Last login: Tue Oct 31 15:06:10 2017 from 192.168.70.3
| | __ ___ | _ | __ ) / /
| | | '/ _ / _ \ |) | _ \ /
| || | | __/ __/ __/| |) /
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NOTICE! You have 9 notifications! Please log into the UI to see them!
Current Network Configuration
±----------±------------------±-------------------------+
| Interface | MAC Address | IP Addresses |
±----------±------------------±-------------------------+
| eth0 | 00:0C:29:34:62:C0 | 192.168.50.1 |
| | | fe80::20c:29ff:fe34:62c0 |
±----------±------------------±-------------------------+
Please note most tasks should be handled through the GUI.
You can access the GUI by typing one of the above IPs in to your web browser.
For support please visit:
http://www.freepbx.org/support-and-professional-services
[root@MBICPBX01 ~]# asterisk -rvvv
Asterisk 13.12.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
Connected to Asterisk 13.12.1 currently running on MBICPBX01 (pid = 2016)
MBICPBX01*CLI> sip set debug on
SIP Debugging re-enabled
<— SIP read from UDP:203.161.160.71:5060 —>
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKn4f9b51040lk43grmpu0.1
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected]
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738605 INVITE
Contact: sip:[email protected]:5060;transport=udp
P-Called-Party-ID: sip:[email protected]
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 220378594 1 IN IP4 203.161.160.71
s=-
c=IN IP4 203.161.160.71
t=0 0
m=audio 23746 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:30
a=sqn:0
a=cdsc: 1 image udptl t38
<------------->
— (17 headers 17 lines) —
Sending to 203.161.160.71:5060 (NAT)
Sending to 203.161.160.71:5060 (NAT)
Using INVITE request as basis request - SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
Found peer ‘Engine’ for ‘anonymous’ from 203.161.160.71:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 203.161.160.71:23746
Looking for s in from-trunk-sip-Engine (domain 110.XXX.XX.XX)
sip_route_dump: route/path hop: sip:[email protected]:5060;transport=udp
<— Transmitting (NAT) to 203.161.160.71:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKn4f9b51040lk43grmpu0.1;received=203.161.160.71;rport=5060
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected]
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738605 INVITE
Server: FPBX-13.0.192.16(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5160
Content-Length: 0
<------------>
– Executing [s@from-trunk-sip-Engine:1] Set(“SIP/Engine-00000005”, “GROUP()=OUT_2”) in new stack
– Executing [s@from-trunk-sip-Engine:2] Goto(“SIP/Engine-00000005”, “from-trunk,s,1”) in new stack
– Goto (from-trunk,s,1)
– Executing [s@from-trunk:1] NoOp(“SIP/Engine-00000005”, “No DID or CID Match”) in new stack
– Executing [s@from-trunk:2] Answer(“SIP/Engine-00000005”, “”) in new stack
Audio is at 10008
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 203.161.160.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKn4f9b51040lk43grmpu0.1;received=203.161.160.71;rport=5060
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738605 INVITE
Server: FPBX-13.0.192.16(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5160
Content-Type: application/sdp
Require: timer
Content-Length: 325
v=0
o=root 1251476768 1251476768 IN IP4 110.XXX.XX.XX
s=Asterisk PBX 13.12.1
c=IN IP4 110.XXX.XX.XX
t=0 0
m=audio 10008 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
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:150
a=sendrecv
<------------>
<— SIP read from UDP:203.161.160.71:5060 —>
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKruao2l009g3eiuvl0vj0.1
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738605 ACK
Contact: sip:[email protected]:5060;transport=udp
Max-Forwards: 29
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:203.161.160.71:5060 —>
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKruao2l009g3eiuvl0vj0cbopj29n0.1
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738606 INVITE
Contact: sip:[email protected]:5060;transport=udp
Remote-Party-ID: "Anonymous"sip:[email protected];screen=yes;party=called;privacy=full;id-type=subscriber
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported: timer
Min-SE: 60
Session-Expires: 1800;refresher=uas
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 220378594 2 IN IP4 203.161.160.71
s=-
c=IN IP4 203.161.160.71
t=0 0
m=audio 23746 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sqn:0
a=cdsc: 1 image udptl t38
<------------->
— (16 headers 17 lines) —
Sending to 203.161.160.71:5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 203.161.160.71:23746
<— Transmitting (NAT) to 203.161.160.71:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKruao2l009g3eiuvl0vj0cbopj29n0.1;received=203.161.160.71;rport=5060
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738606 INVITE
Server: FPBX-13.0.192.16(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5160
Content-Length: 0
<------------>
Audio is at 10008
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 203.161.160.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKruao2l009g3eiuvl0vj0cbopj29n0.1;received=203.161.160.71;rport=5060
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738606 INVITE
Server: FPBX-13.0.192.16(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5160
Content-Type: application/sdp
Require: timer
Content-Length: 325
v=0
o=root 1251476768 1251476769 IN IP4 110.XXX.XX.XX
s=Asterisk PBX 13.12.1
c=IN IP4 110.XXX.XX.XX
t=0 0
m=audio 10008 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
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:150
a=sendrecv
<------------>
<— SIP read from UDP:203.161.160.71:5060 —>
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKin9b3k30a0btiqecfd00.1
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738606 ACK
Contact: sip:[email protected]:5060;transport=udp
Max-Forwards: 29
Content-Length: 0
<------------->
— (9 headers 0 lines) —
[2017-10-31 15:09:58] NOTICE[2250]: chan_sip.c:15603 sip_reregister: – Re-registration for [email protected]
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 203.161.160.71:5060:
REGISTER sip:trunk.engin.com.au SIP/2.0
Via: SIP/2.0/UDP 110.XXX.XX.XX:5160;branch=z9hG4bK78528585;rport
Max-Forwards: 70
From: sip:[email protected];tag=as2268ef53
To: sip:[email protected]
Call-ID: 055fd16e150918a629fdc50157653633@[::1]
CSeq: 1109 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.192.16(13.12.1)
Authorization: Digest username=“0372349870”, realm=“voice.mibroadband.com.au”, algorithm=MD5, uri=“sip:trunk.engin.com.au”, nonce=“BroadWorksXj9f2bkpwTr3vfv8BW”, response=“25161d2540407e7960e589a3b7b05696”, qop=auth, cnonce=“69add447”, nc=0000002c
Expires: 120
Contact: sip:[email protected]:5160
Content-Length: 0
<— SIP read from UDP:203.161.160.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 110.XXX.XX.XX:5160;received=58.96.75.50;branch=z9hG4bK78528585;rport=5160
From: sip:[email protected];tag=as2268ef53
To: sip:[email protected];tag=aprqum37goai33c43-u1vpbm00001a5
Call-ID: 055fd16e150918a629fdc50157653633@[::1]
CSeq: 1109 REGISTER
Contact: sip:[email protected]:5160;expires=60
<------------->
— (7 headers 0 lines) —
[2017-10-31 15:09:58] NOTICE[2250]: chan_sip.c:24396 handle_response_register: Outbound Registration: Expiry for trunk.engin.com.au is 60 sec (Scheduling reregistration in 45 s)
Really destroying SIP dialog ‘055fd16e150918a629fdc50157653633@[::1]’ Method: REGISTER
– Executing [s@from-trunk:3] Log(“SIP/Engine-00000005”, “WARNING,Friendly Scanner from 203.161.160.71”) in new stack
[2017-10-31 15:09:58] WARNING[16409][C-00000006]: Ext. s:3 @ from-trunk: Friendly Scanner from 203.161.160.71
– Executing [s@from-trunk:4] Wait(“SIP/Engine-00000005”, “2”) in new stack
– Executing [s@from-trunk:5] Playback(“SIP/Engine-00000005”, “ss-noservice”) in new stack
– <SIP/Engine-00000005> Playing ‘ss-noservice.ulaw’ (language ‘en_AU’)
<— SIP read from UDP:203.161.160.71:5060 —>
BYE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKruao2l009g3eiuvl0vj0cdopj2pn0.1
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738607 BYE
Max-Forwards: 29
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Sending to 203.161.160.71:5060 (NAT)
Scheduling destruction of SIP dialog ‘SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0’ in 32000 ms (Method: BYE)
<— Transmitting (NAT) to 203.161.160.71:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKruao2l009g3eiuvl0vj0cdopj2pn0.1;received=203.161.160.71;rport=5060
From: "Anonymous"sip:[email protected];tag=SDrketd01-463339434-1509422998105-
To: "Joe Bloggs"sip:[email protected];tag=as538c59fe
Call-ID: SDrketd01-6471232ddad54f73b5f6107ab02834cc-jm6gpa0
CSeq: 944738607 BYE
Server: FPBX-13.0.192.16(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
– Executing [h@from-trunk:1] Macro(“SIP/Engine-00000005”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/Engine-00000005”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/Engine-00000005”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/Engine-00000005”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/Engine-00000005’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/Engine-00000005’
MBICPBX01*CLI>