Configuring trunk to make external calls

Good day,

I am trying to configure trunk to make external calls, first of all since my external phone only supports amrwb encoding, I refer to https://github.com/traud/asterisk-amr webpage to configure amrwb encoding. Then the sip configuration I was using before was only able to accomplish single communication (external calls could not be dialed into asterisk), so I switched to pjsip hoping to solve this problem.
But so far I’m running into some problems,here is my pjsip and extensions configuration.Currently this code also only enables asterisk to dial into the external phone, and when the call is connected asterisk talk the external phone can not hear the sound.

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0:5060
;tcpenable=yes
;tcpbind=0.0.0.0:5060
;srvlookup=yes
;nat=force_rport,comedia

[1001-auth]
type=auth
auth_type=userpass
username=1001
password=1001

[1001]
type=endpoint
context=from-internal
disallow=all
allow=amrwb
auth=1001-auth
aors=1001
transport=simpletrans
;dtls_auto_generate_cert=yes
;webrtc=yes
;rtp_symmetric=yes
;force_rport=yes
;rewrite_contact=yes

[1001]
type=aor
max_contacts=5
remove_existing=yes
contact=sip:192.168.1.2 ; 1001 终端的 IP 地址

; 配置 Trunk: my_sip_trunk

[mytrunk]
type=registration
transport=simpletrans
server_uri=sip:192.168.4.1
client_uri=sip:192.168.1.19
retry_interal=60
endpoint=my_sip_trunk

[my_sip_trunk-auth]
type=auth
auth_type=none
;username= ; 对方 SIP 服务器的用户名
;password=1001password ; 对方 SIP 服务器的密码

[my_sip_trunk]
type=endpoint
context=outbound-calls
disallow=all
allow=amrwb
aors=my_sip_trunk
outbound_auth=my_sip_trunk-auth
;from_user=13801380087 ; 对方号码
transport=simpletrans
force_rport=yes ;It’s a good idea to read the configuration help for each
direct_media=no ;of these options.
ice_support=yes

[my_sip_trunk]
type=aor
contact=sip:192.168.4.1 ; Trunk 目标地址

[my_sip_trunk]
type=identify
endpoint=my_sip_trunk
match=192.168.4.1 ; 匹配 Trunk 目标地址

extensions
[from-internal]
exten => 1001,1,Dial(PJSIP/1001,20)
exten => 1001,n,Hangup()

exten => 13801380087,1,Dial(PJSIP/13801380087@my_sip_trunk,20)
exten => 13801380087,n,Hangup()

[outbond-calls]
exten => _X.,1,Dial(PJSIP/${EXTEN}@my_sip_trunk,20)
exten => _X.,n,Hangup()

Here are some details of the console

[Jan 12 17:22:13] WARNING[32981]: pjproject: <?>: sip_transport.c Dropping 633 bytes packet from UDP 192.168.4.1:5060 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 26

sdr@sdr:~/Desktop$ sudo asterisk -r
[sudo] password for sdr:
Asterisk 20.9.3, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Running as user ‘asterisk’
Running under group ‘asterisk’
Connected to Asterisk 20.9.3 currently running on sdr (pid = 32954)
sdr*CLI> pjsip set logger on
PJSIP Logging enabled
<— Received SIP request (1417 bytes) from UDP:192.168.1.2:11825 —>
INVITE sip:13801380087@192.168.4.1 SIP/2.0
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-daf4f095f32ab15929c5b8f64835b707
Max-Forwards: 70
Contact: “1001” sip:1001@192.168.1.2:11825;transport=udp;registering_acc=192_168_1_19
User-Agent: Jitsi2.10.5550Windows 10
Content-Type: application/sdp
Content-Length: 922

v=0
o=1001-jitsi.org 0 0 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 5213 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 105 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=ptime:20
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:105 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5215 RTP/AVP 106 99
a=recvonly
a=rtpmap:106 H264/90000
a=fmtp:106 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:106 send * recv [x=[1:2880],y=[1:1800]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[1:2880],y=[1:1800]]

<— Transmitting SIP response (560 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-daf4f095f32ab15929c5b8f64835b707
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=z9hG4bK-373834-daf4f095f32ab15929c5b8f64835b707
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1736675348/abf6168fe3bea7d38484b8d6425428ed”,opaque=“4b4e50ea713cd9e1”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.9.3
Content-Length: 0

<— Received SIP request (381 bytes) from UDP:192.168.1.2:11825 —>
ACK sip:13801380087@192.168.4.1 SIP/2.0
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=z9hG4bK-373834-daf4f095f32ab15929c5b8f64835b707
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-daf4f095f32ab15929c5b8f64835b707
CSeq: 1 ACK
Content-Length: 0

<— Received SIP request (1677 bytes) from UDP:192.168.1.2:11825 —>
INVITE sip:13801380087@192.168.4.1 SIP/2.0
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>
Max-Forwards: 70
Contact: “1001” sip:1001@192.168.1.2:11825;transport=udp;registering_acc=192_168_1_19
User-Agent: Jitsi2.10.5550Windows 10
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-5fc6d4cb1797714a8131342638410221
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1736675348/abf6168fe3bea7d38484b8d6425428ed”,uri=“sip:13801380087@192.168.4.1”,response=“c9280cf8ec28669ce91bd7902a15b0bd”,algorithm=MD5,opaque=“4b4e50ea713cd9e1”,qop=auth,cnonce=“xyz”,nc=00000001
Content-Length: 922

v=0
o=1001-jitsi.org 0 0 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 5213 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 105 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=ptime:20
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:105 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5215 RTP/AVP 106 99
a=recvonly
a=rtpmap:106 H264/90000
a=fmtp:106 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:106 send * recv [x=[1:2880],y=[1:1800]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[1:2880],y=[1:1800]]

<— Transmitting SIP response (356 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-5fc6d4cb1797714a8131342638410221
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>
CSeq: 2 INVITE
Server: Asterisk PBX 20.9.3
Content-Length: 0

<— Transmitting SIP request (1202 bytes) to UDP:192.168.4.1:5060 —>
INVITE sip:13801380087@192.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPj9b512c9d-121f-40c2-a74b-4b7502db1edc
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>
Contact: <sip:asterisk@192.168.1.19:5060>
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27019 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.9.3
Content-Type: application/sdp
Content-Length: 522

v=0
o=- 682013853 682013853 IN IP4 192.168.1.19
s=Asterisk
c=IN IP4 192.168.1.19
t=0 0
m=audio 10500 RTP/AVP 98 101 106
a=ice-ufrag:14c200ce114c9f207f980d9360487eb3
a=ice-pwd:1dad859877357d6168cb9c063ab4f977
a=candidate:Hc0a80113 1 UDP 2130706431 192.168.1.19 10500 typ host
a=candidate:Hc0a80113 2 UDP 2130706430 192.168.1.19 10501 typ host
a=rtpmap:98 AMR-WB/16000
a=rtpmap:101 telephone-event/16000
a=fmtp:101 0-16
a=rtpmap:106 telephone-event/8000
a=fmtp:106 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<— Received SIP response (418 bytes) from UDP:192.168.4.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPj9b512c9d-121f-40c2-a74b-4b7502db1edc
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27019 INVITE
Contact: <sip:460110123456787@192.168.4.1:5060>
Max-Forwards: 70
Server: Amarisoft-IMS-2024-10-25
Content-Length: 0

<— Received SIP request (632 bytes) from UDP:192.168.1.2:11825 —>
OPTIONS sip:192.168.1.19 SIP/2.0
Call-ID: 54bd89eb45cf9c6efd7507a77c4881ff@0:0:0:0:0:0:0:0
CSeq: 532 OPTIONS
From: “1001” <sip:1001@192.168.1.19>;tag=e56c82a0
To: “1001” <sip:1001@192.168.1.19>
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-6f3407d269f1b5c7ccacabb62309ecbc
Max-Forwards: 70
Contact: “1001” sip:1001@192.168.1.2:11825;transport=udp;registering_acc=192_168_1_19
User-Agent: Jitsi2.10.5550Windows 10
Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
Content-Length: 0

<— Transmitting SIP response (564 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-6f3407d269f1b5c7ccacabb62309ecbc
Call-ID: 54bd89eb45cf9c6efd7507a77c4881ff@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=e56c82a0
To: “1001” <sip:1001@192.168.1.19>;tag=z9hG4bK-373834-6f3407d269f1b5c7ccacabb62309ecbc
CSeq: 532 OPTIONS
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1736675348/abf6168fe3bea7d38484b8d6425428ed”,opaque=“0da7ca8739e5fc8b”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.9.3
Content-Length: 0

<— Received SIP response (440 bytes) from UDP:192.168.4.1:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPj9b512c9d-121f-40c2-a74b-4b7502db1edc
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>;tag=17622169355288ee
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27019 INVITE
Contact: <sip:460110123456787@192.168.4.1:5060>
Max-Forwards: 70
Server: Amarisoft-IMS-2024-10-25
Content-Length: 0

<— Transmitting SIP response (549 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-5fc6d4cb1797714a8131342638410221
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=69abac92-160c-46f4-b220-b3cd01e5b3a5
CSeq: 2 INVITE
Server: Asterisk PBX 20.9.3
Contact: sip:[192.168.1.19:5060](http://192.168.1.19:5060)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length: 0

<— Received SIP response (970 bytes) from UDP:192.168.4.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPj9b512c9d-121f-40c2-a74b-4b7502db1edc
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>;tag=17622169355288ee
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27019 INVITE
Contact: <sip:460110123456787@192.168.4.1:5060>
Content-Type: application/sdp
Allow: REGISTER,SUBSCRIBE,INVITE,ACK,CANCEL,PRACK,UPDATE,BYE,MESSAGE,INFO
Max-Forwards: 70
Server: Amarisoft-IMS-2024-10-25
Session-Expires: 1800;refresher=uac
Require: timer
Content-Length: 372

v=0
o=Amarisoft-IMS-2024-10-25 3296 1 IN IP4 192.168.4.1
s=Amarisoft-IMS-2024-10-25
c=IN IP4 192.168.4.1
b=AS:41
b=RR:1537
b=RS:512
t=0 0
m=audio 10002 RTP/AVP 98 101
b=AS:41
b=RR:1537
b=RS:512
a=rtpmap:98 AMR-WB/16000/1
a=fmtp:98 mode-change-capability=2;max-red=0
a=rtpmap:101 telephone-event/16000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
a=sendrecv

<— Transmitting SIP request (417 bytes) to UDP:192.168.4.1:5060 —>
ACK sip:460110123456787@192.168.4.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPjfd55ffd8-04e7-4cfd-8ca6-acbc07da3393
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>;tag=17622169355288ee
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27019 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.9.3
Content-Length: 0

<— Transmitting SIP response (877 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-5fc6d4cb1797714a8131342638410221
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=69abac92-160c-46f4-b220-b3cd01e5b3a5
CSeq: 2 INVITE
Server: Asterisk PBX 20.9.3
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: sip:[192.168.1.19:5060](http://192.168.1.19:5060)
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 0 2 IN IP4 192.168.1.19
s=Asterisk
c=IN IP4 192.168.1.19
t=0 0
m=audio 11768 RTP/AVP 105 101
a=rtpmap:105 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
m=video 0 RTP/AVP 106 99

<— Received SIP request (751 bytes) from UDP:192.168.1.2:11825 —>
ACK sip:192.168.1.19:5060 SIP/2.0
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-aa62f1f18da8f04a3085b2e9a7a3ef3f
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=69abac92-160c-46f4-b220-b3cd01e5b3a5
Max-Forwards: 70
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1736675348/abf6168fe3bea7d38484b8d6425428ed”,uri=“sip:13801380087@192.168.4.1”,response=“c9280cf8ec28669ce91bd7902a15b0bd”,algorithm=MD5,opaque=“4b4e50ea713cd9e1”,qop=auth,cnonce=“xyz”,nc=00000001
Contact: “1001” sip:1001@192.168.1.2:11825;transport=udp;registering_acc=192_168_1_19
User-Agent: Jitsi2.10.5550Windows 10
Content-Length: 0

[Jan 12 17:49:21] WARNING[32981]: pjproject: <?>: sip_transport.c Dropping 633 bytes packet from UDP 192.168.4.1:5060 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 26

[Jan 12 17:49:23] WARNING[32981]: pjproject: <?>: sip_transport.c Dropping 633 bytes packet from UDP 192.168.4.1:5060 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 26

<— Received SIP request (751 bytes) from UDP:192.168.1.2:11825 —>
BYE sip:192.168.1.19:5060 SIP/2.0
CSeq: 3 BYE
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=69abac92-160c-46f4-b220-b3cd01e5b3a5
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
Max-Forwards: 70
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-f4c3b4beac71504b99a55e57aed3d4cf
Contact: “1001” sip:1001@192.168.1.2:11825;transport=udp;registering_acc=192_168_1_19
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1736675348/abf6168fe3bea7d38484b8d6425428ed”,uri=“sip:13801380087@192.168.4.1”,response=“c9280cf8ec28669ce91bd7902a15b0bd”,algorithm=MD5,opaque=“4b4e50ea713cd9e1”,qop=auth,cnonce=“xyz”,nc=00000001
User-Agent: Jitsi2.10.5550Windows 10
Content-Length: 0

<— Transmitting SIP response (390 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-f4c3b4beac71504b99a55e57aed3d4cf
Call-ID: 2e59fb3584ee1da9fe5bb9f13a064b86@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=d3d70b6e
To: <sip:13801380087@192.168.4.1>;tag=69abac92-160c-46f4-b220-b3cd01e5b3a5
CSeq: 3 BYE
Server: Asterisk PBX 20.9.3
Content-Length: 0

<— Transmitting SIP request (441 bytes) to UDP:192.168.4.1:5060 —>
BYE sip:460110123456787@192.168.4.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPj58c335a6-19e0-47ab-9282-fbab7c455433
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>;tag=17622169355288ee
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27020 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.9.3
Content-Length: 0

<— Received SIP response (383 bytes) from UDP:192.168.4.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.19:5060;rport;branch=z9hG4bKPj58c335a6-19e0-47ab-9282-fbab7c455433
From: “1001” <sip:1001@192.168.1.19>;tag=14e533f5-75ab-4037-a11b-36caee32ae6f
To: <sip:13801380087@192.168.4.1>;tag=17622169355288ee
Call-ID: 0effc7c0-8e17-4469-94a8-3d6a66d637b2
CSeq: 27020 BYE
Max-Forwards: 70
Server: Amarisoft-IMS-2024-10-25
Content-Length: 0

[Jan 12 17:49:27] WARNING[32981]: pjproject: <?>: sip_transport.c Dropping 633 bytes packet from UDP 192.168.4.1:5060 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 26

<— Received SIP request (632 bytes) from UDP:192.168.1.2:11825 —>
OPTIONS sip:192.168.1.19 SIP/2.0
Call-ID: 02a4ab463cca96ded1a338c738149723@0:0:0:0:0:0:0:0
CSeq: 533 OPTIONS
From: “1001” <sip:1001@192.168.1.19>;tag=780cab53
To: “1001” <sip:1001@192.168.1.19>
Via: SIP/2.0/UDP 192.168.1.2:11825;branch=z9hG4bK-373834-597febdd94079f2cc64761047eb9dcba
Max-Forwards: 70
Contact: “1001” sip:1001@192.168.1.2:11825;transport=udp;registering_acc=192_168_1_19
User-Agent: Jitsi2.10.5550Windows 10
Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
Content-Length: 0

<— Transmitting SIP response (564 bytes) to UDP:192.168.1.2:11825 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:11825;rport=11825;received=192.168.1.2;branch=z9hG4bK-373834-597febdd94079f2cc64761047eb9dcba
Call-ID: 02a4ab463cca96ded1a338c738149723@0:0:0:0:0:0:0:0
From: “1001” <sip:1001@192.168.1.19>;tag=780cab53
To: “1001” <sip:1001@192.168.1.19>;tag=z9hG4bK-373834-597febdd94079f2cc64761047eb9dcba
CSeq: 533 OPTIONS
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1736675373/2f357551155c59eb99e25a48c6ff268a”,opaque=“47db6e171f53e196”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.9.3
Content-Length: 0

[Jan 12 17:49:35] WARNING[32981]: pjproject: <?>: sip_transport.c Dropping 633 bytes packet from UDP 192.168.4.1:5060 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 26