Here is another trace with all numbers left intact. This one includes a sip debugging.
<------------->
— (16 headers 11 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 216.82.225.24 : 5060 (NAT)
Using INVITE request as basis request - c544580e-52fe-122d-6fac-0015c5eaaddb
No user ‘4798998070’ in SIP users list
Found peer ‘fpbx-1-e03ad727’ for ‘4798998070’ from 216.82.225.24:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 67.231.0.102:5940
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.231.0.102:5940
Looking for 4799669999 in from-pstn (domain 98.172.119.87)
list_route: hop: sip:[email protected]:5060
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
– Executing [4799669999@from-pstn:1] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “__FROM_DID=4799669999”) in new stack
– Executing [4799669999@from-pstn:2] Gosub(“SIP/fpbx-1-e03ad727-097c89e8”, “app-blacklist-check,s,1”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/fpbx-1-e03ad727-097c89e8”, “”) in new stack
– Executing [4799669999@from-pstn:3] ExecIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0 ?Set(CALLERID(name)=4798998070)”) in new stack
– Executing [4799669999@from-pstn:4] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “FAX_RX=”) in new stack
– Executing [4799669999@from-pstn:5] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [4799669999@from-pstn:6] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [4799669999@from-pstn:7] Goto(“SIP/fpbx-1-e03ad727-097c89e8”, “from-did-direct,69999,1”) in new stack
– Goto (from-did-direct,69999,1)
– Executing [69999@from-did-direct:1] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “exten-vm,novm,69999”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “user-callerid”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “AMPUSER=4798998070”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?Set(REALCALLERIDNUM=4798998070)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/fpbx-1-e03ad727-097c89e8”, “Using CallerID “+14798998070” <4798998070>”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “VMBOX=novm”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “EXTTOCALL=69999”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CFUEXT=”) in new stack
– Executing [s@macro-exten-vm:6] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “CFBEXT=”) in new stack
– Executing [s@macro-exten-vm:7] Set(“SIP/fpbx-1-e03ad727-097c89e8”, “RT=”"") in new stack
– Executing [s@macro-exten-vm:8] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “record-enable,69999,IN”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/fpbx-1-e03ad727-097c89e8”, “recordingcheck,20091123-121341,1259000021.17”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091123-121341,1259000021.17: Inbound recording not enabled
– <SIP/fpbx-1-e03ad727-097c89e8>AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/fpbx-1-e03ad727-097c89e8”, “”) in new stack
– Executing [s@macro-exten-vm:9] Macro(“SIP/fpbx-1-e03ad727-097c89e8”, “dial,”",tr,69999") in new stack
– Executing [s@macro-dial:1] GotoIf(“SIP/fpbx-1-e03ad727-097c89e8”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [s@macro-dial:3] AGI(“SIP/fpbx-1-e03ad727-097c89e8”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+14798998070’ number is ‘4798998070’
> dialparties.agi: USE_CONFIRMATION: ‘FALSE’
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 69999 to extension map
> dialparties.agi: Extension 69999 has call screening off
– dialparties.agi: Extension 69999 cf is disabled
– dialparties.agi: Extension 69999 do not disturb is disabled
> dialparties.agi: extnum 69999 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/69999 to 4798998070
– dialparties.agi: Filtered ARG3: 69999
– <SIP/fpbx-1-e03ad727-097c89e8>AGI Script dialparties.agi completed, returning 0
– Executing [s@macro-dial:7] Dial(“SIP/fpbx-1-e03ad727-097c89e8”, “SIP/69999,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Audio is at 192.168.0.4 port 17796
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:13:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 1228763685 1228763685 IN IP4 192.168.0.4
s=Asterisk PBX 1.6.0.17
c=IN IP4 192.168.0.4
t=0 0
m=audio 17796 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called 69999
asterisk1*CLI>
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:41 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
— (11 headers 0 lines) —
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:41 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0
<------------->
— (12 headers 0 lines) —
– SIP/69999-097cf6f0 is ringing
asterisk1*CLI>
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK152e08f5
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:42 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 207
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 26710 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 23700 RTP/AVP 0 101
c=IN IP4 192.168.0.253
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (15 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.253:23700
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.253:23700
list_route: hop: sip:[email protected]:5060;transport=udp
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.0.253, port 5060
Transmitting (no NAT) to 192.168.0.253:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5614444f
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.17
Content-Length: 0
-- SIP/69999-097cf6f0 answered SIP/fpbx-1-e03ad727-097c89e8
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
asterisk1*CLI>
<— Reliably Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654096 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #2 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #3 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #4 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654097 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #5 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
Audio is at 98.172.119.87 port 14546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654098 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #6 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as30aa6d49
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 338654095 338654095 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 14546 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Executing [h@macro-dial:1] Macro("SIP/fpbx-1-e03ad727-097c89e8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/fpbx-1-e03ad727-097c89e8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/fpbx-1-e03ad727-097c89e8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/fpbx-1-e03ad727-097c89e8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/fpbx-1-e03ad727-097c89e8", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-097c89e8’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/fpbx-1-e03ad727-097c89e8’
Scheduling destruction of SIP dialog ‘[email protected]’ in 10624 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.0.253, port 5060
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3e2f85a6
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0.17
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/fpbx-1-e03ad727-097c89e8’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-097c89e8’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 69999, 1) exited non-zero on 'SIP/fpbx-1-e03ad727-097c89e8’
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK3e2f85a6
From: “+14798998070” sip:[email protected];tag=as52ae5b74
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b603a8c04b-081b33b9
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:52 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
asterisk1*CLI>
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 216.82.225.24 : 5060 (NAT)
Using INVITE request as basis request - c544580e-52fe-122d-6fac-0015c5eaaddb
No user ‘4798998070’ in SIP users list
Found peer ‘fpbx-1-e03ad727’ for ‘4798998070’ from 216.82.225.24:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 67.231.0.102:5940
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.231.0.102:5940
Looking for 4799669999 in from-pstn (domain 98.172.119.87)
list_route: hop: sip:[email protected]:5060
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
equire: timer>
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
– Executing [4799669999@from-pstn:1] Set(“SIP/fpbx-1-e03ad727-09820028”, “__FROM_DID=4799669999”) in new stack
– Executing [4799669999@from-pstn:2] Gosub(“SIP/fpbx-1-e03ad727-09820028”, “app-blacklist-check,s,1”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/fpbx-1-e03ad727-09820028”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/fpbx-1-e03ad727-09820028”, “”) in new stack
– Executing [4799669999@from-pstn:3] ExecIf(“SIP/fpbx-1-e03ad727-09820028”, “0 ?Set(CALLERID(name)=4798998070)”) in new stack
– Executing [4799669999@from-pstn:4] Set(“SIP/fpbx-1-e03ad727-09820028”, “FAX_RX=”) in new stack
– Executing [4799669999@from-pstn:5] Set(“SIP/fpbx-1-e03ad727-09820028”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [4799669999@from-pstn:6] Set(“SIP/fpbx-1-e03ad727-09820028”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [4799669999@from-pstn:7] Goto(“SIP/fpbx-1-e03ad727-09820028”, “from-did-direct,69999,1”) in new stack
– Goto (from-did-direct,69999,1)
– Executing [69999@from-did-direct:1] Macro(“SIP/fpbx-1-e03ad727-09820028”, “exten-vm,novm,69999”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/fpbx-1-e03ad727-09820028”, “user-callerid”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/fpbx-1-e03ad727-09820028”, “AMPUSER=4798998070”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/fpbx-1-e03ad727-09820028”, “1?Set(REALCALLERIDNUM=4798998070)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/fpbx-1-e03ad727-09820028”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/fpbx-1-e03ad727-09820028”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?report”) in new stack
– Goto (macro-user-callerid,s,10)
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/fpbx-1-e03ad727-09820028”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/fpbx-1-e03ad727-09820028”, “Using CallerID “+14798998070” <4798998070>”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/fpbx-1-e03ad727-09820028”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/fpbx-1-e03ad727-09820028”, “VMBOX=novm”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/fpbx-1-e03ad727-09820028”, “EXTTOCALL=69999”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/fpbx-1-e03ad727-09820028”, “CFUEXT=”) in new stack
– Executing [s@macro-exten-vm:6] Set(“SIP/fpbx-1-e03ad727-09820028”, “CFBEXT=”) in new stack
– Executing [s@macro-exten-vm:7] Set(“SIP/fpbx-1-e03ad727-09820028”, “RT=”"") in new stack
– Executing [s@macro-exten-vm:8] Macro(“SIP/fpbx-1-e03ad727-09820028”, “record-enable,69999,IN”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/fpbx-1-e03ad727-09820028”, “recordingcheck,20091123-121356,1259000036.19”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091123-121356,1259000036.19: Inbound recording not enabled
– <SIP/fpbx-1-e03ad727-09820028>AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/fpbx-1-e03ad727-09820028”, “”) in new stack
– Executing [s@macro-exten-vm:9] Macro(“SIP/fpbx-1-e03ad727-09820028”, “dial,”",tr,69999") in new stack
– Executing [s@macro-dial:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [s@macro-dial:3] AGI(“SIP/fpbx-1-e03ad727-09820028”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘+14798998070’ number is ‘4798998070’
> dialparties.agi: USE_CONFIRMATION: ‘FALSE’
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 69999 to extension map
> dialparties.agi: Extension 69999 has call screening off
– dialparties.agi: Extension 69999 cf is disabled
– dialparties.agi: Extension 69999 do not disturb is disabled
> dialparties.agi: extnum 69999 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/69999 to 4798998070
– dialparties.agi: Filtered ARG3: 69999
– <SIP/fpbx-1-e03ad727-09820028>AGI Script dialparties.agi completed, returning 0
– Executing [s@macro-dial:7] Dial(“SIP/fpbx-1-e03ad727-09820028”, “SIP/69999,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Audio is at 192.168.0.4 port 18784
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:13:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 277980194 277980194 IN IP4 192.168.0.4
s=Asterisk PBX 1.6.0.17
c=IN IP4 192.168.0.4
t=0 0
m=audio 18784 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called 69999
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:56 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
— (11 headers 0 lines) —
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:13:56 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0
<------------->
— (12 headers 0 lines) —
– SIP/69999-b7cfa010 is ringing
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Length: 0
<------------>
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK771d5183
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as45278ced
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:14:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:51094 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK771d5183
From: “Unknown” sip:[email protected];tag=as45278ced
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b81c67d062-60499352
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:14:03 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/8.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 237
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5385 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 0 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
<------------->
— (16 headers 11 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.0.253:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK7e442278
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3b834e4a
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.17
Date: Mon, 23 Nov 2009 18:14:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:51095 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK7e442278
From: “Unknown” sip:[email protected];tag=as3b834e4a
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b9575acab0-2d752a68
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:14:03 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/8.0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 237
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 7377 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 0 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
<------------->
— (16 headers 11 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK60a18d72
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
Call-ID: [email protected]
Date: Mon, 23 Nov 2009 18:14:06 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “9669999” sip:[email protected];party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 206
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 7873 0 IN IP4 192.168.0.253
s=SIP Call
t=0 0
m=audio 23820 RTP/AVP 0 101
c=IN IP4 192.168.0.253
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (15 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.253:23820
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.253:23820
list_route: hop: sip:[email protected]:5060;transport=udp
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.0.253, port 5060
Transmitting (no NAT) to 192.168.0.253:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6b0d8d5c
Max-Forwards: 70
From: “+14798998070” sip:[email protected];tag=as43e496ad
To: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.17
Content-Length: 0
-- SIP/69999-b7cfa010 answered SIP/fpbx-1-e03ad727-09820028
Audio is at 98.172.119.87 port 13290
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
all-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #2 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
asterisk1*CLI>
<— SIP read from UDP://192.168.0.253:50975 —>
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bK5e25a1a3
From: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
To: “+14798998070” sip:[email protected];tag=as43e496ad
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 23 Nov 2009 18:14:08 GMT
CSeq: 101 BYE
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Sending to 192.168.0.253 : 5060 (no NAT)
<— Transmitting (no NAT) to 192.168.0.253:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bK5e25a1a3;received=192.168.0.253
From: sip:[email protected]:5060;transport=udp;tag=0014a97146da00b7193ddf46-7992fe64
To: “+14798998070” sip:[email protected];tag=as43e496ad
Call-ID: [email protected]
CSeq: 101 BYE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
– Executing [h@macro-dial:1] Macro(“SIP/fpbx-1-e03ad727-09820028”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/fpbx-1-e03ad727-09820028”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/fpbx-1-e03ad727-09820028”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/fpbx-1-e03ad727-09820028’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 69999, 1) exited non-zero on 'SIP/fpbx-1-e03ad727-09820028’
Scheduling destruction of SIP dialog ‘c544580e-52fe-122d-6fac-0015c5eaaddb’ in 6400 ms (Method: INVITE)
Really destroying SIP dialog ‘[email protected]’ Method: BYE
Retransmitting #3 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #4 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #5 (NAT) to 216.82.225.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 1445253653 1445253653 IN IP4 98.172.119.87
s=Asterisk PBX 1.6.0.17
c=IN IP4 98.172.119.87
t=0 0
m=audio 13290 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.82.231.10:5060:
REGISTER sip:trunk2.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK203b53c5;rport
Max-Forwards: 70
From: sip:[email protected];tag=as789ef87a
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 1.6.0.17
Authorization: Digest username=“e03ad727”, realm=“trunk2.freepbx.com”, algorithm=MD5, uri=“sip:trunk2.freepbx.com”, nonce=“fdf02a03-69b4-4b24-ba1d-0224c1175980”, response=“cfd38493224719ed52ed544444615ef2”, qop=auth, cnonce=“183f6075”, nc=00000006
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.82.225.24:5060:
REGISTER sip:trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK4160b5d0;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3e90bd6b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 1.6.0.17
Authorization: Digest username=“e03ad727”, realm=“trunk1.freepbx.com”, algorithm=MD5, uri=“sip:trunk1.freepbx.com”, nonce=“c6a02b28-d85a-11de-8068-1fcf8b0b91b7”, response=“487ba8148b61fd2b489192f958b945db”, qop=auth, cnonce=“5785ab89”, nc=00000006
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0
<— SIP read from UDP://216.82.231.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK203b53c5;rport=5060
From: sip:[email protected];tag=as789ef87a
To: sip:[email protected];tag=jS03y4Uy72NZm
Call-ID: [email protected]
CSeq: 108 REGISTER
Contact: sip:[email protected];expires=120
Date: Mon, 23 Nov 2009 18:14:11 GMT
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
asterisk1*CLI> sip se
<— SIP read from UDP://216.82.231.10:5060 —>
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.231.10;rport;branch=z9hG4bK10Fjve7pDX2Qa
Max-Forwards: 70
From: sip:[email protected];tag=K2Sv0Zc24Bcjg
To: sip:[email protected]
Call-ID: d749ee5f-52fe-122d-84a4-001372fb8c08
CSeq: 123381689 NOTIFY
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: terminated;reason=timeout
Content-Type: application/simple-message-summary
Content-Length: 74
Messages-Waiting: no
Message-Account: sip:[email protected]
<------------->
— (16 headers 3 lines) —
<— Transmitting (NAT) to 216.82.231.10:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 216.82.231.10;branch=z9hG4bK10Fjve7pDX2Qa;received=216.82.231.10;rport=5060
From: sip:[email protected];tag=K2Sv0Zc24Bcjg
To: sip:[email protected];tag=as61151a44
Call-ID: d749ee5f-52fe-122d-84a4-001372fb8c08
CSeq: 123381689 NOTIFY
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
asterisk1*CLI> sip se
<— SIP read from UDP://216.82.225.24:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 98.172.119.87:5060;branch=z9hG4bK4160b5d0;rport=5060
From: sip:[email protected];tag=as3e90bd6b
To: sip:[email protected];tag=m4yBaK34Q97Be
Call-ID: [email protected]
CSeq: 108 REGISTER
Contact: sip:[email protected];expires=120
Date: Mon, 23 Nov 2009 18:14:09 GMT
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
asterisk1*CLI> sip set
<— SIP read from UDP://216.82.225.24:5060 —>
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKtgvpFtD9U6UFr
Max-Forwards: 70
From: sip:[email protected];tag=QZapF4NFF4a4g
To: sip:[email protected]
Call-ID: d75dd3cf-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381689 NOTIFY
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: terminated;reason=timeout
Content-Type: application/simple-message-summary
Content-Length: 74
Messages-Waiting: no
Message-Account: sip:[email protected]
<------------->
— (16 headers 3 lines) —
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKtgvpFtD9U6UFr;received=216.82.225.24;rport=5060
From: sip:[email protected];tag=QZapF4NFF4a4g
To: sip:[email protected];tag=as23d96056
Call-ID: d75dd3cf-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381689 NOTIFY
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
asterisk1*CLI> sip set debug
<— SIP read from UDP://216.82.225.24:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKUQ3vZH8ajN0Ne
Max-Forwards: 51
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected]
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
Contact: sip:[email protected]:5060
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 271
Remote-Party-ID: “+14798998070” sip:[email protected];party=calling;screen=yes;privacy=off
v=0
o=Sonus_UAC 28939 21363 IN IP4 192.168.27.72
s=SIP Media Capabilities
c=IN IP4 67.231.0.102
t=0 0
m=audio 5940 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (16 headers 12 lines) —
Ignoring this INVITE request
<— Transmitting (NAT) to 216.82.225.24:5060 —>
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKUQ3vZH8ajN0Ne;received=216.82.225.24;rport=5060
From: “+14798998070” sip:[email protected];tag=76DZc5Fg1cZ6H
To: sip:[email protected];tag=as75d29534
Call-ID: c544580e-52fe-122d-6fac-0015c5eaaddb
CSeq: 123381674 INVITE
User-Agent: Asterisk PBX 1.6.0.17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘c544580e-52fe-122d-6fac-0015c5eaaddb’ in 6400 ms (Method: INVITE)
asterisk1CLI> sip set debug off
SIP Debugging Disabled
asterisk1CLI>