rendered paste body
User-Agent: SJphone/1.60.289a (SJ Labs)
Proxy-Authorization: Digest username="106",realm="asterisk",nonce="67d05f14",uri="sip:97418592@pt3",response="ee61bf0f7210eb0508e02400487d99af",algorithm="MD5"
v=0
o=- 3384830620 3384830620 IN IP4 218.212.116.89
s=SJphone
c=IN IP4 218.212.116.89
t=0 0
a=direction:active
m=audio 49164 RTP/AVP 3 97 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
--- (12 headers 15 lines) ---
Using INVITE request as basis request - 8CCD023D-A4FA-4887-9A8F-AAAD9CDC9B93@192.168.0.145
Sending to 192.168.0.145 : 5060 (NAT)
Found user '106'
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 218.212.116.89:49164
Peer video RTP is at port 218.212.116.89:65535
Found description format GSM
Found description format iLBC
Found description format iLBC
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x1c050e (gsm|ulaw|alaw|g729|ilbc|h261|h263|h263p), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 97418592 in from-internal (domain pt3)
list_route: hop: <sip:106@218.212.116.89:5060>
Transmitting (NAT) to 218.212.116.89:60919:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a800910000004d4615ec1c00006fba0000009d;received=218.212.116.89;rport=60919
From: "unknown"<sip:106@pt3>;tag=5205131213109
To: <sip:97418592@pt3>
Call-ID: 8CCD023D-A4FA-4887-9A8F-AAAD9CDC9B93@192.168.0.145
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:97418592@192.168.0.102>
Content-Length: 0
---
-- Executing Macro("SIP/106-082435b0", "dialout-trunk|2|97418592||") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_TRUNK=2") in new stack
-- Executing Set("SIP/106-082435b0", "_NODEST=") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_NUMBER=97418592") in new stack
-- Executing Set("SIP/106-082435b0", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=trwW") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,8)
-- Executing Set("SIP/106-082435b0", "GROUP()=OUT_2") in new stack
-- Executing Macro("SIP/106-082435b0", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/106-082435b0", "user-callerid: device 106") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?start") in new stack
-- Executing Set("SIP/106-082435b0", "REALCALLERIDNUM=106") in new stack
-- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack
-- Executing Set("SIP/106-082435b0", "AMPUSER=106") in new stack
-- Executing Set("SIP/106-082435b0", "AMPUSERCIDNAME=Remote1") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack
-- Executing Set("SIP/106-082435b0", "CALLERID(all)=Remote1 <106>") in new stack
-- Executing Set("SIP/106-082435b0", "REALCALLERIDNUM=106") in new stack
-- Executing NoOp("SIP/106-082435b0", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/106-082435b0", "Using CallerID "Remote1" <106>") in new stack
-- Executing Macro("SIP/106-082435b0", "record-enable|106|OUT") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/106-082435b0", "recordingcheck|20070406-144312|1175841791.22") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070406-144312|1175841791.22: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/106-082435b0", "No recording needed") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?skipoutcid") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=TrwW") in new stack
-- Executing Macro("SIP/106-082435b0", "outbound-callerid|2") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/106-082435b0", "USEROUTCID=") in new stack
-- Executing Set("SIP/106-082435b0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/106-082435b0", "TRUNKOUTCID="Totalhearing" <6568845638>") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/106-082435b0", "0?usercid") in new stack
-- Executing Set("SIP/106-082435b0", "CALLERID(all)="Totalhearing" <6568845638>") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/106-082435b0", "CallerID set to "Totalhearing" <6568845638>") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,16)
-- Executing DeadAGI("SIP/106-082435b0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 6XXXXXXX
> fixlocalprefix: Using pattern 9XXXXXXX
== fixlocalprefix: Dialpattern 9XXXXXXX matched. 97418592 -> 97418592
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/106-082435b0", "OUTNUM=97418592") in new stack
-- Executing Set("SIP/106-082435b0", "custom=ZAP/4") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?customtrunk") in new stack
-- Executing Dial("SIP/106-082435b0", "ZAP/4/97418592|300|TrwW") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/106-082435b0", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/106-082435b0", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/106-082435b0", "dialout-trunk|1|97418592||") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_TRUNK=1") in new stack
-- Executing Set("SIP/106-082435b0", "_NODEST=") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_NUMBER=97418592") in new stack
-- Executing Set("SIP/106-082435b0", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=trwW") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,8)
-- Executing Set("SIP/106-082435b0", "GROUP()=OUT_1") in new stack
-- Executing Macro("SIP/106-082435b0", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/106-082435b0", "user-callerid: Totalhearing 6568845638") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack
-- Executing Set("SIP/106-082435b0", "AMPUSER=106") in new stack
-- Executing Set("SIP/106-082435b0", "AMPUSERCIDNAME=Remote1") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack
-- Executing Set("SIP/106-082435b0", "CALLERID(all)=Remote1 <106>") in new stack
-- Executing Set("SIP/106-082435b0", "REALCALLERIDNUM=106") in new stack
-- Executing NoOp("SIP/106-082435b0", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/106-082435b0", "Using CallerID "Remote1" <106>") in new stack
-- Executing Macro("SIP/106-082435b0", "record-enable|106|OUT") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/106-082435b0", "recordingcheck|20070406-144312|1175841791.22") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070406-144312|1175841791.22: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/106-082435b0", "No recording needed") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?skipoutcid") in new stack
-- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=TrwW") in new stack
-- Executing Macro("SIP/106-082435b0", "outbound-callerid|1") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/106-082435b0", "USEROUTCID=") in new stack
-- Executing Set("SIP/106-082435b0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/106-082435b0", "TRUNKOUTCID="Totalhearing" <6568845638>") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/106-082435b0", "0?usercid") in new stack
-- Executing Set("SIP/106-082435b0", "CALLERID(all)="Totalhearing" <6568845638>") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/106-082435b0", "CallerID set to "Totalhearing" <6568845638>") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?nomax") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/106-082435b0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 6XXXXXXX
> fixlocalprefix: Using pattern 9XXXXXXX
== fixlocalprefix: Dialpattern 9XXXXXXX matched. 97418592 -> 97418592
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/106-082435b0", "OUTNUM=97418592") in new stack
-- Executing Set("SIP/106-082435b0", "custom=ZAP/3") in new stack
-- Executing GotoIf("SIP/106-082435b0", "0?customtrunk") in new stack
-- Executing Dial("SIP/106-082435b0", "ZAP/3/97418592|300|TrwW") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/106-082435b0", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/106-082435b0", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/106-082435b0", "outisbusy|") in new stack
-- Executing Playback("SIP/106-082435b0", "all-circuits-busy-now|noanswer") in new stack
We're at 192.168.0.102 port 10042
Video is at 192.168.0.102 port 10014
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 218.212.116.89:60919:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a800910000004d4615ec1c00006fba0000009d;received=218.212.116.89;rport=60919
From: "unknown"<sip:106@pt3>;tag=5205131213109
To: <sip:97418592@pt3>;tag=as2afdcbc1
Call-ID: 8CCD023D-A4FA-4887-9A8F-AAAD9CDC9B93@192.168.0.145
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:97418592@192.168.0.102>
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 3990 3990 IN IP4 192.168.0.102
s=session
c=IN IP4 192.168.0.102
t=0 0
m=audio 10042 RTP/AVP 0 8 3 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/106-082435b0", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
pt3*CLI>
<-- SIP read from 192.168.0.9:5060:
REGISTER sip:PT3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKt0RQhwEmSCeedK7s
Max-Forwards: 70
User-Agent: PA168S
From: "101" <sip:101@PT3>;tag=vp3ryfGbJOXpILO4
To: "101" <sip:101@PT3>
Call-ID: cMVIZ3m56skNefSo@192.168.0.9
CSeq: 19512 REGISTER
Contact: <sip:101@192.168.0.9:5060>
Expires: 60
Content-Length: 0
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.9 : 5060 (NAT)
Transmitting (no NAT) to 192.168.0.9:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKt0RQhwEmSCeedK7s;received=192.168.0.9
From: "101" <sip:101@PT3>;tag=vp3ryfGbJOXpILO4
To: "101" <sip:101@PT3>
Call-ID: cMVIZ3m56skNefSo@192.168.0.9
CSeq: 19512 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:101@192.168.0.102>
Content-Length: 0
---
Transmitting (no NAT) to 192.168.0.9:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKt0RQhwEmSCeedK7s;received=192.168.0.9
From: "101" <sip:101@PT3>;tag=vp3ryfGbJOXpILO4
To: "101" <sip:101@PT3>;tag=as19b83778
Call-ID: cMVIZ3m56skNefSo@192.168.0.9
CSeq: 19512 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40b1fd32"
Content-Length: 0
--3*CLI>
Scheduling destruction of call 'cMVIZ3m56skNefSo@192.168.0.9' in 15000 ms
pt3*CLI>
<-- SIP read from 218.212.116.89:60919:
OPTIONS sip:banshing.dyndns.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.145;rport;branch=z9hG4bKc0a80091000000104615eba0000058c400000084
Content-Length: 0
Call-ID: B2ADAA9F-7436-4D5C-97CC-6346281B3645@192.168.0.145
CSeq: 22 OPTIONS
From: <sip:106@pt3>;tag=5192726517924
Max-Forwards: 70
To: <sip:banshing.dyndns.org:5060>
--- (8 headers 0 lines) ---
Looking for s in from-sip-external (domain banshing.dyndns.org)
Transmitting (NAT) to 218.212.116.89:60919:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a80091000000104615eba0000058c400000084;received=218.212.116.89;rport=60919
From: <sip:106@pt3>;tag=5192726517924
To: <sip:banshing.dyndns.org:5060>;tag=as596812f0
Call-ID: B2ADAA9F-7436-4D5C-97CC-6346281B3645@192.168.0.145
CSeq: 22 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.102>
Accept: application/sdp
Content-Length: 0
---
Destroying call 'B2ADAA9F-7436-4D5C-97CC-6346281B3645@192.168.0.145'
-- Executing Macro("SIP/106-082435b0", "hangupcall") in new stack
-- Executing ResetCDR("SIP/106-082435b0", "w") in new stack
-- Executing NoCDR("SIP/106-082435b0", "") in new stack
-- Executing GotoIf("SIP/106-082435b0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/106-082435b0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/106-082435b0", "5") in new stack
pt3*CLI>
<-- SIP read from 192.168.0.9:5060:
REGISTER sip:PT3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKpLWU0Wgovz2JtlIl
Max-Forwards: 70
User-Agent: PA168S
From: "101" <sip:101@PT3>;tag=4QSjH2RsTV3WIFKS
To: "101" <sip:101@PT3>
Call-ID: cMVIZ3m56skNefSo@192.168.0.9
CSeq: 19513 REGISTER
Contact: <sip:101@192.168.0.9:5060>
Expires: 60
Authorization: Digest username="101", realm="asterisk", nonce="40b1fd32", uri="sip:PT3", response="1765a937ced98e75d4bedefb7585e04f", algorithm=MD5
Content-Length: 0
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.9 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.9:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKpLWU0Wgovz2JtlIl;received=192.168.0.9
From: "101" <sip:101@PT3>;tag=4QSjH2RsTV3WIFKS
To: "101" <sip:101@PT3>
Call-ID: cMVIZ3m56skNefSo@192.168.0.9
CSeq: 19513 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontact: <sip:101@192.168.0.102>
Content-Length: 0
---
Transmitting (no NAT) to 192.168.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKpLWU0Wgovz2JtlIl;received=192.168.0.9
From: "101" <sip:101@PT3>;tag=4QSjH2RsTV3WIFKS
To: "101" <sip:101@PT3>;tag=as19b83778
Call-ID: cMVIZ3m56skNefSo@192.168.0.9
CSeq: 19513 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:101@192.168.0.9:5060>;expires=60
Date: Fri, 06 Apr 2007 06:41:08 GMT
Content-Length: 0
---
Scheduling destruction of call 'cMVIZ3m56skNefSo@192.168.0.9' in 15000 ms
-- Executing Hangup("SIP/106-082435b0", "") in new stack
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/106-082435b0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/106-082435b0' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/106-082435b0'
Scheduling destruction of call '69AE3CC2-1D8B-48B2-9B79-12B7900E3C90@192.168.0.145' in 32000 ms
Reliably Transmitting (NAT) to 218.212.116.89:60919:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a80091000000454615eb9c0000127000000081;received=218.212.116.89;rport=60919
From: "unknown"<sip:106@pt3>;tag=5192321815974
To: <sip:97418592@pt3>;tag=as60f2a70c
Call-ID: 69AE3CC2-1D8B-48B2-9B79-12B7900E3C90@192.168.0.145
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:97418592@192.168.0.102>
Content-Length: 0
---
pt3*CLI>
<-- SIP read from 218.212.116.89:60919:
ACK sip:97418592@pt3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.145;rport;branch=z9hG4bKc0a80091000000454615eb9c0000127000000081
Content-Length: 0
Call-ID: 69AE3CC2-1D8B-48B2-9B79-12B7900E3C90@192.168.0.145
CSeq: 2 ACK
From: "unknown"<sip:106@pt3>;tag=5192321815974
Max-Forwards: 70
To: <sip:97418592@pt3>;tag=as60f2a70c
User-Agent: SJphone/1.60.289a (SJ Labs)
--- (9 headers 0 lines) ---
12 headers, 3 lines
Reliably Transmitting (no NAT) to 192.168.0.9:5060:
NOTIFY sip:101@192.168.0.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK475a80db;rport
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as65b628ed
To: <sip:101@192.168.0.9:5060>
Contact: <sip:asterisk@192.168.0.102>
Call-ID: 346ac58824c1e7b9381e8d8408f309a8@192.168.0.102
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 95
Messages-Waiting: yes
Message-Account: sip:asterisk@192.168.0.102
Voice-Message: 13/0 (0/0)
---
Scheduling destruction of call '346ac58824c1e7b9381e8d8408f309a8@192.168.0.102' in 15000 ms
pt3*CLI>
<-- SIP read from 192.168.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK475a80db;rport
Call-ID: 346ac58824c1e7b9381e8d8408f309a8@192.168.0.102
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as65b628ed
To: <sip:101@192.168.0.9:5060>
Contact: <sip:101@192.168.0.9:5060>
Content-Length: 0