<— SIP read from UDP:43.254.109.83:5072 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 13.234.154.122:5060;branch=z9hG4bK382f4669;rport=5060
From: “asterisk” sip:asterisk@13.234.154.122;tag=as15d3c02a
To: sip:812@43.254.109.83:5072;tag=124051058
Call-ID: 5573dea744dfe84b4fc119767494e235@13.234.154.122:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT818 1.0.8.3
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘5573dea744dfe84b4fc119767494e235@13.234.154.122:5060’ Method: OPTIONS
<— SIP read from UDP:103.249.83.166:5060 —>
INVITE sip:1164@13.234.154.122 SIP/2.0
From: sip:08680863226@103.249.83.166:5060;tag=dab6c8-af04a8c0-13c4-5d2313ef-4be675ef-5d2313ef
To: sip:1164@13.234.154.122
Call-ID: db9b60-af04a8c0-13c4-5d2313ef-73be1541-5d2313ef
CSeq: 1 INVITE
Via: SIP/2.0/UDP 103.249.83.166:5060;rport;branch=z9hG4bK-5d2313ef-d105ddb6-fcc4815
P-Asserted-Identity: sip:08680863226@192.168.4.175
Max-Forwards: 70
Supported: replaces
User-Agent: Matrix-SETUVTEP
Contact: sip:103.249.83.166:5060;transport=udp
Allow: INVITE,BYE,ACK,CANCEL,REFER,NOTIFY,OPTIONS
Content-Type: application/sdp
Content-Length: 278
v=0
o=- 3922918716 3922918716 IN IP4 103.249.83.166
s=Matrix-SETUVTEP
c=IN IP4 103.249.83.166
t=0 0
m=audio 8040 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=rtcp:8041 IN IP4 103.249.83.166
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 103.249.83.166:5060 (NAT)
Sending to 103.249.83.166:5060 (NAT)
Using INVITE request as basis request - db9b60-af04a8c0-13c4-5d2313ef-73be1541-5d2313ef
Found peer ‘matrix’ for ‘08680863226’ from 103.249.83.166:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 103.249.83.166:8040
Looking for 1164 in kunamt1 (domain 13.234.154.122)
sip_route_dump: route/path hop: sip:103.249.83.166:5060;transport=udp
<— Transmitting (NAT) to 103.249.83.166:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 103.249.83.166:5060;branch=z9hG4bK-5d2313ef-d105ddb6-fcc4815;received=103.249.83.166;rport=5060
From: sip:08680863226@103.249.83.166:5060;tag=dab6c8-af04a8c0-13c4-5d2313ef-4be675ef-5d2313ef
To: sip:1164@13.234.154.122
Call-ID: db9b60-af04a8c0-13c4-5d2313ef-73be1541-5d2313ef
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1164@13.234.154.122:5060
Content-Length: 0
<------------>
– Executing [1164@kunamt1:1] MixMonitor(“SIP/matrix-00000080”, “/var/www/html/monitor/from-internal/1562579981.194.wav,”) in new stack
– Executing [1164@kunamt1:2] Set(“SIP/matrix-00000080”, “__src=116”) in new stack
– Executing [1164@kunamt1:3] NoOp(“SIP/matrix-00000080”, "1164 SIP/matrix-00000080–116–0 ") in new stack
– Executing [1164@kunamt1:4] Set(“SIP/matrix-00000080”, “__uniqueid=1562579981.194”) in new stack
– Executing [1164@kunamt1:5] Queue(“SIP/matrix-00000080”, “testing,trn,60,qm-testing”) in new stack
[Jul 8 15:29:41] WARNING[14227][C-00000044]: app_queue.c:8299 queue_exec: Unable to join queue ‘testing’
– Executing [1164@kunamt1:6] Wait(“SIP/matrix-00000080”, “1”) in new stack
== Begin MixMonitor Recording SIP/matrix-00000080
<— SIP read from UDP:103.249.83.162:1300 —>
REGISTER sip:zzz.pulxx.com SIP/2.0
Via: SIP/2.0/UDP 103.249.83.162:1300;branch=z9hG4bK371567932DE99EFE
From: “111” sip:111@zzz.pulxx.com;tag=6118B17E-A0F5393
To: sip:111@zzz.pulxx.com
CSeq: 127 REGISTER
Call-ID: a548347e-ab25093-fc1435fe@192.168.4.69
Contact: sip:111@103.249.83.162:1300;methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.7.2514
Accept-Language: en
Authorization: Digest username=“111”, realm=“asterisk”, nonce=“5348f501”, uri=“sip:zzz.pulxx.com”, response=“81f00d28485b171955c7f89a6c67a928”, algorithm=MD5
Max-Forwards: 70
Expires: 10
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 103.249.83.162:1300 (NAT)
[Jul 8 15:29:41] NOTICE[13458]: chan_sip.c:17421 check_auth: Correct auth, but based on stale nonce received from ‘“111” sip:111@zzz.pulxx.com;tag=6118B17E-A0F5393’
<— Transmitting (NAT) to 103.249.83.162:1300 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 103.249.83.162:1300;branch=z9hG4bK371567932DE99EFE;received=103.249.83.162;rport=1300
From: “111” sip:111@zzz.pulxx.com;tag=6118B17E-A0F5393
To: sip:111@zzz.pulxx.com;tag=as27db7c38
Call-ID: a548347e-ab25093-fc1435fe@192.168.4.69
CSeq: 127 REGISTER
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“37981dbc”, stale=true
Content-Length: 0