CLI of sip phone (37301)<β>Asterisk <β> PSTN Gateway <β> Analog Phone(37400) when allow=all
<β Received SIP request (1083 bytes) from UDP:192.168.133.9:5060 β>
INVITE sip:5537400@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.deKa0SJHD;rport
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111
CSeq: 20 INVITE
Call-ID: qAJzmzMXW4
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=βurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6β
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
v=0
o=37200 860 3348 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<β Transmitting SIP response (477 bytes) to UDP:192.168.133.9:5060 β>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.deKa0SJHD
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=z9hG4bK.deKa0SJHD
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=βasteriskβ,nonce=β1582803850/c68d9c4942ea27702747fb2ae45a5fd2β,opaque=β6fdadc21648d812eβ,algorithm=md5,qop=βauthβ
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (384 bytes) from UDP:192.168.133.9:5060 β>
ACK sip:5537400@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.deKa0SJHD;rport
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=z9hG4bK.deKa0SJHD
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=βurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6β
Max-Forwards: 70
CSeq: 20 ACK
<β Received SIP request (1368 bytes) from UDP:192.168.133.9:5060 β>
INVITE sip:5537400@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.64jPZgZQ4;rport
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111
CSeq: 21 INVITE
Call-ID: qAJzmzMXW4
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=βurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6β
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Authorization: Digest realm=βasteriskβ, nonce=β1582803850/c68d9c4942ea27702747fb2ae45a5fd2β, algorithm=md5, opaque=β6fdadc21648d812eβ, username=β37200β, uri="sip:5537400@192.168.133.111", response=β2f8bf71cf466f61770b1b7e676b9f74cβ, cnonce=βfnYHwNjtNYFfx2iSβ, nc=00000001, qop=auth
v=0
o=37200 860 3348 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
== Setting global variable βSIPDOMAINβ to β192.168.133.111β
<β Transmitting SIP response (303 bytes) to UDP:192.168.133.9:5060 β>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
-- Executing [5537400@phones:1] NoOp("PJSIP/37200-0000005c", "") in new stack
-- Executing [5537400@phones:2] Dial("PJSIP/37200-0000005c", "PJSIP/37400@gateway,,25") in new stack
-- Called PJSIP/37400@gateway
<β Transmitting SIP request (648 bytes) to UDP:192.168.133.110:5060 β>
INVITE sip:37400@192.168.133.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12306 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP response (409 bytes) from UDP:192.168.133.110:5060 β>
SIP/2.0 100 Trying
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Content-Length: 0
CSeq: 12306 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
Quintum: 0b03313336
<β Received SIP response (794 bytes) from UDP:192.168.133.110:5060 β>
SIP/2.0 183 Session Progress
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Contact: sip:37400@192.168.133.110
Content-Length: 282
Content-Type: application/sdp
CSeq: 12306 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
Quintum: 070e9777887800313106001e03808081
v=0
o=Quintum 31 31 IN IP4 192.168.133.110
s=VoipCall
c=IN IP4 192.168.133.110
t=0 0
m=audio 10308 RTP/AVP 0 8 0 101
c=IN IP4 192.168.133.110
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
-- PJSIP/gateway-0000005d is making progress passing it to PJSIP/37200-0000005c
-- PJSIP/gateway-0000005d is making progress passing it to PJSIP/37200-0000005c
<β Transmitting SIP response (790 bytes) to UDP:192.168.133.9:5060 β>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Contact: sip:192.168.133.111:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 254
v=0
o=- 860 3350 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Transmitting SIP response (790 bytes) to UDP:192.168.133.9:5060 β>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:192.168.133.111:5060
Content-Type: application/sdp
Content-Length: 254
v=0
o=- 860 3350 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803821/abda582c4f90086ba17566405b3c9887β,opaque=β455484dc74b416e2β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=β47a5d9673ce59d2e9856077099ab93f3β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 639 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0425
<β Transmitting SIP response (548 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0425
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0425
CSeq: 639 REGISTER
WWW-Authenticate: Digest realm=βasteriskβ,nonce=β1582803851/0a8b6a3ea35c8b932bdd2e7d7008585dβ,opaque=β6191bf2630478c73β,stale=true,algorithm=md5,qop=βauthβ
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803851/0a8b6a3ea35c8b932bdd2e7d7008585dβ,opaque=β6191bf2630478c73β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=βccf9e19eb04dcd85904ae482cb684567β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 640 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0427
<β Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0427
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0427
CSeq: 640 REGISTER
Date: Thu, 27 Feb 2020 11:44:11 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP response (737 bytes) from UDP:192.168.133.110:5060 β>
SIP/2.0 200 OK
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Contact: sip:37400@192.168.133.110
Content-Length: 282
Content-Type: application/sdp
CSeq: 12306 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
v=0
o=Quintum 32 32 IN IP4 192.168.133.110
s=VoipCall
c=IN IP4 192.168.133.110
t=0 0
m=audio 10308 RTP/AVP 0 8 0 101
c=IN IP4 192.168.133.110
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
-- PJSIP/gateway-0000005d answered PJSIP/37200-0000005c
<β Transmitting SIP request (674 bytes) to UDP:192.168.133.110:5060 β>
ACK sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPja4bac9e7-55dc-4548-a96c-a73eae2dfd9b
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12306 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 227
v=0
o=- 31 33 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 17872 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Transmitting SIP response (824 bytes) to UDP:192.168.133.9:5060 β>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:192.168.133.111:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 254
v=0
o=- 860 3350 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
-- Channel PJSIP/gateway-0000005d joined 'simple_bridge' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>
-- Channel PJSIP/37200-0000005c joined 'simple_bridge' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>
<β Transmitting SIP request (917 bytes) to UDP:192.168.133.110:5060 β>
INVITE sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 224
v=0
o=- 31 34 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Received SIP request (619 bytes) from UDP:192.168.133.9:5060 β>
ACK sip:192.168.133.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;rport;branch=z9hG4bK.ASt31QCn7
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 ACK
Call-ID: qAJzmzMXW4
Max-Forwards: 70
Authorization: Digest realm=βasteriskβ, nonce=β1582803850/c68d9c4942ea27702747fb2ae45a5fd2β, algorithm=md5, opaque=β6fdadc21648d812eβ, username=β37200β, uri="sip:5537400@192.168.133.111", response=β2f8bf71cf466f61770b1b7e676b9f74cβ, cnonce=βfnYHwNjtNYFfx2iSβ, nc=00000001, qop=auth
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
<β Transmitting SIP request (900 bytes) to UDP:192.168.133.9:5060 β>
INVITE sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj01db1d75-244e-44a8-a7f1-7306378fd8d3
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Contact: sip:192.168.133.111:5060
Call-ID: qAJzmzMXW4
CSeq: 31911 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 230
v=0
o=- 860 3351 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.110
t=0 0
m=audio 10308 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Received SIP response (286 bytes) from UDP:192.168.133.9:5060 β>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj01db1d75-244e-44a8-a7f1-7306378fd8d3
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31911 INVITE
<β Received SIP response (764 bytes) from UDP:192.168.133.9:5060 β>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj01db1d75-244e-44a8-a7f1-7306378fd8d3
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31911 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=βurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6β
Content-Type: application/sdp
Content-Length: 146
v=0
o=37200 860 3350 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 100
a=rtpmap:100 telephone-event/8000
<β Transmitting SIP request (399 bytes) to UDP:192.168.133.9:5060 β>
ACK sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj5444a648-3ecf-43df-8369-b9ab1b4c6386
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31911 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (402 bytes) from UDP:192.168.133.110:5060 β>
BYE sip:asterisk@192.168.133.111:5060 SIP/2.0
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Content-Length: 0
CSeq: 12307 BYE
From: sip:37400@192.168.133.110;tag=c0a8856e-35
Max-Forwards: 70
To: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0428
<β Transmitting SIP response (376 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0428
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
From: sip:37400@192.168.133.110;tag=c0a8856e-35
To: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
CSeq: 12307 BYE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
-- Channel PJSIP/gateway-0000005d left 'native_rtp' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>
-- Channel PJSIP/37200-0000005c left 'native_rtp' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>
<β Transmitting SIP request (900 bytes) to UDP:192.168.133.9:5060 β>
INVITE sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjffb03d42-0e07-41c3-b1be-6e5ba7109d52
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Contact: sip:192.168.133.111:5060
Call-ID: qAJzmzMXW4
CSeq: 31912 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 230
v=0
o=- 860 3352 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
== Spawn extension (phones, 5537400, 2) exited non-zero on βPJSIP/37200-0000005cβ
<β Received SIP response (286 bytes) from UDP:192.168.133.9:5060 β>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjffb03d42-0e07-41c3-b1be-6e5ba7109d52
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31912 INVITE
<β Received SIP response (764 bytes) from UDP:192.168.133.9:5060 β>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjffb03d42-0e07-41c3-b1be-6e5ba7109d52
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31912 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=βurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6β
Content-Type: application/sdp
Content-Length: 146
v=0
o=37200 860 3352 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 100
a=rtpmap:100 telephone-event/8000
<β Transmitting SIP request (399 bytes) to UDP:192.168.133.9:5060 β>
ACK sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj57b41596-43d1-4f2b-8ea1-64615c1b9643
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31912 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Transmitting SIP request (399 bytes) to UDP:192.168.133.9:5060 β>
BYE sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj7f970460-3a57-4fdf-ace6-c9914416ea1e
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31913 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP response (358 bytes) from UDP:192.168.133.9:5060 β>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj7f970460-3a57-4fdf-ace6-c9914416ea1e
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31913 BYE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound
<β Transmitting SIP request (917 bytes) to UDP:192.168.133.110:5060 β>
INVITE sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 224
v=0
o=- 31 34 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Transmitting SIP request (917 bytes) to UDP:192.168.133.110:5060 β>
INVITE sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 224
v=0
o=- 31 34 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<β Received SIP response (360 bytes) from UDP:192.168.133.110:5060 β>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Content-Length: 0
CSeq: 12307 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
<β Transmitting SIP request (413 bytes) to UDP:192.168.133.110:5060 β>
ACK sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803851/0a8b6a3ea35c8b932bdd2e7d7008585dβ,opaque=β6191bf2630478c73β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=βccf9e19eb04dcd85904ae482cb684567β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 641 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0429
<β Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0429
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0429
CSeq: 641 REGISTER
Date: Thu, 27 Feb 2020 11:44:41 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803851/0a8b6a3ea35c8b932bdd2e7d7008585dβ,opaque=β6191bf2630478c73β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=βccf9e19eb04dcd85904ae482cb684567β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 642 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042a
<β Transmitting SIP response (548 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042a
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042a
CSeq: 642 REGISTER
WWW-Authenticate: Digest realm=βasteriskβ,nonce=β1582803910/bd7dfb7b5ddf63009ca0392a4302337cβ,opaque=β2e731e5e691d79d7β,stale=true,algorithm=md5,qop=βauthβ
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803910/bd7dfb7b5ddf63009ca0392a4302337cβ,opaque=β2e731e5e691d79d7β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=β43ae11a82dafa21df4f4d4f4b6e3e630β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 643 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042c
<β Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042c
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042c
CSeq: 643 REGISTER
Date: Thu, 27 Feb 2020 11:45:10 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803910/bd7dfb7b5ddf63009ca0392a4302337cβ,opaque=β2e731e5e691d79d7β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=β43ae11a82dafa21df4f4d4f4b6e3e630β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 644 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042d
<β Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042d
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042d
CSeq: 644 REGISTER
Date: Thu, 27 Feb 2020 11:45:40 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0
<β Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=βasteriskβ, nonce=β1582803910/bd7dfb7b5ddf63009ca0392a4302337cβ,opaque=β2e731e5e691d79d7β,algorithm=md5,qop=auth, username=β37400β, uri=βsip:192.168.133.111β, response=β43ae11a82dafa21df4f4d4f4b6e3e630β, cnonce=β95d98670β, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 645 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042e
<β Transmitting SIP response (548 bytes) to UDP:192.168.133.110:5060 β>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042e
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042e
CSeq: 645 REGISTER
WWW-Authenticate: Digest realm=βasteriskβ,nonce=β1582803969/b5bcfa6c274e8881485c7c8b12db59d9β,opaque=β77f67719619803eeβ,stale=true,algorithm=md5,qop=βauthβ
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0