER
Call-ID: 9k46842rl8q7le1h69ps
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231017”, realm=“asterisk”, nonce=“5ad8605b”, uri=“sip:bp.erss.in”, response=“3982993e7033859fed5267f08e2ed30e”
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52194 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK7383588;received=192.168.2.8;rport=52194
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
To: sip:231017@bp.erss.in;tag=as4584224b
Call-ID: 9k46842rl8q7le1h69ps
CSeq: 18 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“6a543011”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9k46842rl8q7le1h69ps’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:192.168.2.8:52194 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK6581748
To: sip:231017@bp.erss.in
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
CSeq: 19 REGISTER
Call-ID: 9k46842rl8q7le1h69ps
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231017”, realm=“asterisk”, nonce=“6a543011”, uri=“sip:bp.erss.in”, response=“adcde933b81cb54cce40ec89f14004cf”
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52194 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK6581748;received=192.168.2.8;rport=52194
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
To: sip:231017@bp.erss.in;tag=as4584224b
Call-ID: 9k46842rl8q7le1h69ps
CSeq: 19 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=300
Date: Mon, 19 Sep 2022 08:36:44 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9k46842rl8q7le1h69ps’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:192.168.2.8:52195 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK158520
To: sip:231025@bp.erss.in
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
CSeq: 18 REGISTER
Call-ID: 0vjbdj56g3gmda1pnsmu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231025”, realm=“asterisk”, nonce=“6e96b715”, uri=“sip:bp.erss.in”, response=“0b32d9bffcd5b85de584ddde2d255754”
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52195 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK158520;received=192.168.2.8;rport=52195
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
To: sip:231025@bp.erss.in;tag=as6393c4f2
Call-ID: 0vjbdj56g3gmda1pnsmu
CSeq: 18 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“33544a9d”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0vjbdj56g3gmda1pnsmu’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:192.168.2.8:52195 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK5220333
To: sip:231025@bp.erss.in
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
CSeq: 19 REGISTER
Call-ID: 0vjbdj56g3gmda1pnsmu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231025”, realm=“asterisk”, nonce=“33544a9d”, uri=“sip:bp.erss.in”, response=“b051f2fa0e05c2a45cf595df13c49ff2”
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52195 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK5220333;received=192.168.2.8;rport=52195
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
To: sip:231025@bp.erss.in;tag=as6393c4f2
Call-ID: 0vjbdj56g3gmda1pnsmu
CSeq: 19 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Date: Mon, 19 Sep 2022 08:37:14 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0vjbdj56g3gmda1pnsmu’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘9k46842rl8q7le1h69ps’ Method: REGISTER
Really destroying SIP dialog ‘0vjbdj56g3gmda1pnsmu’ Method: REGISTER

<— SIP read from WS:192.168.2.8:52194 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK5987607
To: sip:231017@bp.erss.in
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
CSeq: 20 REGISTER
Call-ID: 9k46842rl8q7le1h69ps
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231017”, realm=“asterisk”, nonce=“6a543011”, uri=“sip:bp.erss.in”, response=“adcde933b81cb54cce40ec89f14004cf”
Contact: sip:22pacab1@192.0.2.142;transport=wss;expires=0
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (12 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52194 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.142;branch=z9hG4bK5987607;received=192.168.2.8;rport=52194
From: “prathibha” sip:231017@bp.erss.in;tag=uneon461fk
To: sip:231017@bp.erss.in;tag=as02f2afb8
Call-ID: 9k46842rl8q7le1h69ps
CSeq: 20 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3bf52318”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9k46842rl8q7le1h69ps’ in 32000 ms (Method: REGISTER)
== WebSocket connection from ‘192.168.2.8:52427’ for protocol ‘sip’ accepted using version ‘13’

<— SIP read from WS:192.168.2.8:52195 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK7223413
To: sip:231025@bp.erss.in
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
CSeq: 20 REGISTER
Call-ID: 0vjbdj56g3gmda1pnsmu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231025”, realm=“asterisk”, nonce=“33544a9d”, uri=“sip:bp.erss.in”, response=“b051f2fa0e05c2a45cf595df13c49ff2”
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52195 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK7223413;received=192.168.2.8;rport=52195
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
To: sip:231025@bp.erss.in;tag=as6ee9b882
Call-ID: 0vjbdj56g3gmda1pnsmu
CSeq: 20 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1aec36da”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0vjbdj56g3gmda1pnsmu’ in 32000 ms (Method: REGISTER)

<— SIP read from WS:192.168.2.8:52195 —>
REGISTER sip:bp.erss.in SIP/2.0
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK5926554
To: sip:231025@bp.erss.in
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
CSeq: 21 REGISTER
Call-ID: 0vjbdj56g3gmda1pnsmu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“231025”, realm=“asterisk”, nonce=“1aec36da”, uri=“sip:bp.erss.in”, response=“3d339bb83a7dd0168de5651109d0424b”
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.8 (SIPJS - 0.20.0)
Content-Length: 0

<------------->
— (13 headers 0 lines) —

<— Transmitting (NAT) to 192.168.2.8:52195 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.180;branch=z9hG4bK5926554;received=192.168.2.8;rport=52195
From: “prathibha” sip:231025@bp.erss.in;tag=9e099tpqbg
To: sip:231025@bp.erss.in;tag=as6ee9b882
Call-ID: 0vjbdj56g3gmda1pnsmu
CSeq: 21 REGISTER
Server: Asterisk PBX certified/13.21-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: sip:fkmq0i9g@192.0.2.180;transport=wss;expires=300
Date: Mon, 19 Sep 2022 08:42:12 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0vjbdj56g3gmda1pnsmu’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘9k46842rl8q7le1h69ps’ Method: REGISTER
== WebSocket connection from ‘192.168.2.8:52194’ forcefully closed due to fatal write error
Really destroying SIP dialog ‘0vjbdj56g3gmda1pnsmu’ Method: REGISTER

It is a webRTC call.

Please provide the log for the whole call. It would also help if you removed the REGISTER transactions, at least initially, as they clutter the log. Furthermore, please mark up your logs as pre-formatted text, otherwise the forum garbles them.

Also, you should be moving to chan_pjsip. As you are using WebRTC, I can think of no reason not to, and chan_sip is, effectively, unsupported.