SIP Debug ON
Connected to Asterisk 16.1.1 currently running on CEESO-INTL01 (pid = 89095)
CEESO-INTL01CLI> sip set debug on
SIP Debugging enabled
CEESO-INTL01CLI>
CEESO-INTL01CLI>
CEESO-INTL01CLI>
CEESO-INTL01CLI>
CEESO-INTL01CLI>
CEESO-INTL01CLI>
CEESO-INTL01CLI>
CEESO-INTL01*CLI> sip reload
Reloading SIP
Reliably Transmitting (no NAT) to 162.223.83.245:5061:
OPTIONS sip:tlsnyc.granitevoip.com SIP/2.0
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK00739899
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.27.189.133;tag=as457c2304
To: sip:tlsnyc.granitevoip.com
Contact: sip:asterisk@10.27.189.133:5061;transport=tls
Call-ID: 059021da6e8625fd2a72533b357e0855@10.27.189.133:5061
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.1
Date: Thu, 14 Feb 2019 18:21:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from TLS:162.223.83.245:5061 —>
SIP/2.0 200 OK
Call-ID: 059021da6e8625fd2a72533b357e0855@10.27.189.133:5061
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@10.27.189.133;tag=as457c2304
To: sip:tlsnyc.granitevoip.com;tag=sip+1+81700108+4adca2be
Via: SIP/2.0/TLS 10.27.189.133:5061;received=52.232.129.229;branch=z9hG4bK00739899
Content-Length: 0
Supported: resource-priority, siprec, 100rel
Server: DC-SIP/2.0
Organization: Metaswitch Networks
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH
Accept-Encoding: identity
Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/hook-flash, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml, application/vnd.telekom.service_indication+xml, application/calling-name-info
<------------->
— (14 headers 0 lines) —
[Feb 14 18:21:33] NOTICE[89161]: chan_sip.c:15821 sip_reregister: – Re-registration for 5813196241@tlsnyc.granitevoip.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 162.223.83.245:5061:
REGISTER sip:tlsnyc.granitevoip.com:5061 SIP/2.0
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK280e519b
Max-Forwards: 70
From: sip:5813196241@tlsnyc.granitevoip.com;tag=as2fad55bb
To: sip:5813196241@tlsnyc.granitevoip.com
Call-ID: 61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]
CSeq: 193 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 16.1.1
Authorization: Digest username=“5813196241”, realm=“tlsnyc.granitevoip.com”, algorithm=MD5, uri=“sip:tlsnyc.granitevoip.com:5061”, nonce=“36661b27dd27”, response=“7a25fd50608b4ed302f833f369c37c14”, qop=auth, cnonce=“6de6ffe4”, nc=0000005b
Expires: 120
Contact: sip:5813196241@10.27.189.133:5061;transport=tls
Content-Length: 0
Really destroying SIP dialog ‘059021da6e8625fd2a72533b357e0855@10.27.189.133:5061’ Method: OPTIONS
<— SIP read from TLS:162.223.83.245:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK280e519b
Max-Forwards: 70
From: sip:5813196241@tlsnyc.granitevoip.com;tag=as2fad55bb
To: sip:5813196241@tlsnyc.granitevoip.com
Call-ID: 61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]
CSeq: 193 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 16.1.1
Authorization: Digest username=“5813196241”, realm=“tlsnyc.granitevoip.com”, algorithm=MD5, uri=“sip:tlsnyc.granitevoip.com:5061”, nonce=“36661b27dd27”, response=“7a25fd50608b4ed302f833f369c37c14”, qop=auth, cnonce=“6de6ffe4”, nc=0000005b
Expires: 30
Contact: sip:5813196241@10.27.189.133:5061;transport=tls;expires=30
Content-Length: 0
<------------->
— (13 headers 0 lines) —
[Feb 14 18:21:33] NOTICE[89163]: chan_sip.c:24828 handle_response_register: Outbound Registration: Expiry for tlsnyc.granitevoip.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog ‘61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]’ Method: REGISTER
CEESO-INTL01*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
tlsnyc.granitevoip.com:5061 N 5813196241 24 Registered Thu, 14 Feb 2019 18:21:33
1 SIP registrations.
[Feb 14 18:21:58] NOTICE[89161]: chan_sip.c:15821 sip_reregister: – Re-registration for 5813196241@tlsnyc.granitevoip.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 162.223.83.245:5061:
REGISTER sip:tlsnyc.granitevoip.com:5061 SIP/2.0
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK3519619d
Max-Forwards: 70
From: sip:5813196241@tlsnyc.granitevoip.com;tag=as2fad55bb
To: sip:5813196241@tlsnyc.granitevoip.com
Call-ID: 61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]
CSeq: 194 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 16.1.1
Authorization: Digest username=“5813196241”, realm=“tlsnyc.granitevoip.com”, algorithm=MD5, uri=“sip:tlsnyc.granitevoip.com:5061”, nonce=“36661b27dd27”, response=“6c2d07034408c34788c2b19c6ed1a9ed”, qop=auth, cnonce=“6763bf33”, nc=0000005c
Expires: 120
Contact: sip:5813196241@10.27.189.133:5061;transport=tls
Content-Length: 0
<— SIP read from TLS:162.223.83.245:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK3519619d
Max-Forwards: 70
From: sip:5813196241@tlsnyc.granitevoip.com;tag=as2fad55bb
To: sip:5813196241@tlsnyc.granitevoip.com
Call-ID: 61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]
CSeq: 194 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 16.1.1
Authorization: Digest username=“5813196241”, realm=“tlsnyc.granitevoip.com”, algorithm=MD5, uri=“sip:tlsnyc.granitevoip.com:5061”, nonce=“36661b27dd27”, response=“6c2d07034408c34788c2b19c6ed1a9ed”, qop=auth, cnonce=“6763bf33”, nc=0000005c
Expires: 30
Contact: sip:5813196241@10.27.189.133:5061;transport=tls;expires=30
Content-Length: 0
<------------->
— (13 headers 0 lines) —
[Feb 14 18:21:58] NOTICE[89163]: chan_sip.c:24828 handle_response_register: Outbound Registration: Expiry for tlsnyc.granitevoip.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog ‘61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]’ Method: REGISTER
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
[Feb 14 18:22:23] NOTICE[89161]: chan_sip.c:15821 sip_reregister: – Re-registration for 5813196241@tlsnyc.granitevoip.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 162.223.83.245:5061:
REGISTER sip:tlsnyc.granitevoip.com:5061 SIP/2.0
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK08ff3161
Max-Forwards: 70
From: sip:5813196241@tlsnyc.granitevoip.com;tag=as2fad55bb
To: sip:5813196241@tlsnyc.granitevoip.com
Call-ID: 61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]
CSeq: 195 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 16.1.1
Authorization: Digest username=“5813196241”, realm=“tlsnyc.granitevoip.com”, algorithm=MD5, uri=“sip:tlsnyc.granitevoip.com:5061”, nonce=“36661b27dd27”, response=“e01987e06b137d54d520156d4565032a”, qop=auth, cnonce=“62c35850”, nc=0000005d
Expires: 120
Contact: sip:5813196241@10.27.189.133:5061;transport=tls
Content-Length: 0
<— SIP read from TLS:162.223.83.245:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK08ff3161
Max-Forwards: 70
From: sip:5813196241@tlsnyc.granitevoip.com;tag=as2fad55bb
To: sip:5813196241@tlsnyc.granitevoip.com
Call-ID: 61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]
CSeq: 195 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 16.1.1
Authorization: Digest username=“5813196241”, realm=“tlsnyc.granitevoip.com”, algorithm=MD5, uri=“sip:tlsnyc.granitevoip.com:5061”, nonce=“36661b27dd27”, response=“e01987e06b137d54d520156d4565032a”, qop=auth, cnonce=“62c35850”, nc=0000005d
Expires: 30
Contact: sip:5813196241@10.27.189.133:5061;transport=tls;expires=30
Content-Length: 0
<------------->
— (13 headers 0 lines) —
[Feb 14 18:22:23] NOTICE[89163]: chan_sip.c:24828 handle_response_register: Outbound Registration: Expiry for tlsnyc.granitevoip.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog ‘61a19cf078c6781345e7ce3372f69fef@[fe80::20d:3aff:fef4:9a6c]’ Method: REGISTER
Reliably Transmitting (no NAT) to 162.223.83.245:5061:
OPTIONS sip:tlsnyc.granitevoip.com SIP/2.0
Via: SIP/2.0/TLS 10.27.189.133:5061;branch=z9hG4bK7809db1a
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.27.189.133;tag=as1bc1ea04
To: sip:tlsnyc.granitevoip.com
Contact: sip:asterisk@10.27.189.133:5061;transport=tls
Call-ID: 3b7503eb649c11be75e639432d6fe1fc@10.27.189.133:5061
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.1
Date: Thu, 14 Feb 2019 18:22:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from TLS:162.223.83.245:5061 —>
SIP/2.0 200 OK
Call-ID: 3b7503eb649c11be75e639432d6fe1fc@10.27.189.133:5061
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@10.27.189.133;tag=as1bc1ea04
To: sip:tlsnyc.granitevoip.com;tag=sip+3+bcee011f+d913be92
Via: SIP/2.0/TLS 10.27.189.133:5061;received=52.232.129.229;branch=z9hG4bK7809db1a
Content-Length: 0
Supported: resource-priority, siprec, 100rel
Server: DC-SIP/2.0
Organization: Metaswitch Networks
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH
Accept-Encoding: identity
Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/hook-flash, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml, application/vnd.telekom.service_indication+xml, application/calling-name-info
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘3b7503eb649c11be75e639432d6fe1fc@10.27.189.133:5061’ Method: OPTIONS
CEESO-INTL01*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@CEESO-INTL01 asterisk]# ^C
[root@CEESO-INTL01 asterisk]#