Custom Header Information lost in an Outbound Calls

We operate two Asterisk servers, each hosting WebRTC users who are capable of making calls to one another. Our goal is to transmit custom header information from a user on one server to a user on the other server. While we can observe our custom header data in the CLI of the sending Asterisk server, this information does not appear at the receiving end.

Note : Both the server has identical extension and pjsip configurations except for IP in the pjsip file for an outbound call

pjsip conf:

[global]
type=global
user_agent=ULG 1

[wss]
type=transport
protocol=wss ; Use 'wss' for WebSocket Secure
bind=0.0.0.0 ; Replace with your Asterisk server's IP and secure port

[udp]
type=transport
protocol=udp
bind=0.0.0.0
tos=af42
cos=3

[tcp]
type=transport
protocol=tcp
bind=0.0.0.0

[endpoint-basic](!)
type=endpoint
transport=wss
context=outside ; Replace with the context you use in your dialplan
disallow=all
allow=ulaw,alaw,opus
auth=auth_user
aors=sip_user
webrtc=yes ; Enable WebRTC support
direct_media = no
send_pai=yes

; Authentication for the User
[auth_user](!)
type=auth
auth_type=userpass

; AOR for the User
[sip_user](!)
type=aor
max_contacts=5 ; Adjust as needed

; Specific User Configuration
[001](endpoint-basic)
auth=auth001
aors=001

[auth001](auth_user)
username=001 ; Replace with your username
password=1234 ; Replace with a strong password

[001](sip_user)

[User2](endpoint-basic)
auth=authUser2
aors=User2

[authUser2](auth_user)
username=User2 ; Replace with your username
password=1234 ; Replace with a strong password

[User2](sip_user)

;==================softphone=============
[auth100]
type=auth
username=100
password=1234
auth_type=userpass

[100]
type=aor
max_contacts=2
qualify_frequency=60

[100]
type=endpoint
context=outside
auth=auth100
aors=100
transport=tcp
disallow=all
allow=ulaw,alaw,opus
direct_media = no



[outside-call]
type=aor
contact=sip:other-Asterisk-server-ip

[outside-call]
type=endpoint
context=from-outside-call
disallow=all
allow=ulaw,alaw,opus
aors=outside-call
transport=wss
send_pai=yes
direct_media = yes

[outside-call]
type=identify
endpoint=outside-call
match=other-Asterisk-server-ip

extension.conf

[outside]
exten => _XXX,1,NoOp(Calling: ${EXTEN}))
same => n,Set(X-MyHeader=${PJSIP_HEADER(read,X-MyHeader)})
  same => n,NoOp(Received X-MyHeader: ${X-MyHeader})
same => n,Dial(PJSIP/${EXTEN}@outside-call)
same => n,Hangup()

[from-outside-call]
exten => _XXX,1,NoOp(Calling: ${EXTEN}))
 same => n,Set(X-MyHeader=${PJSIP_HEADER(read,X-MyHeader)})
 same => n,NoOp(Received X-MyHeader: ${X-MyHeader})
same => n,Dial(PJSIP/${EXTEN})
same => n,Hangup()

logs

Senders Log:

Connected to Asterisk 20.5.2 currently running on jaydeep-virtual-machine (pid = 8613)
<--- Received SIP request (1889 bytes) from WSS:192.168.255.134:57138 --->
INVITE sip:001@www.test445.com SIP/2.0
Via: SIP/2.0/WSS 3107r59n41oj.invalid;branch=z9hG4bK1036347
Max-Forwards: 69
To: <sip:001@www.test445.com>
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
Call-ID: 0srr4dnqpu38vhgjjhlf
CSeq: 2529 INVITE
language: dutch
LanguageCode: 334455
Contact: <sip:li4la5ll@3107r59n41oj.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.10.1
Content-Length: 1327

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 997224200938282024 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 70:39:7F:12:47:C1:74:B0:6D:3E:77:50:14:93:D6:D5:2C:11:D2:7D:8F:F0:27:E8:52:FF:5B:1E:9E:95:04:CD
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 55055 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 192.168.255.134
a=candidate:0 1 UDP 2122252543 192.168.255.134 55055 typ host
a=candidate:1 1 TCP 2105524479 192.168.255.134 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.255.134 48993 typ host
a=candidate:1 2 TCP 2105524478 192.168.255.134 9 typ host tcptype active
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:085618f85e155e4666cf9c7fb1ac9970
a=ice-ufrag:86e7319e
a=mid:0
a=msid:{c8ea2a21-89d7-4734-9f35-d16249e33351} {2f878c22-9c72-442d-8f43-6b7949a0491e}
a=rtcp:48993 IN IP4 192.168.255.134
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2865515309 cname:{5f8e9137-27e1-4b8a-801b-6f8e5e18ccb9}

<--- Transmitting SIP response (455 bytes) to WSS:192.168.255.134:57138 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 3107r59n41oj.invalid;rport=57138;received=192.168.255.134;branch=z9hG4bK1036347
Call-ID: 0srr4dnqpu38vhgjjhlf
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
To: <sip:001@www.test445.com>;tag=z9hG4bK1036347
CSeq: 2529 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1706084576/d81c03d3d5bd0ac29632357354939fd1",opaque="519f2a2b78850664",algorithm=MD5,qop="auth"
Server: ULG 1
Content-Length:  0


<--- Received SIP request (400 bytes) from WSS:192.168.255.134:57138 --->
ACK sip:001@www.test445.com SIP/2.0
Via: SIP/2.0/WSS 3107r59n41oj.invalid;branch=z9hG4bK1036347
Max-Forwards: 69
To: <sip:001@www.test445.com>;tag=z9hG4bK1036347
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
Call-ID: 0srr4dnqpu38vhgjjhlf
CSeq: 2529 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.10.1
Content-Length: 0


<--- Received SIP request (2162 bytes) from WSS:192.168.255.134:57138 --->
INVITE sip:001@www.test445.com SIP/2.0
Via: SIP/2.0/WSS 3107r59n41oj.invalid;branch=z9hG4bK8788534
Max-Forwards: 69
To: <sip:001@www.test445.com>
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
Call-ID: 0srr4dnqpu38vhgjjhlf
CSeq: 2530 INVITE
Authorization: Digest algorithm=MD5, username="001", realm="asterisk", nonce="1706084576/d81c03d3d5bd0ac29632357354939fd1", uri="sip:001@www.test445.com", response="3591d4604292a5e202fa36f2f3cbec99", opaque="519f2a2b78850664", qop=auth, cnonce="ss1rqgjvml3q", nc=00000001
language: dutch
LanguageCode: 334455
Contact: <sip:li4la5ll@3107r59n41oj.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.10.1
Content-Length: 1327

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 997224200938282024 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 70:39:7F:12:47:C1:74:B0:6D:3E:77:50:14:93:D6:D5:2C:11:D2:7D:8F:F0:27:E8:52:FF:5B:1E:9E:95:04:CD
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 55055 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 192.168.255.134
a=candidate:0 1 UDP 2122252543 192.168.255.134 55055 typ host
a=candidate:1 1 TCP 2105524479 192.168.255.134 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.255.134 48993 typ host
a=candidate:1 2 TCP 2105524478 192.168.255.134 9 typ host tcptype active
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:085618f85e155e4666cf9c7fb1ac9970
a=ice-ufrag:86e7319e
a=mid:0
a=msid:{c8ea2a21-89d7-4734-9f35-d16249e33351} {2f878c22-9c72-442d-8f43-6b7949a0491e}
a=rtcp:48993 IN IP4 192.168.255.134
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2865515309 cname:{5f8e9137-27e1-4b8a-801b-6f8e5e18ccb9}

<--- Transmitting SIP response (284 bytes) to WSS:192.168.255.134:57138 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 3107r59n41oj.invalid;rport=57138;received=192.168.255.134;branch=z9hG4bK8788534
Call-ID: 0srr4dnqpu38vhgjjhlf
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
To: <sip:001@www.test445.com>
CSeq: 2530 INVITE
Server: ULG 1
Content-Length:  0


    -- Executing [001@outside:1] NoOp("PJSIP/001-00000004", "Calling: 001)") in new stack
    -- Executing [001@outside:2] Set("PJSIP/001-00000004", "X-MyHeader=x_header_value") in new stack
    -- Executing [001@outside:5] NoOp("PJSIP/001-00000004", "Received X-MyHeader: x_header_value") in new stack
    -- Executing [001@outside:6] Dial("PJSIP/001-00000004", "PJSIP/001@outside-call") in new stack
    -- Called PJSIP/001@outside-call
<--- Transmitting SIP request (1023 bytes) to UDP:192.168.255.131:5060 --->
INVITE sip:001@192.168.255.131 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>
Contact: <sip:asterisk@192.168.255.134:5060>
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
CSeq: 29890 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.134>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   322

v=0
o=- 768705638 768705638 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
m=audio 10048 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (357 bytes) from UDP:192.168.255.131:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>
CSeq: 29890 INVITE
Server: ULG 1
Content-Length:  0


<--- Received SIP response (595 bytes) from UDP:192.168.255.131:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
CSeq: 29890 INVITE
Server: ULG 1
Contact: <sip:192.168.255.131:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Length:  0


    -- PJSIP/outside-call-00000005 is ringing
<--- Transmitting SIP response (535 bytes) to WSS:192.168.255.134:57138 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 3107r59n41oj.invalid;rport=57138;received=192.168.255.134;branch=z9hG4bK8788534
Call-ID: 0srr4dnqpu38vhgjjhlf
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
To: <sip:001@www.test445.com>;tag=cd6cc2a2-a52a-4cc0-acb3-febd22653d6a
CSeq: 2530 INVITE
Server: ULG 1
Contact: <sip:192.168.255.134:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: <sip:001@www.test445.com>
Content-Length:  0


<--- Received SIP response (1047 bytes) from UDP:192.168.255.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
CSeq: 29890 INVITE
Server: ULG 1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.255.131:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Type: application/sdp
Content-Length:   322

v=0
o=- 768705638 768705640 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
m=audio 18058 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

       > 0x7f5ed401c5c0 -- Strict RTP learning after remote address set to: 192.168.255.131:18058
<--- Transmitting SIP request (405 bytes) to UDP:192.168.255.131:5060 --->
ACK sip:192.168.255.131:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPjcf0c86fa-3a8d-4c9b-a267-cafe6ebfa489
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
CSeq: 29890 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


    -- PJSIP/outside-call-00000005 answered PJSIP/001-00000004
       > 0x7f5ed4017450 -- Strict RTP learning after remote address set to: 192.168.255.134:55055
<--- Transmitting SIP response (1533 bytes) to WSS:192.168.255.134:57138 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 3107r59n41oj.invalid;rport=57138;received=192.168.255.134;branch=z9hG4bK8788534
Call-ID: 0srr4dnqpu38vhgjjhlf
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
To: <sip:001@www.test445.com>;tag=cd6cc2a2-a52a-4cc0-acb3-febd22653d6a
CSeq: 2530 INVITE
Server: ULG 1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.255.134:8089;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
P-Asserted-Identity: <sip:001@www.test445.com>
Content-Type: application/sdp
Content-Length:   870

v=0
o=- 2455329832 2 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 10256 UDP/TLS/RTP/SAVPF 0 8 109 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 CA:68:E5:14:ED:BF:18:30:C6:43:22:35:3D:27:44:48:8B:3D:34:B7:A5:B4:DD:BA:11:9A:7F:6E:E3:20:B4:28
a=ice-ufrag:480313c86f14ba997f4ea5374a22aec5
a=ice-pwd:6e1213a018e5ed8e525143361d7c69be
a=candidate:Hc0a8ff86 1 UDP 2130706431 192.168.255.134 10256 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:635629141 cname:81f05fb8-2d88-4827-bfd0-ea1acccbeda3
a=msid:f0324380-33ca-4666-a94e-9b0d179d2e5e 98588772-a210-4017-ab64-60d1cf7e383a
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/outside-call-00000005 joined 'simple_bridge' basic-bridge <83228793-1cf7-45e7-8af6-391e0cc8a8d4>
    -- Channel PJSIP/001-00000004 joined 'simple_bridge' basic-bridge <83228793-1cf7-45e7-8af6-391e0cc8a8d4>
       > 0x7f5ed4017450 -- Strict RTP learning after ICE completion
       > 0x7f5ed4017450 -- Strict RTP learning after remote address set to: 192.168.255.134:55055
<--- Received SIP request (436 bytes) from WSS:192.168.255.134:57138 --->
ACK sip:192.168.255.134:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 3107r59n41oj.invalid;branch=z9hG4bK2650671
Max-Forwards: 69
To: <sip:001@www.test445.com>;tag=cd6cc2a2-a52a-4cc0-acb3-febd22653d6a
From: <sip:001@www.test445.com>;tag=bqbsa6h06p
Call-ID: 0srr4dnqpu38vhgjjhlf
CSeq: 2530 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.10.1
Content-Length: 0


       > 0x7f5ed401c5c0 -- Strict RTP switching to RTP target address 192.168.255.131:18058 as source
       > 0x7f5ed4017450 -- Strict RTP switching to RTP target address 192.168.255.134:55055 as source
<--- Received SIP request (437 bytes) from UDP:192.168.255.131:5060 --->
BYE sip:asterisk@192.168.255.134:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPj44264fc1-1239-454d-aa6c-a1102f60d0ec
From: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
To: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
CSeq: 1651 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Transmitting SIP response (390 bytes) to UDP:192.168.255.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPj44264fc1-1239-454d-aa6c-a1102f60d0ec
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
To: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
CSeq: 1651 BYE
Server: ULG 1
Content-Length:  0


    -- Channel PJSIP/outside-call-00000005 left 'simple_bridge' basic-bridge <83228793-1cf7-45e7-8af6-391e0cc8a8d4>
    -- Channel PJSIP/001-00000004 left 'simple_bridge' basic-bridge <83228793-1cf7-45e7-8af6-391e0cc8a8d4>
  == Spawn extension (outside, 001, 6) exited non-zero on 'PJSIP/001-00000004'
<--- Transmitting SIP request (419 bytes) to WSS:192.168.255.134:57138 --->
BYE sip:li4la5ll@192.168.255.134:57138;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 192.168.255.134:8089;rport;branch=z9hG4bKPj78527f8f-de90-4a76-9941-6c1e9d92e3c5;alias
From: <sip:001@www.test445.com>;tag=cd6cc2a2-a52a-4cc0-acb3-febd22653d6a
To: <sip:001@www.test445.com>;tag=bqbsa6h06p
Call-ID: 0srr4dnqpu38vhgjjhlf
CSeq: 29679 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Received SIP response (330 bytes) from WSS:192.168.255.134:57138 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.255.134:8089;rport;branch=z9hG4bKPj78527f8f-de90-4a76-9941-6c1e9d92e3c5;alias
To: <sip:001@www.test445.com>;tag=bqbsa6h06p
From: <sip:001@www.test445.com>;tag=cd6cc2a2-a52a-4cc0-acb3-febd22653d6a
Call-ID: 0srr4dnqpu38vhgjjhlf
CSeq: 29679 BYE
Supported: outbound
Content-Length: 0

Receivers Logs

Connected to Asterisk 20.5.0 currently running on jaydeep-virtual-machine (pid = 1299)
<--- Received SIP request (1023 bytes) from UDP:192.168.255.134:5060 --->
INVITE sip:001@192.168.255.131 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>
Contact: <sip:asterisk@192.168.255.134:5060>
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
CSeq: 29890 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.134>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   322

v=0
o=- 768705638 768705638 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
m=audio 10048 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP response (357 bytes) to UDP:192.168.255.134:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>
CSeq: 29890 INVITE
Server: ULG 1
Content-Length:  0


    -- Executing [001@from-outside-call:1] NoOp("PJSIP/outside-call-00000004", "Calling: 001)") in new stack
    -- Executing [001@from-outside-call:2] Set("PJSIP/outside-call-00000004", "X-MyHeader=") in new stack
    -- Executing [001@from-outside-call:5] NoOp("PJSIP/outside-call-00000004", "Received X-MyHeader: ") in new stack
    -- Executing [001@from-outside-call:6] Dial("PJSIP/outside-call-00000004", "PJSIP/001") in new stack
    -- Called PJSIP/001
<--- Transmitting SIP request (1721 bytes) to WSS:192.168.1.9:53176 --->
INVITE sip:5urph5vl@192.168.1.9:53176;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPjee4caf17-37e2-4c50-8a71-6112a2285e37;alias
From: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
To: <sip:5urph5vl@192.168.1.9>
Contact: <sip:asterisk@jaydeep-virtual-machine:5060;transport=ws>
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
CSeq: 26789 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@jaydeep-virtual-machine>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   960

v=0
o=- 1304552821 1304552821 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 19444 UDP/TLS/RTP/SAVPF 0 8 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 19:5D:64:29:B8:8C:A4:43:A5:77:74:FC:62:9D:96:3A:3C:80:DE:FF:D0:C4:EB:23:70:88:85:5F:3E:BE:71:66
a=ice-ufrag:7e343b22731581713b0f090b5f6119ee
a=ice-pwd:70b694f86f1d73ad1f9fcd5e4626360b
a=candidate:Hc0a80109 1 UDP 2130706431 192.168.1.9 19444 typ host
a=candidate:Hc0a8ff83 1 UDP 2130706431 192.168.255.131 19444 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1024335326 cname:9bf192bc-767a-4c70-aded-e5c6110593a3
a=msid:b43604c6-4c3b-40cf-a49f-0f00aaea9cf9 ef3f01a1-2f63-4b35-9678-bf055c80455c
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP response (473 bytes) from WSS:192.168.1.9:53176 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPjee4caf17-37e2-4c50-8a71-6112a2285e37;alias
From: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
To: <sip:5urph5vl@192.168.1.9>
CSeq: 26789 INVITE
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Content-Length: 0


<--- Received SIP response (540 bytes) from WSS:192.168.1.9:53176 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPjee4caf17-37e2-4c50-8a71-6112a2285e37;alias
From: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
To: <sip:5urph5vl@192.168.1.9>;tag=af86de3h33
CSeq: 26789 INVITE
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Contact: <sip:5urph5vl@192.0.2.174;transport=wss>
Content-Length: 0


    -- PJSIP/001-00000005 is ringing
<--- Transmitting SIP response (595 bytes) to UDP:192.168.255.134:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
CSeq: 29890 INVITE
Server: ULG 1
Contact: <sip:192.168.255.131:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Length:  0


<--- Received SIP response (1935 bytes) from WSS:192.168.1.9:53176 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPjee4caf17-37e2-4c50-8a71-6112a2285e37;alias
From: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
To: <sip:5urph5vl@192.168.1.9>;tag=af86de3h33
CSeq: 26789 INVITE
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: <sip:5urph5vl@192.0.2.174;transport=wss>
Content-Type: application/sdp
Content-Length: 1277

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 3962809889098787176 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 EF:2E:31:16:7B:C7:C8:F1:F2:8B:55:A9:0F:91:D3:F9:DD:9E:BC:19:B7:86:84:0C:2A:6B:7E:3B:49:B2:F6:39
a=group:BUNDLE audio-0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 28096 UDP/TLS/RTP/SAVPF 0 8 107 101
c=IN IP4 106.213.87.3
a=candidate:0 1 UDP 2122252543 192.168.1.9 59679 typ host
a=candidate:2 1 UDP 2122187007 192.168.255.131 53486 typ host
a=candidate:4 1 TCP 2105524479 192.168.1.9 9 typ host tcptype active
a=candidate:5 1 TCP 2105458943 192.168.255.131 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 106.213.87.3 28096 typ srflx raddr 192.168.1.9 rport 59679
a=candidate:3 1 UDP 1685987327 106.213.87.3 3573 typ srflx raddr 192.168.255.131 rport 53486
a=sendrecv
a=end-of-candidates
a=fmtp:107 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:9ccf42b4d69f1f84243ee87d8de9b6bf
a=ice-ufrag:5229619e
a=mid:audio-0
a=msid:{d2a523fc-36e2-44ce-87c4-2ffab69387d0} {205630dd-5c9c-48dc-8178-593c14b192d7}
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=setup:active
a=ssrc:679611 cname:{6c6b1ebb-eb25-4d99-91a9-878a0cef1dde}

       > 0x7f6f5c112e60 -- Strict RTP learning after remote address set to: 106.213.87.3:28096
<--- Transmitting SIP request (409 bytes) to WSS:192.168.1.9:53176 --->
ACK sip:5urph5vl@192.168.1.9:53176;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPj5529c8a8-3974-4c54-bdbc-8f54110f3870;alias
From: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
To: <sip:5urph5vl@192.168.1.9>;tag=af86de3h33
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
CSeq: 26789 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


    -- PJSIP/001-00000005 answered PJSIP/outside-call-00000004
       > 0x7f6f5c01e5e0 -- Strict RTP learning after remote address set to: 192.168.255.134:10048
<--- Transmitting SIP response (1047 bytes) to UDP:192.168.255.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPj60ef2c83-36e4-4a08-b683-0875320304d4
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
CSeq: 29890 INVITE
Server: ULG 1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.255.131:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Type: application/sdp
Content-Length:   322

v=0
o=- 768705638 768705640 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
m=audio 18058 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

    -- Channel PJSIP/001-00000005 joined 'simple_bridge' basic-bridge <6786a5a5-9c3b-403c-bffb-9bd620e748d2>
    -- Channel PJSIP/outside-call-00000004 joined 'simple_bridge' basic-bridge <6786a5a5-9c3b-403c-bffb-9bd620e748d2>
<--- Received SIP request (405 bytes) from UDP:192.168.255.134:5060 --->
ACK sip:192.168.255.131:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPjcf0c86fa-3a8d-4c9b-a267-cafe6ebfa489
From: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
To: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
CSeq: 29890 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


       > 0x7f6f5c112e60 -- Strict RTP learning after ICE completion
       > 0x7f6f5c112e60 -- Strict RTP learning after remote address set to: 192.168.1.9:59679
       > 0x7f6f5c112e60 -- Strict RTP switching to RTP target address 192.168.1.9:59679 as source
       > 0x7f6f5c01e5e0 -- Strict RTP switching to RTP target address 192.168.255.134:10048 as source
<--- Received SIP request (505 bytes) from WSS:192.168.1.9:53176 --->
BYE sip:asterisk@jaydeep-virtual-machine:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.174;branch=z9hG4bK8030033
To: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
From: "001" <sip:5urph5vl@192.168.1.9>;tag=af86de3h33
CSeq: 1 BYE
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Content-Length: 0


<--- Transmitting SIP response (333 bytes) to WSS:192.168.1.9:53176 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.174;rport=53176;received=192.168.1.9;branch=z9hG4bK8030033
Call-ID: 6845cbab-8650-4be3-a9de-c0898af174e9
From: "001" <sip:5urph5vl@192.168.1.9>;tag=af86de3h33
To: <sip:001@jaydeep-virtual-machine>;tag=603cbd2b-3508-43a6-9313-bf6c084a8b1c
CSeq: 1 BYE
Server: ULG 1
Content-Length:  0


    -- Channel PJSIP/001-00000005 left 'simple_bridge' basic-bridge <6786a5a5-9c3b-403c-bffb-9bd620e748d2>
    -- Channel PJSIP/outside-call-00000004 left 'simple_bridge' basic-bridge <6786a5a5-9c3b-403c-bffb-9bd620e748d2>
  == Spawn extension (from-outside-call, 001, 6) exited non-zero on 'PJSIP/outside-call-00000004'
<--- Transmitting SIP request (437 bytes) to UDP:192.168.255.134:5060 --->
BYE sip:asterisk@192.168.255.134:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPj44264fc1-1239-454d-aa6c-a1102f60d0ec
From: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
To: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
CSeq: 1651 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Received SIP response (390 bytes) from UDP:192.168.255.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPj44264fc1-1239-454d-aa6c-a1102f60d0ec
Call-ID: d9bd7d45-4fc4-40b3-ad10-efa955c15e38
From: <sip:001@192.168.255.131>;tag=49bff9eb-131a-4abf-8a1d-3517c2bf6eb1
To: <sip:001@192.168.255.134>;tag=cb6d4af4-7425-45e8-a894-da6348679a42
CSeq: 1651 BYE
Server: ULG 1
Content-Length:  0

Asterisk isn’t a SIP proxy. Each call leg is independent, and it’s up to you to add headers if you want using PJSIP_HEADER[1]. Asterisk doesn’t forward such information on its own.

[1] PJSIP_HEADER - Asterisk Documentation

Header Infromation is already there in the sender cli.
I want to forward the same header infromation to the other asterisk server, What modification is needed?

I told you how to do it. You use the PJSIP_HEADER dialplan function to add headers on the outgoing side. There’s even examples in what I linked.

I have manually added the header however I didn’t receive the header in the receiver
Asterisk cli

Senders Logs

<--- Received SIP request (1890 bytes) from WSS:192.168.255.134:52026 --->
INVITE sip:001@www.test445.com SIP/2.0
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;branch=z9hG4bK2328164
Max-Forwards: 69
To: <sip:001@www.test445.com>
From: <sip:001@www.test445.com>;tag=935f5pt3rt
Call-ID: 410a8t496r7p34h35p0f
CSeq: 7343 INVITE
language: dutch
LanguageCode: 334455
Contact: <sip:9rpvjn9a@0uklg33ag9hj.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.10.1
Content-Length: 1328

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 4620876043088852670 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 A7:D5:5A:DD:8E:5C:C4:88:23:EA:0C:D7:9F:84:6C:42:52:FE:23:10:F1:98:78:C4:22:33:07:15:E3:CD:8A:C7
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 52813 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 192.168.255.134
a=candidate:0 1 UDP 2122252543 192.168.255.134 52813 typ host
a=candidate:1 1 TCP 2105524479 192.168.255.134 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.255.134 55415 typ host
a=candidate:1 2 TCP 2105524478 192.168.255.134 9 typ host tcptype active
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:4bd1d7309b3f1b122951528391505edd
a=ice-ufrag:13647c07
a=mid:0
a=msid:{31a78fa7-f5f0-481f-9843-4848a9e57e45} {3517c519-1ae7-4091-a56b-7457f36e9922}
a=rtcp:55415 IN IP4 192.168.255.134
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2423474753 cname:{ae117760-29b7-4035-b09b-bb48aa66d754}

<--- Transmitting SIP response (455 bytes) to WSS:192.168.255.134:52026 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;rport=52026;received=192.168.255.134;branch=z9hG4bK2328164
Call-ID: 410a8t496r7p34h35p0f
From: <sip:001@www.test445.com>;tag=935f5pt3rt
To: <sip:001@www.test445.com>;tag=z9hG4bK2328164
CSeq: 7343 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1706095907/bc8225ffefd2dd4f3b1b38e8130ae204",opaque="5f158e2a5e3f9e94",algorithm=MD5,qop="auth"
Server: ULG 1
Content-Length:  0


<--- Received SIP request (400 bytes) from WSS:192.168.255.134:52026 --->
ACK sip:001@www.test445.com SIP/2.0
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;branch=z9hG4bK2328164
Max-Forwards: 69
To: <sip:001@www.test445.com>;tag=z9hG4bK2328164
From: <sip:001@www.test445.com>;tag=935f5pt3rt
Call-ID: 410a8t496r7p34h35p0f
CSeq: 7343 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.10.1
Content-Length: 0


<--- Received SIP request (2163 bytes) from WSS:192.168.255.134:52026 --->
INVITE sip:001@www.test445.com SIP/2.0
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;branch=z9hG4bK2662047
Max-Forwards: 69
To: <sip:001@www.test445.com>
From: <sip:001@www.test445.com>;tag=935f5pt3rt
Call-ID: 410a8t496r7p34h35p0f
CSeq: 7344 INVITE
Authorization: Digest algorithm=MD5, username="001", realm="asterisk", nonce="1706095907/bc8225ffefd2dd4f3b1b38e8130ae204", uri="sip:001@www.test445.com", response="35cea122486f5e59342817f13c64a30d", opaque="5f158e2a5e3f9e94", qop=auth, cnonce="bjlknjn446gj", nc=00000001
language: dutch
LanguageCode: 334455
Contact: <sip:9rpvjn9a@0uklg33ag9hj.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.10.1
Content-Length: 1328

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 4620876043088852670 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 A7:D5:5A:DD:8E:5C:C4:88:23:EA:0C:D7:9F:84:6C:42:52:FE:23:10:F1:98:78:C4:22:33:07:15:E3:CD:8A:C7
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 52813 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 192.168.255.134
a=candidate:0 1 UDP 2122252543 192.168.255.134 52813 typ host
a=candidate:1 1 TCP 2105524479 192.168.255.134 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.255.134 55415 typ host
a=candidate:1 2 TCP 2105524478 192.168.255.134 9 typ host tcptype active
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:4bd1d7309b3f1b122951528391505edd
a=ice-ufrag:13647c07
a=mid:0
a=msid:{31a78fa7-f5f0-481f-9843-4848a9e57e45} {3517c519-1ae7-4091-a56b-7457f36e9922}
a=rtcp:55415 IN IP4 192.168.255.134
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2423474753 cname:{ae117760-29b7-4035-b09b-bb48aa66d754}

<--- Transmitting SIP response (284 bytes) to WSS:192.168.255.134:52026 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;rport=52026;received=192.168.255.134;branch=z9hG4bK2662047
Call-ID: 410a8t496r7p34h35p0f
From: <sip:001@www.test445.com>;tag=935f5pt3rt
To: <sip:001@www.test445.com>
CSeq: 7344 INVITE
Server: ULG 1
Content-Length:  0


    -- Executing [001@outside:1] NoOp("PJSIP/001-00000004", "Calling: 001)") in new stack
    -- Executing [001@outside:2] Set("PJSIP/001-00000004", "X-MyHeader=x_myheader_value") in new stack
    -- Executing [001@outside:6] NoOp("PJSIP/001-00000004", "Received X-MyHeader: x_myheader_value") in new stack
    -- Executing [001@outside:7] Dial("PJSIP/001-00000004", "PJSIP/001@outside-call") in new stack
    -- Called PJSIP/001@outside-call
<--- Transmitting SIP request (1024 bytes) to UDP:192.168.255.131:5060 --->
INVITE sip:001@192.168.255.131 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>
Contact: <sip:asterisk@192.168.255.134:5060>
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
CSeq: 3250 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.134>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 2029888497 2029888497 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
m=audio 19052 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (356 bytes) from UDP:192.168.255.131:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>
CSeq: 3250 INVITE
Server: ULG 1
Content-Length:  0


<--- Received SIP response (594 bytes) from UDP:192.168.255.131:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
CSeq: 3250 INVITE
Server: ULG 1
Contact: <sip:192.168.255.131:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Length:  0


    -- PJSIP/outside-call-00000005 is ringing
<--- Transmitting SIP response (535 bytes) to WSS:192.168.255.134:52026 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;rport=52026;received=192.168.255.134;branch=z9hG4bK2662047
Call-ID: 410a8t496r7p34h35p0f
From: <sip:001@www.test445.com>;tag=935f5pt3rt
To: <sip:001@www.test445.com>;tag=b825a3c3-0f2d-4e65-a5bf-a88401b3b0a0
CSeq: 7344 INVITE
Server: ULG 1
Contact: <sip:192.168.255.134:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: <sip:001@www.test445.com>
Content-Length:  0


<--- Received SIP response (1048 bytes) from UDP:192.168.255.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
CSeq: 3250 INVITE
Server: ULG 1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.255.131:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 2029888497 2029888499 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
m=audio 11484 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

       > 0x7f5be000d4d0 -- Strict RTP learning after remote address set to: 192.168.255.131:11484
<--- Transmitting SIP request (404 bytes) to UDP:192.168.255.131:5060 --->
ACK sip:192.168.255.131:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPjcb8677a1-2636-4765-9ff0-c36c15920469
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
CSeq: 3250 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


    -- PJSIP/outside-call-00000005 answered PJSIP/001-00000004
       > 0x7f5be0031a50 -- Strict RTP learning after remote address set to: 192.168.255.134:52813
<--- Transmitting SIP response (1533 bytes) to WSS:192.168.255.134:52026 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;rport=52026;received=192.168.255.134;branch=z9hG4bK2662047
Call-ID: 410a8t496r7p34h35p0f
From: <sip:001@www.test445.com>;tag=935f5pt3rt
To: <sip:001@www.test445.com>;tag=b825a3c3-0f2d-4e65-a5bf-a88401b3b0a0
CSeq: 7344 INVITE
Server: ULG 1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.255.134:8089;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
P-Asserted-Identity: <sip:001@www.test445.com>
Content-Type: application/sdp
Content-Length:   870

v=0
o=- 1533834942 2 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 11606 UDP/TLS/RTP/SAVPF 0 8 109 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 38:4F:77:91:C8:72:EE:61:B7:8C:98:34:35:5E:E8:48:3D:12:8A:5D:73:A9:76:55:C5:0E:06:62:60:5A:53:18
a=ice-ufrag:35b35465247e886d7875f3d52f1e518d
a=ice-pwd:7fc134d038cdf41127208c5c552ef8f9
a=candidate:Hc0a8ff86 1 UDP 2130706431 192.168.255.134 11606 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:382018504 cname:7e8c2859-fb8a-4ffb-a410-403146536604
a=msid:de6c2ab8-aee2-4033-b196-34ea5608e3e5 410fe227-89d6-49fd-a5a9-648ee953fe41
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/outside-call-00000005 joined 'simple_bridge' basic-bridge <85ed7cc1-fb0f-4876-bdaf-6557a1be7750>
    -- Channel PJSIP/001-00000004 joined 'simple_bridge' basic-bridge <85ed7cc1-fb0f-4876-bdaf-6557a1be7750>
       > 0x7f5be0031a50 -- Strict RTP learning after ICE completion
       > 0x7f5be0031a50 -- Strict RTP learning after remote address set to: 192.168.255.134:52813
<--- Received SIP request (435 bytes) from WSS:192.168.255.134:52026 --->
ACK sip:192.168.255.134:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 0uklg33ag9hj.invalid;branch=z9hG4bK459408
Max-Forwards: 69
To: <sip:001@www.test445.com>;tag=b825a3c3-0f2d-4e65-a5bf-a88401b3b0a0
From: <sip:001@www.test445.com>;tag=935f5pt3rt
Call-ID: 410a8t496r7p34h35p0f
CSeq: 7344 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.10.1
Content-Length: 0


       > 0x7f5be000d4d0 -- Strict RTP switching to RTP target address 192.168.255.131:11484 as source
<--- Received SIP request (438 bytes) from UDP:192.168.255.131:5060 --->
BYE sip:asterisk@192.168.255.134:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPjdc0018af-7c90-452e-8cb7-183830b08bae
From: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
To: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
CSeq: 14240 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Transmitting SIP response (391 bytes) to UDP:192.168.255.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPjdc0018af-7c90-452e-8cb7-183830b08bae
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
To: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
CSeq: 14240 BYE
Server: ULG 1
Content-Length:  0


    -- Channel PJSIP/outside-call-00000005 left 'simple_bridge' basic-bridge <85ed7cc1-fb0f-4876-bdaf-6557a1be7750>
    -- Channel PJSIP/001-00000004 left 'simple_bridge' basic-bridge <85ed7cc1-fb0f-4876-bdaf-6557a1be7750>
  == Spawn extension (outside, 001, 7) exited non-zero on 'PJSIP/001-00000004'
<--- Transmitting SIP request (418 bytes) to WSS:192.168.255.134:52026 --->
BYE sip:9rpvjn9a@192.168.255.134:52026;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 192.168.255.134:8089;rport;branch=z9hG4bKPj1b148161-d9b3-44af-8e52-060544953f0f;alias
From: <sip:001@www.test445.com>;tag=b825a3c3-0f2d-4e65-a5bf-a88401b3b0a0
To: <sip:001@www.test445.com>;tag=935f5pt3rt
Call-ID: 410a8t496r7p34h35p0f
CSeq: 2294 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Received SIP response (329 bytes) from WSS:192.168.255.134:52026 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.255.134:8089;rport;branch=z9hG4bKPj1b148161-d9b3-44af-8e52-060544953f0f;alias
To: <sip:001@www.test445.com>;tag=935f5pt3rt
From: <sip:001@www.test445.com>;tag=b825a3c3-0f2d-4e65-a5bf-a88401b3b0a0
Call-ID: 410a8t496r7p34h35p0f
CSeq: 2294 BYE
Supported: outbound
Content-Length: 0

Receiver Logs

<--- Received SIP request (1024 bytes) from UDP:192.168.255.134:5060 --->
INVITE sip:001@192.168.255.131 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>
Contact: <sip:asterisk@192.168.255.134:5060>
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
CSeq: 3250 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.134>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 2029888497 2029888497 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
m=audio 19052 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP response (356 bytes) to UDP:192.168.255.134:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>
CSeq: 3250 INVITE
Server: ULG 1
Content-Length:  0


    -- Executing [001@from-outside-call:1] NoOp("PJSIP/outside-call-00000003", "Calling: 001)") in new stack
    -- Executing [001@from-outside-call:2] Set("PJSIP/outside-call-00000003", "X-MyHeader=") in new stack
    -- Executing [001@from-outside-call:7] NoOp("PJSIP/outside-call-00000003", "Received X-MyHeader:") in new stack
    -- Executing [001@from-outside-call:8] Dial("PJSIP/outside-call-00000003", "PJSIP/001") in new stack
    -- Called PJSIP/001
<--- Transmitting SIP request (1718 bytes) to WSS:192.168.1.9:52134 --->
INVITE sip:skubna0b@192.168.1.9:52134;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPj24618e29-0fb8-4328-be4d-a0b50ee7dee1;alias
From: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
To: <sip:skubna0b@192.168.1.9>
Contact: <sip:asterisk@jaydeep-virtual-machine:5060;transport=ws>
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
CSeq: 26333 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@jaydeep-virtual-machine>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   957

v=0
o=- 530785449 530785449 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 15120 UDP/TLS/RTP/SAVPF 0 8 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 78:59:C8:B6:BD:D3:33:22:5A:2D:61:FF:2A:A1:8C:65:E5:AD:F0:04:04:8A:06:17:F2:1D:96:2D:7E:62:19:C2
a=ice-ufrag:65b44d3a6dcb536b54d61e27346d9773
a=ice-pwd:412e4a836ed7e4f663cddcbd7b890daf
a=candidate:Hc0a80109 1 UDP 2130706431 192.168.1.9 15120 typ host
a=candidate:Hc0a8ff83 1 UDP 2130706431 192.168.255.131 15120 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:212435625 cname:981a0a26-194d-42c4-8c84-aee51174ec6d
a=msid:6018aa5b-0c5a-479c-8bce-d6ad86b6ab33 d5915d07-41c4-41da-8106-9c70e33068a4
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP response (473 bytes) from WSS:192.168.1.9:52134 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPj24618e29-0fb8-4328-be4d-a0b50ee7dee1;alias
From: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
To: <sip:skubna0b@192.168.1.9>
CSeq: 26333 INVITE
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Content-Length: 0


<--- Received SIP response (540 bytes) from WSS:192.168.1.9:52134 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPj24618e29-0fb8-4328-be4d-a0b50ee7dee1;alias
From: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
To: <sip:skubna0b@192.168.1.9>;tag=30pqhalh52
CSeq: 26333 INVITE
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Contact: <sip:skubna0b@192.0.2.172;transport=wss>
Content-Length: 0


    -- PJSIP/001-00000004 is ringing
<--- Transmitting SIP response (594 bytes) to UDP:192.168.255.134:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
CSeq: 3250 INVITE
Server: ULG 1
Contact: <sip:192.168.255.131:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Length:  0


<--- Received SIP response (1940 bytes) from WSS:192.168.1.9:52134 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPj24618e29-0fb8-4328-be4d-a0b50ee7dee1;alias
From: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
To: <sip:skubna0b@192.168.1.9>;tag=30pqhalh52
CSeq: 26333 INVITE
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: <sip:skubna0b@192.0.2.172;transport=wss>
Content-Type: application/sdp
Content-Length: 1282

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 2854123816446200083 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 40:4D:27:13:82:E9:BB:53:ED:8D:C7:97:32:08:5D:21:00:DF:FD:74:11:E3:C4:67:0D:5F:46:C5:C3:AA:97:AC
a=group:BUNDLE audio-0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 16307 UDP/TLS/RTP/SAVPF 0 8 107 101
c=IN IP4 106.213.87.3
a=candidate:0 1 UDP 2122252543 192.168.1.9 55341 typ host
a=candidate:2 1 UDP 2122187007 192.168.255.131 51113 typ host
a=candidate:4 1 TCP 2105524479 192.168.1.9 9 typ host tcptype active
a=candidate:5 1 TCP 2105458943 192.168.255.131 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 106.213.87.3 16307 typ srflx raddr 192.168.1.9 rport 55341
a=candidate:3 1 UDP 1685987327 106.213.87.3 26982 typ srflx raddr 192.168.255.131 rport 51113
a=sendrecv
a=end-of-candidates
a=fmtp:107 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:5dd68b97c02c6fa4c0315e3757bd3911
a=ice-ufrag:0d44b5fa
a=mid:audio-0
a=msid:{77a6dbee-650b-48f9-8f02-01bef321a57c} {0672c265-5069-4cdf-b587-9551298fee50}
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=setup:active
a=ssrc:2532538201 cname:{eb5542f4-ffc0-46f8-bf1d-537687dfed8e}

       > 0x7f510402ff60 -- Strict RTP learning after remote address set to: 106.213.87.3:16307
<--- Transmitting SIP request (409 bytes) to WSS:192.168.1.9:52134 --->
ACK sip:skubna0b@192.168.1.9:52134;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.168.1.9:8089;rport;branch=z9hG4bKPjd9267b98-d1d3-4ed3-8c19-a97c6f99e2c9;alias
From: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
To: <sip:skubna0b@192.168.1.9>;tag=30pqhalh52
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
CSeq: 26333 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


    -- PJSIP/001-00000004 answered PJSIP/outside-call-00000003
       > 0x7f510401cdc0 -- Strict RTP learning after remote address set to: 192.168.255.134:19052
<--- Transmitting SIP response (1048 bytes) to UDP:192.168.255.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.134:5060;rport=5060;received=192.168.255.134;branch=z9hG4bKPjbb796e3b-2e3d-4016-9d9a-4e9b6365fa90
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
CSeq: 3250 INVITE
Server: ULG 1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.255.131:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
P-Asserted-Identity: <sip:001@192.168.255.131>
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 2029888497 2029888499 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
m=audio 11484 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

    -- Channel PJSIP/001-00000004 joined 'simple_bridge' basic-bridge <5c011e34-cfb3-447a-97bb-bf254ec721e5>
    -- Channel PJSIP/outside-call-00000003 joined 'simple_bridge' basic-bridge <5c011e34-cfb3-447a-97bb-bf254ec721e5>
       > 0x7f510402ff60 -- Strict RTP learning after ICE completion
<--- Received SIP request (404 bytes) from UDP:192.168.255.134:5060 --->
ACK sip:192.168.255.131:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.134:5060;rport;branch=z9hG4bKPjcb8677a1-2636-4765-9ff0-c36c15920469
From: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
To: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
CSeq: 3250 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


       > 0x7f510402ff60 -- Strict RTP learning after remote address set to: 192.168.1.9:55341
       > 0x7f510402ff60 -- Strict RTP switching to RTP target address 192.168.1.9:55341 as source
<--- Received SIP request (505 bytes) from WSS:192.168.1.9:52134 --->
BYE sip:asterisk@jaydeep-virtual-machine:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.172;branch=z9hG4bK2765745
To: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
From: "001" <sip:skubna0b@192.168.1.9>;tag=30pqhalh52
CSeq: 1 BYE
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Content-Length: 0


<--- Transmitting SIP response (333 bytes) to WSS:192.168.1.9:52134 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.172;rport=52134;received=192.168.1.9;branch=z9hG4bK2765745
Call-ID: dc0b1a57-9be8-42c4-bcd7-779cddd52cbe
From: "001" <sip:skubna0b@192.168.1.9>;tag=30pqhalh52
To: <sip:001@jaydeep-virtual-machine>;tag=848cdbce-b664-41f5-bd08-3f7d851e2b6a
CSeq: 1 BYE
Server: ULG 1
Content-Length:  0


    -- Channel PJSIP/001-00000004 left 'simple_bridge' basic-bridge <5c011e34-cfb3-447a-97bb-bf254ec721e5>
    -- Channel PJSIP/outside-call-00000003 left 'simple_bridge' basic-bridge <5c011e34-cfb3-447a-97bb-bf254ec721e5>
  == Spawn extension (from-outside-call, 001, 8) exited non-zero on 'PJSIP/outside-call-00000003'
<--- Transmitting SIP request (438 bytes) to UDP:192.168.255.134:5060 --->
BYE sip:asterisk@192.168.255.134:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPjdc0018af-7c90-452e-8cb7-183830b08bae
From: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
To: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
CSeq: 14240 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Received SIP response (391 bytes) from UDP:192.168.255.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPjdc0018af-7c90-452e-8cb7-183830b08bae
Call-ID: 4179050c-78bb-4b00-af99-41742ae80a9c
From: <sip:001@192.168.255.131>;tag=fdb0ed80-433e-4401-9365-8f4683e38521
To: <sip:001@192.168.255.134>;tag=b41a10d5-8938-4969-b890-7b121ec2c2eb
CSeq: 14240 BYE
Server: ULG 1
Content-Length:  0


<--- Received SIP request (822 bytes) from WSS:192.168.1.9:52134 --->
REGISTER sip:www.test445.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.172;branch=z9hG4bK1656233
To: <sip:001@www.test445.com>
From: "001" <sip:001@www.test445.com>;tag=km7b16e03g
CSeq: 4 REGISTER
Call-ID: mnqfbj660t0hg2q21mpv
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="001", realm="asterisk", nonce="1706095694/e9c88a389a73dcabc47169040a5c827b", uri="sip:www.test445.com", response="e639e653866bf778842ce725c131b245", opaque="1561cc5d2c7bf431", qop=auth, cnonce="hav5i5kfabqe", nc=00000001
Contact: <sip:skubna0b@192.0.2.172;transport=wss>;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Content-Length: 0


<--- Transmitting SIP response (458 bytes) to WSS:192.168.1.9:52134 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.172;rport=52134;received=192.168.1.9;branch=z9hG4bK1656233
Call-ID: mnqfbj660t0hg2q21mpv
From: "001" <sip:001@www.test445.com>;tag=km7b16e03g
To: <sip:001@www.test445.com>;tag=z9hG4bK1656233
CSeq: 4 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1706095918/44335ec00a3cf411e87cb928b1201088",opaque="000bd54639c879ff",stale=true,algorithm=MD5,qop="auth"
Server: ULG 1
Content-Length:  0


<--- Received SIP request (822 bytes) from WSS:192.168.1.9:52134 --->
REGISTER sip:www.test445.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.172;branch=z9hG4bK8876673
To: <sip:001@www.test445.com>
From: "001" <sip:001@www.test445.com>;tag=km7b16e03g
CSeq: 5 REGISTER
Call-ID: mnqfbj660t0hg2q21mpv
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="001", realm="asterisk", nonce="1706095918/44335ec00a3cf411e87cb928b1201088", uri="sip:www.test445.com", response="04866a52810fca069c46c313eabf77af", opaque="000bd54639c879ff", qop=auth, cnonce="vlb7b6eegdks", nc=00000001
Contact: <sip:skubna0b@192.0.2.172;transport=wss>;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:109.0) Gecko/20100101 Firefox/116.0
Content-Length: 0


<--- Transmitting SIP response (390 bytes) to WSS:192.168.1.9:52134 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.172;rport=52134;received=192.168.1.9;branch=z9hG4bK8876673
Call-ID: mnqfbj660t0hg2q21mpv
From: "001" <sip:001@www.test445.com>;tag=km7b16e03g
To: <sip:001@www.test445.com>;tag=z9hG4bK8876673
CSeq: 5 REGISTER
Date: Wed, 24 Jan 2024 11:31:58 GMT
Contact: <sip:skubna0b@192.0.2.172;transport=ws>;expires=299
Server: ULG 1
Content-Length:  0

You didn’t add it to the outgoing side. From the documentation I linked:

If you call PJSIP_HEADER in a normal dialplan context you’ll be operating on the caller’s (incoming) channel which may not be what you want. To operate on the callee’s (outgoing) channel call PJSIP_HEADER in a pre-dial handler.

And under it is an example of using a pre-dial handler.

@jcolp thanks

[handler]
exten => addheader,1,Set(PJSIP_HEADER(add,X-MyHeader)=myvalue)
exten => addheader,2,Set(PJSIP_HEADER(add,X-MyHeader2)=myvalue2)

[somecontext]
exten => 1,1,Dial(PJSIP/${EXTEN},,b(handler^addheader^1))

the above code from the documentation helped me to forward the header infromation

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