rendered paste bodyCSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 57i/2.6.0.1007
Supported: timer, 100rel, replaces, path
Content-Length: 0
<------------->
[Sep 27 11:10:33] VERBOSE[8938] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 27 11:10:33] VERBOSE[8938] chan_sip.c: Really destroying SIP dialog '0253549d0891d9215a01e8f67e5407e4@192.168.0.246' Method: OPTIONS
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:192.168.2.43:2060 --->
<------------->
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:67.53.0.131:52837 --->
INVITE sip:4655@demo.ipiphony.com SIP/2.0
Via: SIP/2.0/UDP 192.168.155.109:40568;branch=z9hG4bK-d8754z-f03d6d5610070bae-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:fakename@192.168.155.109:40568>
To: <sip:4655@demo.ipiphony.com>
From: <sip:fakename@74.222.60.237>;tag=a72fb939
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 418
v=0
o=- 12961613363519143 1 IN IP4 192.168.155.109
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.155.109
t=0 0
a=ice-ufrag:f8b9ad
a=ice-pwd:0555ca4f9f00ba8e7b564a628f3f7ffd
m=audio 62470 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.155.109 62470 typ host
a=candidate:1 2 UDP 659134 192.168.155.109 62471 typ host
<------------->
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: --- (13 headers 14 lines) ---
[Sep 27 11:10:39] VERBOSE[8938] netsock.c: == Using SIP RTP TOS bits 184
[Sep 27 11:10:39] VERBOSE[8938] netsock.c: == Using SIP RTP CoS mark 5
[Sep 27 11:10:39] VERBOSE[8938] netsock.c: == Using SIP VRTP TOS bits 136
[Sep 27 11:10:39] VERBOSE[8938] netsock.c: == Using SIP VRTP CoS mark 6
[Sep 27 11:10:39] VERBOSE[8938] netsock.c: == Using UDPTL TOS bits 184
[Sep 27 11:10:39] VERBOSE[8938] netsock.c: == Using UDPTL CoS mark 5
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Sending to 67.53.0.131 : 52837 (NAT)
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Using INVITE request as basis request - NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: No matching peer for 'fakename' from '67.53.0.131:52837'
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Found RTP audio format 107
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Found RTP audio format 0
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Found RTP audio format 8
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Found RTP audio format 101
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Found audio description format BV32 for ID 107
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Found audio description format telephone-event for ID 101
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Capabilities: us - 0x3c0104 (ulaw|g729|h261|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Peer audio RTP is at port 192.168.155.109:62470
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Peer doesn't provide video
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: Looking for 4655 in from-trunk (domain demo.ipiphony.com)
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: list_route: hop: <sip:fakename@192.168.155.109:40568>
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c:
<--- Transmitting (NAT) to 67.53.0.131:52837 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.155.109:40568;branch=z9hG4bK-d8754z-f03d6d5610070bae-1---d8754z-;received=67.53.0.131;rport=52837
From: <sip:fakename@74.222.60.237>;tag=a72fb939
To: <sip:4655@demo.ipiphony.com>
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4655@74.222.60.237>
Content-Length: 0
<------------>
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [4655@from-trunk:1] GotoIf("SIP/74.222.60.237-00000002", "1?ext-local,4655,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (ext-local,4655,1)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [4655@ext-local:1] Macro("SIP/74.222.60.237-00000002", "exten-vm,4655,4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/74.222.60.237-00000002", "user-callerid,") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/74.222.60.237-00000002", "AMPUSER=fakename") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/74.222.60.237-00000002", "0?report") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/74.222.60.237-00000002", "1?Set(REALCALLERIDNUM=fakename)") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/74.222.60.237-00000002", "AMPUSER=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/74.222.60.237-00000002", "AMPUSERCIDNAME=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/74.222.60.237-00000002", "1?report") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-user-callerid,s,9)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:9] GotoIf("SIP/74.222.60.237-00000002", "0?continue") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:10] Set("SIP/74.222.60.237-00000002", "__TTL=64") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/74.222.60.237-00000002", "1?continue") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-user-callerid,s,18)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-user-callerid:18] NoOp("SIP/74.222.60.237-00000002", "Using CallerID "" <fakename>") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/74.222.60.237-00000002", "RingGroupMethod=none") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/74.222.60.237-00000002", "VMBOX=4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/74.222.60.237-00000002", "__EXTTOCALL=4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/74.222.60.237-00000002", "CFUEXT=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/74.222.60.237-00000002", "CFBEXT=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/74.222.60.237-00000002", "RT=15") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/74.222.60.237-00000002", "record-enable,4655,IN") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/74.222.60.237-00000002", "1?check") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-record-enable,s,4)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/74.222.60.237-00000002", "0?MacroExit()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/74.222.60.237-00000002", "0?Group:OUT") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-record-enable,s,15)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/74.222.60.237-00000002", "1?IN") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-record-enable,s,20)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/74.222.60.237-00000002", "1?MacroExit()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/74.222.60.237-00000002", "dial-one,15,trT,4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:1] Set("SIP/74.222.60.237-00000002", "DEXTEN=4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:2] Set("SIP/74.222.60.237-00000002", "DIALSTATUS_CW=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:3] GosubIf("SIP/74.222.60.237-00000002", "0?screen,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:4] GosubIf("SIP/74.222.60.237-00000002", "0?cf,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:5] GotoIf("SIP/74.222.60.237-00000002", "1?skip1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,s,8)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:8] GotoIf("SIP/74.222.60.237-00000002", "0?nodial") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:9] GotoIf("SIP/74.222.60.237-00000002", "0?continue") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:10] Set("SIP/74.222.60.237-00000002", "EXTHASCW=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:11] GotoIf("SIP/74.222.60.237-00000002", "1?next1:cwinusebusy") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,s,12)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:12] GotoIf("SIP/74.222.60.237-00000002", "0?docfu:skip3") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,s,16)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:16] GotoIf("SIP/74.222.60.237-00000002", "1?next2:continue") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,s,17)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:17] GotoIf("SIP/74.222.60.237-00000002", "1?continue") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,s,25)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:25] GotoIf("SIP/74.222.60.237-00000002", "0?nodial") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:26] GosubIf("SIP/74.222.60.237-00000002", "1?dstring,1:dlocal,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:1] Set("SIP/74.222.60.237-00000002", "DSTRING=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:2] Set("SIP/74.222.60.237-00000002", "DEVICES=4655&46551") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/74.222.60.237-00000002", "0?Return()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:4] Set("SIP/74.222.60.237-00000002", "LOOPCNT=2") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:5] Set("SIP/74.222.60.237-00000002", "ITER=1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:6] Set("SIP/74.222.60.237-00000002", "THISDIAL=SIP/4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:7] GosubIf("SIP/74.222.60.237-00000002", "1?zap2dahdi,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/74.222.60.237-00000002", "0?Return()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/74.222.60.237-00000002", "NEWDIAL=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/74.222.60.237-00000002", "LOOPCNT2=1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/74.222.60.237-00000002", "ITER2=1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/74.222.60.237-00000002", "THISPART2=SIP/4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/74.222.60.237-00000002", "0?Set(THISPART2=DAHDI/4655)") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/74.222.60.237-00000002", "NEWDIAL=SIP/4655&") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/74.222.60.237-00000002", "ITER2=2") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/74.222.60.237-00000002", "0?begin2") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/74.222.60.237-00000002", "THISDIAL=SIP/4655") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/74.222.60.237-00000002", "") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:8] Set("SIP/74.222.60.237-00000002", "DSTRING=SIP/4655&") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:9] Set("SIP/74.222.60.237-00000002", "ITER=2") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:10] GotoIf("SIP/74.222.60.237-00000002", "1?begin") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,dstring,6)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:6] Set("SIP/74.222.60.237-00000002", "THISDIAL=SIP/46551") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:7] GosubIf("SIP/74.222.60.237-00000002", "1?zap2dahdi,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/74.222.60.237-00000002", "0?Return()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/74.222.60.237-00000002", "NEWDIAL=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/74.222.60.237-00000002", "LOOPCNT2=1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/74.222.60.237-00000002", "ITER2=1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/74.222.60.237-00000002", "THISPART2=SIP/46551") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/74.222.60.237-00000002", "0?Set(THISPART2=DAHDI/46551)") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/74.222.60.237-00000002", "NEWDIAL=SIP/46551&") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/74.222.60.237-00000002", "ITER2=2") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/74.222.60.237-00000002", "0?begin2") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/74.222.60.237-00000002", "THISDIAL=SIP/46551") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/74.222.60.237-00000002", "") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:8] Set("SIP/74.222.60.237-00000002", "DSTRING=SIP/4655&SIP/46551&") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:9] Set("SIP/74.222.60.237-00000002", "ITER=3") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:10] GotoIf("SIP/74.222.60.237-00000002", "0?begin") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:11] Set("SIP/74.222.60.237-00000002", "DSTRING=SIP/4655&SIP/46551") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [dstring@macro-dial-one:12] Return("SIP/74.222.60.237-00000002", "") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:27] GotoIf("SIP/74.222.60.237-00000002", "0?nodial") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:28] GotoIf("SIP/74.222.60.237-00000002", "1?skiptrace") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Goto (macro-dial-one,s,30)
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:30] Set("SIP/74.222.60.237-00000002", "D_OPTIONS=trT") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:31] ExecIf("SIP/74.222.60.237-00000002", "0?SIPAddHeader(Alert-Info: )") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:32] ExecIf("SIP/74.222.60.237-00000002", "0?SIPAddHeader()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:33] ExecIf("SIP/74.222.60.237-00000002", "0?SetMusicOnHold()") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:34] GosubIf("SIP/74.222.60.237-00000002", "0?qwait,1") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:35] Set("SIP/74.222.60.237-00000002", "__CWIGNORE=") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:36] Set("SIP/74.222.60.237-00000002", "__KEEPCID=TRUE") in new stack
[Sep 27 11:10:39] VERBOSE[9222] pbx.c: -- Executing [s@macro-dial-one:37] Dial("SIP/74.222.60.237-00000002", "SIP/4655&SIP/46551,15,trT") in new stack
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP RTP TOS bits 184
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP RTP CoS mark 5
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP VRTP TOS bits 136
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP VRTP CoS mark 6
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using UDPTL TOS bits 184
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using UDPTL CoS mark 5
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Audio is at 192.168.0.246 port 10012
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Video is at 192.168.0.246 port 10046
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Adding video codec 0x40000 (h261) to SDP
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Adding video codec 0x80000 (h263) to SDP
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Adding video codec 0x200000 (h264) to SDP
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Reliably Transmitting (NAT) to 192.168.155.118:5060:
INVITE sip:4655@192.168.155.118:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.246:5060;branch=z9hG4bK21e70676;rport
Max-Forwards: 70
From: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
To: <sip:4655@192.168.155.118:5060>
Contact: <sip:fakename@192.168.0.246>
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Tue, 27 Sep 2011 16:10:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 396
v=0
o=root 327667647 327667647 IN IP4 192.168.0.246
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.0.246
b=CT:1024
t=0 0
m=audio 10012 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 10046 RTP/AVP 31 34 98 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
[Sep 27 11:10:39] VERBOSE[9222] app_dial.c: -- Called 4655
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP RTP TOS bits 184
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP RTP CoS mark 5
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP VRTP TOS bits 136
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using SIP VRTP CoS mark 6
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using UDPTL TOS bits 184
[Sep 27 11:10:39] VERBOSE[9222] netsock.c: == Using UDPTL CoS mark 5
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c: Really destroying SIP dialog '0ac9fbf90dd58c284b92689a08f21903@127.0.1.1' Method: INVITE
[Sep 27 11:10:39] WARNING[9222] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c:
<--- Transmitting (NAT) to 67.53.0.131:52837 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.155.109:40568;branch=z9hG4bK-d8754z-f03d6d5610070bae-1---d8754z-;received=67.53.0.131;rport=52837
From: <sip:fakename@74.222.60.237>;tag=a72fb939
To: <sip:4655@demo.ipiphony.com>;tag=as09fdbdc5
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4655@74.222.60.237>
Content-Length: 0
<------------>
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:192.168.155.118:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.246:5060;branch=z9hG4bK21e70676;rport
From: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
To: "4655" <sip:4655@192.168.155.118:5060>;tag=AA1DA289-8365F42A
CSeq: 102 INVITE
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
Contact: <sip:4655@192.168.155.118:5060>
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en
Content-Length: 0
<------------->
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:192.168.155.118:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.246:5060;branch=z9hG4bK21e70676;rport
From: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
To: "4655" <sip:4655@192.168.155.118:5060>;tag=AA1DA289-8365F42A
CSeq: 102 INVITE
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
Contact: <sip:4655@192.168.155.118:5060>
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Allow-Events: talk,hold,conference
Accept-Language: en
Content-Length: 0
<------------->
[Sep 27 11:10:39] VERBOSE[8938] chan_sip.c: --- (11 headers 0 lines) ---
[Sep 27 11:10:39] VERBOSE[9222] app_dial.c: -- SIP/4655-00000003 is ringing
[Sep 27 11:10:39] VERBOSE[9222] chan_sip.c:
<--- Transmitting (NAT) to 67.53.0.131:52837 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.155.109:40568;branch=z9hG4bK-d8754z-f03d6d5610070bae-1---d8754z-;received=67.53.0.131;rport=52837
From: <sip:fakename@74.222.60.237>;tag=a72fb939
To: <sip:4655@demo.ipiphony.com>;tag=as09fdbdc5
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4655@74.222.60.237>
Content-Length: 0
<------------>
[Sep 27 11:10:40] VERBOSE[8938] chan_sip.c: Really destroying SIP dialog '0fb9c11605676be959dbb58701443971@62.135.165.12' Method: OPTIONS
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:192.168.155.118:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.246:5060;branch=z9hG4bK21e70676;rport
From: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
To: "4655" <sip:4655@192.168.155.118:5060>;tag=AA1DA289-8365F42A
CSeq: 102 INVITE
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
Contact: <sip:4655@192.168.155.118:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en
Content-Type: application/sdp
Content-Length: 361
v=0
o=- 1317116795 1317116795 IN IP4 192.168.155.118
s=Polycom IP Phone
c=IN IP4 192.168.155.118
t=0 0
a=sendrecv
m=audio 2264 RTP/AVP 0 127
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
m=video 0 RTP/AVP 31 34 98 99
a=sendrecv
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
<------------->
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: --- (13 headers 16 lines) ---
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found RTP audio format 0
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found RTP audio format 127
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found audio description format PCMU for ID 0
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found audio description format telephone-event for ID 127
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found RTP video format 31
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found RTP video format 34
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found RTP video format 98
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found RTP video format 99
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found video description format H261 for ID 31
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found video description format H263 for ID 34
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found video description format h263-1998 for ID 98
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Found video description format H264 for ID 99
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Capabilities: us - 0x3c0104 (ulaw|g729|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x3c0000 (h261|h263|h263p|h264)/text=0x0 (nothing), combined - 0x3c0004 (ulaw|h261|h263|h263p|h264)
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Peer audio RTP is at port 192.168.155.118:2264
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Peer doesn't provide video
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: list_route: hop: <sip:4655@192.168.155.118:5060>
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: set_destination: Parsing <sip:4655@192.168.155.118:5060> for address/port to send to
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: set_destination: set destination to 192.168.155.118, port 5060
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: Transmitting (NAT) to 192.168.155.118:5060:
ACK sip:4655@192.168.155.118:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.246:5060;branch=z9hG4bK362a5f15;rport
Max-Forwards: 70
From: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
To: <sip:4655@192.168.155.118:5060>;tag=AA1DA289-8365F42A
Contact: <sip:fakename@192.168.0.246>
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.18
Content-Length: 0
---
[Sep 27 11:10:41] VERBOSE[9222] app_dial.c: -- SIP/4655-00000003 answered SIP/74.222.60.237-00000002
[Sep 27 11:10:41] VERBOSE[9222] chan_sip.c: Audio is at 74.222.60.237 port 10080
[Sep 27 11:10:41] VERBOSE[9222] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 27 11:10:41] VERBOSE[9222] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 27 11:10:41] VERBOSE[9222] chan_sip.c:
<--- Reliably Transmitting (NAT) to 67.53.0.131:52837 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.155.109:40568;branch=z9hG4bK-d8754z-f03d6d5610070bae-1---d8754z-;received=67.53.0.131;rport=52837
From: <sip:fakename@74.222.60.237>;tag=a72fb939
To: <sip:4655@demo.ipiphony.com>;tag=as09fdbdc5
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4655@74.222.60.237>
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 1281197182 1281197182 IN IP4 74.222.60.237
s=Asterisk PBX 1.6.2.18
c=IN IP4 74.222.60.237
t=0 0
m=audio 10080 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:67.53.0.131:52837 --->
ACK sip:4655@74.222.60.237 SIP/2.0
Via: SIP/2.0/UDP 192.168.155.109:40568;branch=z9hG4bK-d8754z-41556be55dcb2bf3-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:fakename@192.168.155.109:40568>
To: <sip:4655@demo.ipiphony.com>;tag=as09fdbdc5
From: <sip:fakename@74.222.60.237>;tag=a72fb939
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 1 ACK
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0
<------------->
[Sep 27 11:10:41] VERBOSE[8938] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:192.168.155.118:5060 --->
BYE sip:fakename@192.168.0.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.155.118:5060;branch=z9hG4bK4f1771d352D173C4
From: "4655" <sip:4655@192.168.155.118:5060>;tag=AA1DA289-8365F42A
To: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
CSeq: 1 BYE
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
Contact: <sip:4655@192.168.155.118:5060>
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
<------------->
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c: --- (11 headers 0 lines) ---
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c: Sending to 192.168.155.118 : 5060 (NAT)
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c:
<--- Transmitting (NAT) to 192.168.155.118:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.155.118:5060;branch=z9hG4bK4f1771d352D173C4;received=192.168.155.118
From: "4655" <sip:4655@192.168.155.118:5060>;tag=AA1DA289-8365F42A
To: "fakename" <sip:fakename@192.168.0.246>;tag=as151e8c1f
Call-ID: 265547396a29103c101f241f3332679a@192.168.0.246
CSeq: 1 BYE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [h@macro-dial-one:1] Macro("SIP/74.222.60.237-00000002", "hangupcall,") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/74.222.60.237-00000002", "1?skiprg") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Goto (macro-hangupcall,s,4)
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/74.222.60.237-00000002", "1?skipblkvm") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Goto (macro-hangupcall,s,7)
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/74.222.60.237-00000002", "1?theend") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Goto (macro-hangupcall,s,9)
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/74.222.60.237-00000002", "") in new stack
[Sep 27 11:10:42] VERBOSE[9222] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/74.222.60.237-00000002' in macro 'hangupcall'
[Sep 27 11:10:42] VERBOSE[9222] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/74.222.60.237-00000002'
[Sep 27 11:10:42] VERBOSE[9222] app_macro.c: == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/74.222.60.237-00000002' in macro 'dial-one'
[Sep 27 11:10:42] VERBOSE[9222] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/74.222.60.237-00000002' in macro 'exten-vm'
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: == Spawn extension (ext-local, 4655, 1) exited non-zero on 'SIP/74.222.60.237-00000002'
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [h@ext-local:1] Macro("SIP/74.222.60.237-00000002", "hangupcall,") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/74.222.60.237-00000002", "1?skiprg") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Goto (macro-hangupcall,s,4)
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/74.222.60.237-00000002", "1?skipblkvm") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Goto (macro-hangupcall,s,7)
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/74.222.60.237-00000002", "1?theend") in new stack
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Goto (macro-hangupcall,s,9)
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/74.222.60.237-00000002", "") in new stack
[Sep 27 11:10:42] VERBOSE[9222] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/74.222.60.237-00000002' in macro 'hangupcall'
[Sep 27 11:10:42] VERBOSE[9222] pbx.c: == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/74.222.60.237-00000002'
[Sep 27 11:10:42] VERBOSE[9222] chan_sip.c: Scheduling destruction of SIP dialog 'NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.' in 32000 ms (Method: ACK)
[Sep 27 11:10:42] VERBOSE[9222] chan_sip.c: set_destination: Parsing <sip:fakename@192.168.155.109:40568> for address/port to send to
[Sep 27 11:10:42] VERBOSE[9222] chan_sip.c: set_destination: set destination to 192.168.155.109, port 40568
[Sep 27 11:10:42] VERBOSE[9222] chan_sip.c: Reliably Transmitting (NAT) to 67.53.0.131:52837:
BYE sip:fakename@192.168.155.109:40568 SIP/2.0
Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK4bf96382;rport
Max-Forwards: 70
From: <sip:4655@demo.ipiphony.com>;tag=as09fdbdc5
To: <sip:fakename@74.222.60.237>;tag=a72fb939
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.18
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c:
<--- SIP read from UDP:67.53.0.131:52837 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK4bf96382;rport=5060
Contact: <sip:fakename@192.168.155.109:40568>
To: <sip:fakename@74.222.60.237>;tag=a72fb939
From: <sip:4655@demo.ipiphony.com>;tag=as09fdbdc5
Call-ID: NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.
CSeq: 102 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0
<------------->
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c: Really destroying SIP dialog '265547396a29103c101f241f3332679a@192.168.0.246' Method: BYE
[Sep 27 11:10:42] VERBOSE[8938] chan_sip.c: Really destroying SIP dialog 'NjQyYmY1ZDNlOGEzOTZhM2ZmYzcyZjYzNjZmNmNjYzI.' Method: ACK