<--- Received SIP request (1143 bytes) from UDP:192.168.0.21:5070 --->
INVITE sip:1104@192.168.0.21;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5070;branch=z9hG4bKf53de24dd0dc8af9;rport
Contact: <sip:1101-0x5647aca05110@192.168.0.21:5070>
Max-Forwards: 70
To: <sip:1104@192.168.0.21;transport=udp>
From: "CIRCLE" <sip:1101@192.168.0.21:5060>;tag=e52b5e34d07e7221
Call-ID: 9e16ba342b24b6d5
CSeq: 10092 INVITE
User-Agent: baresip v1.0.0 (x86_64/linux)
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER
Supported: gruu
Content-Type: application/sdp
Content-Length: 602

v=0
o=- 3836576958 341476008 IN IP4 192.168.0.21
s=-
c=IN IP4 192.168.0.21
t=0 0
a=tool:baresip 1.0.0
m=audio 42526 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=label:1
a=rtcp-rsize
a=ssrc:4072846663 cname:sip:1101@192.168.0.21:5060
a=minptime:20
a=ptime:20
m=video 2318 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=0;profile-level-id=42e01f
a=sendrecv
a=label:2
a=rtcp-rsize
a=ssrc:2565444252 cname:sip:1101@192.168.0.21:5060
a=framerate:25.00
a=rtcp-fb:* nack pli
a=content:main

<--- Transmitting SIP response (488 bytes) to UDP:192.168.0.21:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.21:5070;rport=5070;received=192.168.0.21;branch=z9hG4bKf53de24dd0dc8af9
Call-ID: 9e16ba342b24b6d5
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=e52b5e34d07e7221
To: <sip:1104@192.168.0.21>;tag=z9hG4bKf53de24dd0dc8af9
CSeq: 10092 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1616760764/5441545f56b676ef896e283d96541159",opaque="098a0faf57580f4d",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.2.2
Content-Length:  0


<--- Received SIP request (371 bytes) from UDP:192.168.0.21:5070 --->
ACK sip:1104@192.168.0.21;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5070;branch=z9hG4bKf53de24dd0dc8af9;rport
Max-Forwards: 70
To: <sip:1104@192.168.0.21>;tag=z9hG4bKf53de24dd0dc8af9
From: "CIRCLE" <sip:1101@192.168.0.21:5060>;tag=e52b5e34d07e7221
Call-ID: 9e16ba342b24b6d5
CSeq: 10092 ACK
User-Agent: baresip v1.0.0 (x86_64/linux)
Content-Length: 0


<--- Received SIP request (1418 bytes) from UDP:192.168.0.21:5070 --->
INVITE sip:1104@192.168.0.21;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5070;branch=z9hG4bKb4b7a6e6c9b58f5e;rport
Contact: <sip:1101-0x5647aca05110@192.168.0.21:5070>
Max-Forwards: 70
Authorization: Digest username="1101", realm="asterisk", nonce="1616760764/5441545f56b676ef896e283d96541159", uri="sip:1104@192.168.0.21;transport=udp", response="13d6bfbebd0c454da180b6a15babdf66", opaque="098a0faf57580f4d", cnonce="a128508cd6898acb", qop=auth, nc=00000001
To: <sip:1104@192.168.0.21;transport=udp>
From: "CIRCLE" <sip:1101@192.168.0.21:5060>;tag=e52b5e34d07e7221
Call-ID: 9e16ba342b24b6d5
CSeq: 10093 INVITE
User-Agent: baresip v1.0.0 (x86_64/linux)
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER
Supported: gruu
Content-Type: application/sdp
Content-Length: 602

v=0
o=- 3836576958 341476008 IN IP4 192.168.0.21
s=-
c=IN IP4 192.168.0.21
t=0 0
a=tool:baresip 1.0.0
m=audio 42526 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=label:1
a=rtcp-rsize
a=ssrc:4072846663 cname:sip:1101@192.168.0.21:5060
a=minptime:20
a=ptime:20
m=video 2318 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=0;profile-level-id=42e01f
a=sendrecv
a=label:2
a=rtcp-rsize
a=ssrc:2565444252 cname:sip:1101@192.168.0.21:5060
a=framerate:25.00
a=rtcp-fb:* nack pli
a=content:main

<--- Transmitting SIP response (308 bytes) to UDP:192.168.0.21:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.21:5070;rport=5070;received=192.168.0.21;branch=z9hG4bKb4b7a6e6c9b58f5e
Call-ID: 9e16ba342b24b6d5
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=e52b5e34d07e7221
To: <sip:1104@192.168.0.21>
CSeq: 10093 INVITE
Server: Asterisk PBX 18.2.2
Content-Length:  0


    -- Executing [1104@default:1] Dial("PJSIP/1101-00000012", "PJSIP/1104,10") in new stack
<--- Transmitting SIP request (636 bytes) to UDP:192.168.0.10:10242 --->
INVITE sip:1104@192.168.0.10:10242 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPjf2f98a52-6963-4482-916f-ef9884f84aee
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
To: <sip:1104@192.168.0.10>
Contact: <sip:asterisk@192.168.0.21:5060>
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 18625 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.2.2
Content-Length:  0


    -- Called PJSIP/1104
<--- Received SIP response (467 bytes) from UDP:192.168.0.10:10242 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.21:5060;rport=5060;branch=z9hG4bKPjf2f98a52-6963-4482-916f-ef9884f84aee
Contact: <sip:1104@192.168.0.10:10242>;+sip.instance="<urn:uuid:8b0afe03-3d8c-4484-a466-f4ef50e82332>"
To: <sip:1104@192.168.0.10>;tag=7c2c4c63
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 18625 INVITE
User-Agent: PortSIP UC Client
Content-Length: 0


    -- PJSIP/1104-00000013 is ringing
<--- Transmitting SIP response (495 bytes) to UDP:192.168.0.21:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.21:5070;rport=5070;received=192.168.0.21;branch=z9hG4bKb4b7a6e6c9b58f5e
Call-ID: 9e16ba342b24b6d5
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=e52b5e34d07e7221
To: <sip:1104@192.168.0.21>;tag=e86f6989-aa9e-4357-b508-4eeffe9a7e00
CSeq: 10093 INVITE
Server: Asterisk PBX 18.2.2
Contact: <sip:192.168.0.21:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (1026 bytes) from UDP:192.168.0.10:10242 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5060;rport=5060;branch=z9hG4bKPjf2f98a52-6963-4482-916f-ef9884f84aee
Contact: <sip:1104@192.168.0.10:10242>;+sip.instance="<urn:uuid:8b0afe03-3d8c-4484-a466-f4ef50e82332>"
To: <sip:1104@192.168.0.10>;tag=7c2c4c63
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 18625 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Allow-Events: hold, talk, conference, dialog
Content-Length: 323

v=0
o=- 1616760764 1 IN IP4 192.168.0.10
s=ps
c=IN IP4 192.168.0.10
t=0 0
m=audio 10132 RTP/AVP 18 8 0 105 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:105 opus/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[Mar 26 13:12:46] WARNING[11168]: res_pjsip_session.c:1100 handle_negotiated_sdp: PJSIP/1104-00000013: Local SDP contains 2 media streams while we expected it to contain 1
<--- Transmitting SIP request (680 bytes) to UDP:192.168.0.10:10242 --->
ACK sip:1104@192.168.0.10:10242 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPjeca143c8-3378-49fd-b88a-26a6a4ac1d12
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
To: <sip:1104@192.168.0.10>;tag=7c2c4c63
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 18625 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.2.2
Content-Type: application/sdp
Content-Length:   250

v=0
o=- 1616760764 3 IN IP4 192.168.0.21
s=Asterisk
c=IN IP4 192.168.0.21
t=0 0
m=audio 13496 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

    -- PJSIP/1104-00000013 answered PJSIP/1101-00000012
       > 0x7f246806ba50 -- Strict RTP learning after remote address set to: 192.168.0.21:42526
       > 0x7f2468045330 -- Strict RTP learning after remote address set to: 192.168.0.21:2318
<--- Transmitting SIP response (950 bytes) to UDP:192.168.0.21:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5070;rport=5070;received=192.168.0.21;branch=z9hG4bKb4b7a6e6c9b58f5e
Call-ID: 9e16ba342b24b6d5
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=e52b5e34d07e7221
To: <sip:1104@192.168.0.21>;tag=e86f6989-aa9e-4357-b508-4eeffe9a7e00
CSeq: 10093 INVITE
Server: Asterisk PBX 18.2.2
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.0.21:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   378

v=0
o=- 3836576958 341476010 IN IP4 192.168.0.21
s=Asterisk
c=IN IP4 192.168.0.21
t=0 0
m=audio 18060 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 14256 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=0;profile-level-id=42E01F
a=sendrecv

    -- Channel PJSIP/1104-00000013 joined 'simple_bridge' basic-bridge <d6ed66bd-99ea-4762-b139-a19ceda2e0af>
    -- Channel PJSIP/1101-00000012 joined 'simple_bridge' basic-bridge <d6ed66bd-99ea-4762-b139-a19ceda2e0af>
       > 0x7f246806ba50 -- Strict RTP switching to RTP target address 192.168.0.21:42526 as source
<--- Received SIP request (631 bytes) from UDP:192.168.0.21:5070 --->
ACK sip:192.168.0.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5070;branch=z9hG4bK57ce622b6158b53d;rport
Max-Forwards: 70
Authorization: Digest username="1101", realm="asterisk", nonce="1616760764/5441545f56b676ef896e283d96541159", uri="sip:192.168.0.21:5060", response="c81271992d026d9f8b52c6e4b5f9feb9", opaque="098a0faf57580f4d", cnonce="411504516bc8c04d", qop=auth, nc=00000002
To: <sip:1104@192.168.0.21>;tag=e86f6989-aa9e-4357-b508-4eeffe9a7e00
From: "CIRCLE" <sip:1101@192.168.0.21:5060>;tag=e52b5e34d07e7221
Call-ID: 9e16ba342b24b6d5
CSeq: 10093 ACK
User-Agent: baresip v1.0.0 (x86_64/linux)
Content-Length: 0


       > 0x7f2468045330 -- Strict RTP switching to RTP target address 192.168.0.21:2318 as source
<--- Received SIP request (1199 bytes) from UDP:192.168.0.10:10242 --->
INVITE sip:asterisk@192.168.0.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:10242;branch=z9hG4bK-524287-1---f7f9546a2b2f3934;rport
Max-Forwards: 70
Contact: <sip:1104@192.168.0.10:10242>;+sip.instance="<urn:uuid:8b0afe03-3d8c-4484-a466-f4ef50e82332>"
To: "CIRCLE"<sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
From: <sip:1104@192.168.0.10>;tag=7c2c4c63
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Allow-Events: hold, talk, conference, dialog
Content-Length: 466

v=0
o=- 1616760764 2 IN IP4 192.168.0.10
s=ps
c=IN IP4 192.168.0.10
t=0 0
m=audio 10132 RTP/AVP 18 8 0 105 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:105 opus/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 15126 RTP/AVP 125
a=rtpmap:125 H264/90000
a=fmtp:125 profile-level-id=42801E;packetization-mode=0
a=sendrecv
a=rtcp-fb:* nack pli

       > 0x7f24680700b0 -- Strict RTP learning after remote address set to: 192.168.0.10:10132
       > 0x7f2468050250 -- Strict RTP learning after remote address set to: 192.168.0.10:15126
<--- Transmitting SIP response (974 bytes) to UDP:192.168.0.10:10242 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:10242;rport=10242;received=192.168.0.10;branch=z9hG4bK-524287-1---f7f9546a2b2f3934
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
From: <sip:1104@192.168.0.10>;tag=7c2c4c63
To: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
CSeq: 2 INVITE
Contact: <sip:asterisk@192.168.0.21:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 18.2.2
Content-Type: application/sdp
Content-Length:   371

v=0
o=- 1616760764 4 IN IP4 192.168.0.21
s=Asterisk
c=IN IP4 192.168.0.21
t=0 0
m=audio 17042 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
m=video 16388 RTP/AVP 125
a=rtpmap:125 H264/90000
a=fmtp:125 packetization-mode=0;profile-level-id=42801E
a=sendrecv

<--- Received SIP request (486 bytes) from UDP:192.168.0.10:10242 --->
ACK sip:asterisk@192.168.0.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:10242;branch=z9hG4bK-524287-1---ec44e5617146d255;rport
Max-Forwards: 70
Contact: <sip:1104@192.168.0.10:10242>;+sip.instance="<urn:uuid:8b0afe03-3d8c-4484-a466-f4ef50e82332>"
To: "CIRCLE"<sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
From: <sip:1104@192.168.0.10>;tag=7c2c4c63
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 2 ACK
User-Agent: PortSIP UC Client
Content-Length: 0


       > 0x7f24680700b0 -- Strict RTP switching to RTP target address 192.168.0.10:10132 as source
[Mar 26 13:12:49] WARNING[11195][C-0000000a]: translate.c:488 ast_translator_build_path: No translator path: (ending codec is not valid)
[Mar 26 13:12:49] WARNING[11195][C-0000000a]: translate.c:488 ast_translator_build_path: No translator path: (ending codec is not valid)
[Mar 26 13:12:49] WARNING[11189][C-0000000a]: channel.c:5674 set_format: Unable to find a codec translation path: (g729) -> (ulaw|h264)
    -- Channel PJSIP/1101-00000012 left 'simple_bridge' basic-bridge <d6ed66bd-99ea-4762-b139-a19ceda2e0af>
  == Spawn extension (default, 1104, 1) exited non-zero on 'PJSIP/1101-00000012'
    -- Channel PJSIP/1104-00000013 left 'simple_bridge' basic-bridge <d6ed66bd-99ea-4762-b139-a19ceda2e0af>
[Mar 26 13:12:49] WARNING[11195][C-0000000a]: channel.c:5674 set_format: Unable to find a codec translation path: (h264|g729) -> (ulaw)
<--- Transmitting SIP request (420 bytes) to UDP:192.168.0.10:10242 --->
BYE sip:1104@192.168.0.10:10242 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPj83eb4061-a939-4f3c-8427-40b45d77ad6a
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
To: <sip:1104@192.168.0.10>;tag=7c2c4c63
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 18626 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.2.2
Content-Length:  0


<--- Transmitting SIP request (422 bytes) to UDP:192.168.0.21:5070 --->
BYE sip:1101-0x5647aca05110@192.168.0.21:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;rport;branch=z9hG4bKPj38ac6fe7-c26f-4c7d-b017-3ca1ea5f678b
From: <sip:1104@192.168.0.21>;tag=e86f6989-aa9e-4357-b508-4eeffe9a7e00
To: "CIRCLE" <sip:1101@192.168.0.21>;tag=e52b5e34d07e7221
Call-ID: 9e16ba342b24b6d5
CSeq: 27076 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.2.2
Content-Length:  0


<--- Received SIP response (373 bytes) from UDP:192.168.0.21:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bKPj38ac6fe7-c26f-4c7d-b017-3ca1ea5f678b;rport=5060;received=192.168.0.21
From: <sip:1104@192.168.0.21>;tag=e86f6989-aa9e-4357-b508-4eeffe9a7e00
To: "CIRCLE" <sip:1101@192.168.0.21>;tag=e52b5e34d07e7221
Call-ID: 9e16ba342b24b6d5
CSeq: 27076 BYE
Server: baresip v1.0.0 (x86_64/linux)
Content-Length: 0


<--- Received SIP response (459 bytes) from UDP:192.168.0.10:10242 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.21:5060;rport=5060;branch=z9hG4bKPj83eb4061-a939-4f3c-8427-40b45d77ad6a
Contact: <sip:1104@192.168.0.10:10242>;+sip.instance="<urn:uuid:8b0afe03-3d8c-4484-a466-f4ef50e82332>"
To: <sip:1104@192.168.0.10>;tag=7c2c4c63
From: "CIRCLE" <sip:1101@192.168.0.21>;tag=9c6601c5-b633-4ab0-a9ac-2b972ca3ad4b
Call-ID: 2e9551a2-4457-49df-aef3-9230532cb823
CSeq: 18626 BYE
User-Agent: PortSIP UC Client
Content-Length: 0