Вход | Регистрация
Вы здесь: Главная / Форум / Главный форум по Asterisk / TrixBox, Elastix, FreePbx / 15 секунд и тишина ( Elastix)

15 секунд и тишина ( Elastix)

Логи и подробносьти внутри
1 2>
n00b
Сообщений: 32

15 секунд и тишина ( Elastix)

Собственно было 3 офиса объединенные тунелями( назовем их 1, 4 и 5 )
В офисе 1 стоял еластикс и телефоны, в 4 и 5 только телефоны.
Все работало.

Еластик перекочевал в офис 5.

Теперь при звонках на офис 1 или с офиса 1 разговор длиться 15 секунд, потом идет обрыв.

[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [96@from-internal:1] Macro("SIP/90-0000012a", "exten-vm,novm,96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/90-0000012a", "user-callerid,") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/90-0000012a", "AMPUSER=90") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/90-0000012a", "0?report") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/90-0000012a", "1?Set(REALCALLERIDNUM=90)") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/90-0000012a", "AMPUSER=90") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/90-0000012a", "AMPUSERCIDNAME=90") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/90-0000012a", "0?report") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/90-0000012a", "AMPUSERCID=90") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/90-0000012a", "CALLERID(all)="90" <90>") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/90-0000012a", "0?Set(CHANNEL(language)=)") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/90-0000012a", "0?continue") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/90-0000012a", "__TTL=64") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/90-0000012a", "1?continue") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-user-callerid,s,19)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/90-0000012a", "Using CallerID "90" <90>") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/90-0000012a", "RingGroupMethod=none") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/90-0000012a", "VMBOX=novm") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/90-0000012a", "__EXTTOCALL=96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/90-0000012a", "CFUEXT=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/90-0000012a", "CFBEXT=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/90-0000012a", "RT=""") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/90-0000012a", "record-enable,96,IN") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/90-0000012a", "1?check") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-record-enable,s,4)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/90-0000012a", "0?MacroExit()") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/90-0000012a", "0?Group:OUT") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-record-enable,s,15)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/90-0000012a", "1?IN") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-record-enable,s,20)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/90-0000012a", "1?MacroExit()") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/90-0000012a", "dial-one,,tr,96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:1] Set("SIP/90-0000012a", "DEXTEN=96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:2] Set("SIP/90-0000012a", "DIALSTATUS_CW=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:3] GosubIf("SIP/90-0000012a", "0?screen,1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:4] GosubIf("SIP/90-0000012a", "0?cf,1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:5] GotoIf("SIP/90-0000012a", "1?skip1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-dial-one,s,8)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:8] GotoIf("SIP/90-0000012a", "0?nodial") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:9] GotoIf("SIP/90-0000012a", "0?continue") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:10] Set("SIP/90-0000012a", "EXTHASCW=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:11] GotoIf("SIP/90-0000012a", "1?next1:cwinusebusy") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-dial-one,s,12)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:12] GotoIf("SIP/90-0000012a", "0?docfu:skip3") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-dial-one,s,16)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:16] GotoIf("SIP/90-0000012a", "1?next2:continue") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-dial-one,s,17)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:17] GotoIf("SIP/90-0000012a", "1?continue") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-dial-one,s,25)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:25] GotoIf("SIP/90-0000012a", "0?nodial") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:26] GosubIf("SIP/90-0000012a", "1?dstring,1:dlocal,1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:1] Set("SIP/90-0000012a", "DSTRING=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:2] Set("SIP/90-0000012a", "DEVICES=96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/90-0000012a", "0?Return()") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/90-0000012a", "0?Set(DEVICES=6)") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:5] Set("SIP/90-0000012a", "LOOPCNT=1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:6] Set("SIP/90-0000012a", "ITER=1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:7] Set("SIP/90-0000012a", "THISDIAL=SIP/96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/90-0000012a", "1?zap2dahdi,1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/90-0000012a", "0?Return()") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/90-0000012a", "NEWDIAL=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/90-0000012a", "LOOPCNT2=1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/90-0000012a", "ITER2=1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/90-0000012a", "THISPART2=SIP/96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/90-0000012a", "0?Set(THISPART2=DAHDI/96)") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/90-0000012a", "NEWDIAL=SIP/96&") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/90-0000012a", "ITER2=2") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/90-0000012a", "0?begin2") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/90-0000012a", "THISDIAL=SIP/96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/90-0000012a", "") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:9] Set("SIP/90-0000012a", "DSTRING=SIP/96&") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:10] Set("SIP/90-0000012a", "ITER=2") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/90-0000012a", "0?begin") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:12] Set("SIP/90-0000012a", "DSTRING=SIP/96") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [dstring@macro-dial-one:13] Return("SIP/90-0000012a", "") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:27] GotoIf("SIP/90-0000012a", "0?nodial") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:28] GotoIf("SIP/90-0000012a", "1?skiptrace") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Goto (macro-dial-one,s,30)
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:30] Set("SIP/90-0000012a", "D_OPTIONS=tr") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:31] ExecIf("SIP/90-0000012a", "0?SIPAddHeader(Alert-Info: )") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:32] ExecIf("SIP/90-0000012a", "0?SIPAddHeader()") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:33] ExecIf("SIP/90-0000012a", "0?SetMusicOnHold()") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:34] GosubIf("SIP/90-0000012a", "0?qwait,1") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:35] Set("SIP/90-0000012a", "__CWIGNORE=") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:36] Set("SIP/90-0000012a", "__KEEPCID=TRUE") in new stack
[Oct 11 16:24:13] VERBOSE[26041] pbx.c: -- Executing [s@macro-dial-one:37] Dial("SIP/90-0000012a", "SIP/96,,tr") in new stack
[Oct 11 16:24:13] VERBOSE[26041] netsock.c: == Using SIP RTP TOS bits 184
[Oct 11 16:24:13] VERBOSE[26041] netsock.c: == Using SIP RTP CoS mark 5
[Oct 11 16:24:13] VERBOSE[26041] app_dial.c: -- Called 96
[Oct 11 16:24:13] VERBOSE[26041] app_dial.c: -- SIP/96-0000012b is ringing
[Oct 11 16:24:16] VERBOSE[26041] app_dial.c: -- SIP/96-0000012b answered SIP/90-0000012a
[Oct 11 16:24:26] WARNING[2713] chan_sip.c: Maximum retries exceeded on transmission 38871f72-c692dec4@192.168.1.1 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct 11 16:24:26] WARNING[2713] chan_sip.c: Hanging up call 38871f72-c692dec4@192.168.1.1 - no reply to our critical packet (see doc/sip-retransmit.txt).
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [h@macro-dial-one:1] Macro("SIP/90-0000012a", "hangupcall,") in new stack
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/90-0000012a", "1?noautomon") in new stack
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Goto (macro-hangupcall,s,3)
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/90-0000012a", "TOUCH_MONITOR_OUTPUT=") in new stack
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/90-0000012a", "1?skiprg") in new stack
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Goto (macro-hangupcall,s,7)
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/90-0000012a", "1?skipblkvm") in new stack
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Goto (macro-hangupcall,s,10)
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/90-0000012a", "1?theend") in new stack
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Goto (macro-hangupcall,s,12)
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/90-0000012a", "") in new stack
[Oct 11 16:24:26] VERBOSE[26041] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/90-0000012a' in macro 'hangupcall'
[Oct 11 16:24:26] VERBOSE[26041] app_macro.c: == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/90-0000012a' in macro 'dial-one'
[Oct 11 16:24:26] VERBOSE[26041] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/90-0000012a' in macro 'exten-vm'
[Oct 11 16:24:26] VERBOSE[26041] pbx.c: == Spawn extension (from-internal, 96, 1) exited non-zero on 'SIP/90-0000012a'
[Oct 11 16:25:02] VERBOSE[2519] asterisk.c: -- Remote UNIX connection
[Oct 11 16:25:02] VERBOSE[26046] asterisk.c: -- Remote UNIX connection disconnected
[Oct 11 16:30:01] VERBOSE[2519] asterisk.c: -- Remote UNIX connection
[Oct 11 16:30:01] VERBOSE[26053] asterisk.c: -- Remote UNIX connection disconnected


[Oct 11 16:24:26] WARNING[2713] chan_sip.c: Maximum retries exceeded on transmission 38871f72-c692dec4@192.168.1.1 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct 11 16:24:26] WARNING[2713] chan_sip.c: Hanging up call 38871f72-c692dec4@192.168.1.1 - no reply to our critical packet (see doc/sip-retransmit.txt).

Вот эти 2 строчки постоянно повторяются.

Подскажите что и как ? А то директор нервничает, заказчик материт а я в инете ничего не могу нарыть по этому поводу.
2010-10-11 18:14

Evgen
Откуда: Саратов
Сообщений: 414

Re: 15 секунд и тишина ( Elastix)

sip debug поможет 100%.
+7(925)140-7438
2010-10-11 18:26

n00b
Сообщений: 32

Re: 15 секунд и тишина ( Elastix)

Evgen:

sip debug поможет 100%.
Я только этому буду рад, но если бы еще знать как мне он поможет. И что мне с этим делать?

Извините за сравнение, но это то же самое что сказать что осиновый листок мне поможет, но не говорить что с этим листком делать.
2010-10-12 10:31

svoy
Avatara of svoy
Откуда: Киев
Сообщений: 1096

Re: 15 секунд и тишина ( Elastix)

sip debug покажет сигнализацию между сторонами, походу какие то ответы не возвращаются.

15 cек и отбой похоже на проблемы с натом, который закрывает соединение по истечению таймаута
2010-10-12 10:48

n00b
Сообщений: 32

Re: 15 секунд и тишина ( Elastix)

Я попытался выполнить но вот что мне выдало.

No such command 'sip debug' (type 'core show help sip debug' for other possible commands)
2010-10-12 11:15

svoy
Avatara of svoy
Откуда: Киев
Сообщений: 1096

Re: 15 секунд и тишина ( Elastix)

sip set debug

лично мне удобнее пользоваться tcpdump для снифа и wireshark для анализа
2010-10-12 11:51

n00b
Сообщений: 32

Re: 15 секунд и тишина ( Elastix)

No such command 'sip set debug ' (type 'core show help sip set debug' for other possible commands)

2010-10-12 12:00

Evgen
Откуда: Саратов
Сообщений: 414

Re: 15 секунд и тишина ( Elastix)

sip set debug on
+7(925)140-7438
2010-10-12 12:39

svoy
Avatara of svoy
Откуда: Киев
Сообщений: 1096

Re: 15 секунд и тишина ( Elastix)

табом или на крайний случай гуглом пользоваться вера не позволяет?
2010-10-12 13:35

n00b
Сообщений: 32

Re: 15 секунд и тишина ( Elastix)

Про таб: я свебморды залез. Как вариант стоит путти, но с вебморды мне немного понятнее.
Гуглом пользуюсь.
Написано очень мало.
В частности лишь то что приходит какой-то пакет который обрывает линию или же что проблемы с натом.

<------------->
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '47d0d8e951899de548fe8329439afc04@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.176:60392 --->
INVITE sip:7777@192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-141b0051d171d30f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:666@192.168.1.176:60392>
To: "7777"<sip:7777@192.168.5.200>
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 267

v=0
o=- 1 2 IN IP4 192.168.1.176
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.176
t=0 0
m=audio 33206 RTP/AVP 107 0 8 101
a=alt:1 1 : vmEc19hk yY2mWdXY 192.168.1.176 33206
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: --- (12 headers 11 lines) ---
[Oct 12 13:10:34] VERBOSE[2713] netsock.c: == Using SIP RTP TOS bits 184
[Oct 12 13:10:34] VERBOSE[2713] netsock.c: == Using SIP RTP CoS mark 5
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.176 : 60392 (NAT)
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Using INVITE request as basis request - OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found peer '666' for '666' from 192.168.1.176:60392
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.176:60392 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-141b0051d171d30f-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as1111a848
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c22e053"
Content-Length: 0


<------------>
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.' in 11264 ms (Method: INVITE)
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.176:60392 --->
ACK sip:7777@192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-141b0051d171d30f-1---d8754z-;rport
To: "7777"<sip:7777@192.168.5.200>;tag=as1111a848
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 1 ACK
Content-Length: 0


<------------->
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: --- (7 headers 0 lines) ---
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.176:60392 --->
INVITE sip:7777@192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:666@192.168.1.176:60392>
To: "7777"<sip:7777@192.168.5.200>
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="666",realm="asterisk",nonce="3c22e053",uri="sip:7777@192.168.5.200",response="d8c7376e19ea59ef8059ea0e7789d3ba",algorithm=MD5
Content-Length: 267

v=0
o=- 1 2 IN IP4 192.168.1.176
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.176
t=0 0
m=audio 33206 RTP/AVP 107 0 8 101
a=alt:1 1 : vmEc19hk yY2mWdXY 192.168.1.176 33206
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: --- (13 headers 11 lines) ---
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.176 : 60392 (NAT)
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Using INVITE request as basis request - OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found peer '666' for '666' from 192.168.1.176:60392
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found RTP audio format 107
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found RTP audio format 0
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found RTP audio format 8
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found RTP audio format 101
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found audio description format BV32 for ID 107
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Peer audio RTP is at port 192.168.1.176:33206
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: Looking for 7777 in from-internal (domain 192.168.5.200)
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c: list_route: hop: <sip:666@192.168.1.176:60392>
[Oct 12 13:10:34] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.176:60392 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Length: 0


<------------>
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [7777@from-internal:1] Goto("SIP/666-00000162", "from-pstn,7777,1") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Goto (from-pstn,7777,1)
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [7777@from-pstn:1] NoOp("SIP/666-00000162", "Catch-All DID Match - Found 7777 - You probably want a DID for this.") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [7777@from-pstn:2] Goto("SIP/666-00000162", "ext-did,s,1") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Goto (ext-did,s,1)
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:1] Set("SIP/666-00000162", "__FROM_DID=s") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:2] Gosub("SIP/666-00000162", "app-blacklist-check,s,1") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@app-blacklist-check:1] GotoIf("SIP/666-00000162", "0?blacklisted") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@app-blacklist-check:2] Set("SIP/666-00000162", "CALLED_BLACKLIST=1") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@app-blacklist-check:3] Return("SIP/666-00000162", "") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:3] ExecIf("SIP/666-00000162", "0 ?Set(CALLERID(name)=666)") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:4] Set("SIP/666-00000162", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:5] Set("SIP/666-00000162", "CALLERPRES()=allowed_not_screened") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:6] Set("SIP/666-00000162", "FAX_DEST=from-did-direct^88^1") in new stack
[Oct 12 13:10:34] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:7] Answer("SIP/666-00000162", "") in new stack
[Oct 12 13:10:34] VERBOSE[29403] chan_sip.c: Audio is at 82.144.205.16 port 16924
[Oct 12 13:10:34] VERBOSE[29403] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 12 13:10:34] VERBOSE[29403] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 12 13:10:34] VERBOSE[29403] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.176:60392 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Oct 12 13:10:35] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:8] Wait("SIP/666-00000162", "2") in new stack
[Oct 12 13:10:35] VERBOSE[2713] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.176:60392:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 12 13:10:35] VERBOSE[2713] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.176:60392:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 12 13:10:35] VERBOSE[2713] chan_sip.c: Retransmitting #3 (NAT) to 78.141.179.70:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK30f76c2b;rport
Max-Forwards: 70
From: <sip:99051000113331@sip.skype.com>;tag=as3c2c3989
To: <sip:99051000113331@sip.skype.com>
Call-ID: 66e6d27578caf3e3161eefc547e1339e@193.200.173.1
CSeq: 495 REGISTER
User-Agent: Asterisk PBX 1.6.2.10
Authorization: Digest username="99051000113331", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com", nonce="4cb42add00017688f69421f5e656119d300014e781135ff8", response="56f0455b61cb433953a5e329d5bb1039"
Expires: 600
Contact: <sip:s@82.144.205.16>
Content-Length: 0


---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-73ae9d35;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003601 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="3efa84b6",uri="sip:192.168.5.200",algorithm=MD5,response="1b163dea980581f775045ed9e0c83556"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-73ae9d35;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003601 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:36] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-73ae9d35;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003601 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07b315a6", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.176:60392:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-523b1ab3;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003602 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="07b315a6",uri="sip:192.168.5.200",algorithm=MD5,response="c5b3f8f5e5d2d1e15e952e7abf3424db"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-523b1ab3;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003602 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK016aa282;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as62b0e5d7
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 3cfe040e5c7b248502b264870dc9b1dd@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-523b1ab3;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003602 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:36 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as62b0e5d7
Call-ID: 3cfe040e5c7b248502b264870dc9b1dd@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK016aa282;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '3cfe040e5c7b248502b264870dc9b1dd@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.5.93:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.93:5060;branch=z9hG4bK-15cc230c
From: "93" <sip:93@192.168.5.200>;tag=2ad53ac615d8c834o0
To: "93" <sip:93@192.168.5.200>
Call-ID: c2460617-9ff18cf0@192.168.5.93
CSeq: 42563 REGISTER
Max-Forwards: 70
Authorization: Digest username="93",realm="asterisk",nonce="1078f724",uri="sip:192.168.5.200",algorithm=MD5,response="9c854909abaf61f4832ce79de9791f15"
Contact: "93" <sip:93@192.168.5.93:5060>;expires=200
User-Agent: Linksys/SPA901-4.1.11(c)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER


<------------->
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: --- (12 headers 0 lines) ---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Sending to 192.168.5.93 : 5060 (NAT)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.5.93:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.93:5060;branch=z9hG4bK-15cc230c;received=192.168.5.93
From: "93" <sip:93@192.168.5.200>;tag=2ad53ac615d8c834o0
To: "93" <sip:93@192.168.5.200>
Call-ID: c2460617-9ff18cf0@192.168.5.93
CSeq: 42563 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.5.93:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.93:5060;branch=z9hG4bK-15cc230c;received=192.168.5.93
From: "93" <sip:93@192.168.5.200>;tag=2ad53ac615d8c834o0
To: "93" <sip:93@192.168.5.200>;tag=as29b9af25
Call-ID: c2460617-9ff18cf0@192.168.5.93
CSeq: 42563 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60e83232"
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'c2460617-9ff18cf0@192.168.5.93' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.5.93:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.93:5060;branch=z9hG4bK-e00b83f5
From: "93" <sip:93@192.168.5.200>;tag=2ad53ac615d8c834o0
To: "93" <sip:93@192.168.5.200>
Call-ID: c2460617-9ff18cf0@192.168.5.93
CSeq: 42564 REGISTER
Max-Forwards: 70
Authorization: Digest username="93",realm="asterisk",nonce="60e83232",uri="sip:192.168.5.200",algorithm=MD5,response="74545e03f2f0fffe5a1fa35b72b16e6a"
Contact: "93" <sip:93@192.168.5.93:5060>;expires=200
User-Agent: Linksys/SPA901-4.1.11(c)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER


<------------->
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: --- (12 headers 0 lines) ---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Sending to 192.168.5.93 : 5060 (NAT)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.5.93:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.93:5060;branch=z9hG4bK-e00b83f5;received=192.168.5.93
From: "93" <sip:93@192.168.5.200>;tag=2ad53ac615d8c834o0
To: "93" <sip:93@192.168.5.200>
Call-ID: c2460617-9ff18cf0@192.168.5.93
CSeq: 42564 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.93:5060:
OPTIONS sip:93@192.168.5.93:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK3b01293b;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.5.200>;tag=as576a14ab
To: <sip:93@192.168.5.93:5060>
Contact: <sip:Unknown@192.168.5.200>
Call-ID: 6ca0fca52d929ade18f0201f47d69697@192.168.5.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.5.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.93:5060;branch=z9hG4bK-e00b83f5;received=192.168.5.93
From: "93" <sip:93@192.168.5.200>;tag=2ad53ac615d8c834o0
To: "93" <sip:93@192.168.5.200>;tag=as29b9af25
Call-ID: c2460617-9ff18cf0@192.168.5.93
CSeq: 42564 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 200
Contact: <sip:93@192.168.5.93:5060>;expires=200
Date: Tue, 12 Oct 2010 10:10:36 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'c2460617-9ff18cf0@192.168.5.93' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.5.93:5060 --->
SIP/2.0 200 OK
To: <sip:93@192.168.5.93:5060>;tag=3162b3ae3b6b646ci0
From: "Unknown" <sip:Unknown@192.168.5.200>;tag=as576a14ab
Call-ID: 6ca0fca52d929ade18f0201f47d69697@192.168.5.200
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK3b01293b
Server: Linksys/SPA901-4.1.11(c)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER


<------------->
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: --- (9 headers 0 lines) ---
[Oct 12 13:10:36] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '6ca0fca52d929ade18f0201f47d69697@192.168.5.200' Method: OPTIONS
[Oct 12 13:10:37] VERBOSE[29403] pbx.c: -- Executing [s@ext-did:9] Goto("SIP/666-00000162", "app-announcement-8,s,1") in new stack
[Oct 12 13:10:37] VERBOSE[29403] pbx.c: -- Goto (app-announcement-8,s,1)
[Oct 12 13:10:37] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-8:1] GotoIf("SIP/666-00000162", "1?begin") in new stack
[Oct 12 13:10:37] VERBOSE[29403] pbx.c: -- Goto (app-announcement-8,s,4)
[Oct 12 13:10:37] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-8:4] NoOp("SIP/666-00000162", "Playing announcement Privet1") in new stack
[Oct 12 13:10:37] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-8:5] BackGround("SIP/666-00000162", "custom/p1_1,nm") in new stack
[Oct 12 13:10:37] VERBOSE[29403] file.c: -- <SIP/666-00000162> Playing 'custom/p1_1.slin' (language 'en')
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.176:60392 --->



<------------->
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-664e65cd;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003603 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="07b315a6",uri="sip:192.168.5.200",algorithm=MD5,response="c5b3f8f5e5d2d1e15e952e7abf3424db"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-664e65cd;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003603 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:37] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-664e65cd;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003603 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71468929", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.176:60392:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-8350eda3;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003604 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="71468929",uri="sip:192.168.5.200",algorithm=MD5,response="0a80ba1cb1d5b6e9302f2a2633a6b24d"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-8350eda3;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003604 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK2e395b76;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as34f662ae
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 1a95a67667acc0a838fa1f0a75a04c26@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-8350eda3;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003604 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:37 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:37] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:38] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as34f662ae
Call-ID: 1a95a67667acc0a838fa1f0a75a04c26@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK2e395b76;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:38] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:38] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '1a95a67667acc0a838fa1f0a75a04c26@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-e5639801;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003605 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="71468929",uri="sip:192.168.5.200",algorithm=MD5,response="0a80ba1cb1d5b6e9302f2a2633a6b24d"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-e5639801;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003605 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:39] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-e5639801;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003605 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="784bf158", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-b67bf78d;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003606 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="784bf158",uri="sip:192.168.5.200",algorithm=MD5,response="663a21bcecab6fcc5b37ec1a4500892a"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-b67bf78d;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003606 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK43dd4af6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as32bfb714
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 00eef04724660c7e71c7ca26253101ba@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-b67bf78d;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003606 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:39 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as32bfb714
Call-ID: 00eef04724660c7e71c7ca26253101ba@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK43dd4af6;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '00eef04724660c7e71c7ca26253101ba@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:39] VERBOSE[2713] chan_sip.c: Retransmitting #4 (NAT) to 78.141.179.70:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK30f76c2b;rport
Max-Forwards: 70
From: <sip:99051000113331@sip.skype.com>;tag=as3c2c3989
To: <sip:99051000113331@sip.skype.com>
Call-ID: 66e6d27578caf3e3161eefc547e1339e@193.200.173.1
CSeq: 495 REGISTER
User-Agent: Asterisk PBX 1.6.2.10
Authorization: Digest username="99051000113331", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com", nonce="4cb42add00017688f69421f5e656119d300014e781135ff8", response="56f0455b61cb433953a5e329d5bb1039"
Expires: 600
Contact: <sip:s@82.144.205.16>
Content-Length: 0


---
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-47889c8b;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003607 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="784bf158",uri="sip:192.168.5.200",algorithm=MD5,response="663a21bcecab6fcc5b37ec1a4500892a"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-47889c8b;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003607 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:40] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-47889c8b;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003607 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="792e5657", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Retransmitting #5 (NAT) to 192.168.1.176:60392:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 12 13:10:40] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-8:6] Goto("SIP/666-00000162", "app-announcement-7,s,1") in new stack
[Oct 12 13:10:40] VERBOSE[29403] pbx.c: -- Goto (app-announcement-7,s,1)
[Oct 12 13:10:40] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-7:1] GotoIf("SIP/666-00000162", "1?begin") in new stack
[Oct 12 13:10:40] VERBOSE[29403] pbx.c: -- Goto (app-announcement-7,s,4)
[Oct 12 13:10:40] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-7:4] NoOp("SIP/666-00000162", "Playing announcement Privet2") in new stack
[Oct 12 13:10:40] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-7:5] BackGround("SIP/666-00000162", "custom/p2_1,nm") in new stack
[Oct 12 13:10:40] VERBOSE[29403] file.c: -- <SIP/666-00000162> Playing 'custom/p2_1.slin' (language 'en')
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-3c39f69c;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003608 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="792e5657",uri="sip:192.168.5.200",algorithm=MD5,response="65b148e1a7490de85df19326dbd4dbf3"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-3c39f69c;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003608 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK199f992e;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as4c6d9abf
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 20f5cd8e1a307e89151782fd5e9b5740@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-3c39f69c;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003608 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:40 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as4c6d9abf
Call-ID: 20f5cd8e1a307e89151782fd5e9b5740@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK199f992e;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:40] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '20f5cd8e1a307e89151782fd5e9b5740@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.91:5060:
OPTIONS sip:91@192.168.5.91:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK4199401b;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.5.200>;tag=as4e85b562
To: <sip:91@192.168.5.91:5060>
Contact: <sip:Unknown@192.168.5.200>
Call-ID: 7e0857fe2a368d0e64feafb177e5594d@192.168.5.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.5.91:5060 --->
SIP/2.0 200 OK
To: <sip:91@192.168.5.91:5060>;tag=6390a3b3d7f02ed6i0
From: "Unknown" <sip:Unknown@192.168.5.200>;tag=as4e85b562
Call-ID: 7e0857fe2a368d0e64feafb177e5594d@192.168.5.200
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK4199401b
Server: Linksys/SPA942-5.1.15(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


<------------->
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c: --- (10 headers 0 lines) ---
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '7e0857fe2a368d0e64feafb177e5594d@192.168.5.200' Method: OPTIONS
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.94:5060:
OPTIONS sip:94@192.168.5.94:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK00066ab5;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.5.200>;tag=as054851c7
To: <sip:94@192.168.5.94:5060>
Contact: <sip:Unknown@192.168.5.200>
Call-ID: 1316ce251fe404ed5ce3c4063804e01e@192.168.5.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.5.94:5060 --->
SIP/2.0 200 OK
To: <sip:94@192.168.5.94:5060>;tag=f1a85ce786ea7c5i0
From: "Unknown" <sip:Unknown@192.168.5.200>;tag=as054851c7
Call-ID: 1316ce251fe404ed5ce3c4063804e01e@192.168.5.200
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK00066ab5
Server: Linksys/SPA942-5.1.15(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


<------------->
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c: --- (10 headers 0 lines) ---
[Oct 12 13:10:41] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '1316ce251fe404ed5ce3c4063804e01e@192.168.5.200' Method: OPTIONS
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-439386ec;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003609 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="792e5657",uri="sip:192.168.5.200",algorithm=MD5,response="65b148e1a7490de85df19326dbd4dbf3"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-439386ec;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003609 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:42] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-439386ec;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003609 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72b1b871", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-842506eb;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003610 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="72b1b871",uri="sip:192.168.5.200",algorithm=MD5,response="63c78dc4d768a51d173b16cca92e5cc2"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-842506eb;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003610 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK26a090eb;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as06335a73
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 0c26857310f3f3ae6c8d83e20dd75dc8@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-842506eb;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003610 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:42 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as06335a73
Call-ID: 0c26857310f3f3ae6c8d83e20dd75dc8@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK26a090eb;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:42] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '0c26857310f3f3ae6c8d83e20dd75dc8@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-b8fab389;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003611 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="72b1b871",uri="sip:192.168.5.200",algorithm=MD5,response="63c78dc4d768a51d173b16cca92e5cc2"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-b8fab389;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003611 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:43] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-b8fab389;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003611 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e9deb13", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-7e2cc2;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003612 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="7e9deb13",uri="sip:192.168.5.200",algorithm=MD5,response="1a8f8e0a0f4b7f15d68866adefb361b3"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-7e2cc2;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003612 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK73496655;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as0bfa44b0
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 204f70d1402c725d2bbdcafe087b8846@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-7e2cc2;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003612 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:43 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Retransmitting #5 (NAT) to 78.141.179.70:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK30f76c2b;rport
Max-Forwards: 70
From: <sip:99051000113331@sip.skype.com>;tag=as3c2c3989
To: <sip:99051000113331@sip.skype.com>
Call-ID: 66e6d27578caf3e3161eefc547e1339e@193.200.173.1
CSeq: 495 REGISTER
User-Agent: Asterisk PBX 1.6.2.10
Authorization: Digest username="99051000113331", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com", nonce="4cb42add00017688f69421f5e656119d300014e781135ff8", response="56f0455b61cb433953a5e329d5bb1039"
Expires: 600
Contact: <sip:s@82.144.205.16>
Content-Length: 0


---
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as0bfa44b0
Call-ID: 204f70d1402c725d2bbdcafe087b8846@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK73496655;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:43] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '204f70d1402c725d2bbdcafe087b8846@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@app-announcement-7:6] Goto("SIP/666-00000162", "ivr-5,s,1") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Goto (ivr-5,s,1)
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:1] Set("SIP/666-00000162", "MSG=custom/m1_2") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:2] Set("SIP/666-00000162", "LOOPCOUNT=0") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:3] Set("SIP/666-00000162", "__DIR-CONTEXT=") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:4] Set("SIP/666-00000162", "_IVR_CONTEXT_ivr-5=") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:5] Set("SIP/666-00000162", "_IVR_CONTEXT=ivr-5") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:6] GotoIf("SIP/666-00000162", "1?begin") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Goto (ivr-5,s,9)
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:9] Set("SIP/666-00000162", "TIMEOUT(digit)=3") in new stack
[Oct 12 13:10:44] VERBOSE[29403] func_timeout.c: -- Digit timeout set to 3.000
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:10] Set("SIP/666-00000162", "TIMEOUT(response)=7") in new stack
[Oct 12 13:10:44] VERBOSE[29403] func_timeout.c: -- Response timeout set to 7.000
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:11] Set("SIP/666-00000162", "__IVR_RETVM=") in new stack
[Oct 12 13:10:44] VERBOSE[29403] pbx.c: -- Executing [s@ivr-5:12] ExecIf("SIP/666-00000162", "1?Background(custom/m1_2)") in new stack
[Oct 12 13:10:44] VERBOSE[29403] file.c: -- <SIP/666-00000162> Playing 'custom/m1_2.slin' (language 'en')
[Oct 12 13:10:44] VERBOSE[2713] chan_sip.c: Retransmitting #6 (NAT) to 192.168.1.176:60392:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.176:60392;branch=z9hG4bK-d8754z-2d61b607fc7a2a4a-1---d8754z-;received=192.168.1.176;rport=60392
From: "Admin"<sip:666@192.168.5.200>;tag=9e488857
To: "7777"<sip:7777@192.168.5.200>;tag=as76ce5606
Call-ID: OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7777@82.144.205.16>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1576933076 1576933076 IN IP4 82.144.205.16
s=Asterisk PBX 1.6.2.10
c=IN IP4 82.144.205.16
t=0 0
m=audio 16924 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-dc4ff02a;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003613 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="7e9deb13",uri="sip:192.168.5.200",algorithm=MD5,response="1a8f8e0a0f4b7f15d68866adefb361b3"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-dc4ff02a;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003613 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:45] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-dc4ff02a;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003613 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2de0488b", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-3208103e;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003614 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="2de0488b",uri="sip:192.168.5.200",algorithm=MD5,response="3f2af175fb70652e0e96c1c3ea8aabb6"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-3208103e;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003614 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK6bab6868;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as1e6a7911
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 2c123d7e4d32d40a51ccb9cd2bc5e248@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-3208103e;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003614 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:45 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as1e6a7911
Call-ID: 2c123d7e4d32d40a51ccb9cd2bc5e248@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK6bab6868;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:45] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '2c123d7e4d32d40a51ccb9cd2bc5e248@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-719dcd2c;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003615 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="2de0488b",uri="sip:192.168.5.200",algorithm=MD5,response="3f2af175fb70652e0e96c1c3ea8aabb6"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-719dcd2c;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003615 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:46] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-719dcd2c;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003615 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f1ac540", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-d2627b53;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003616 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="2f1ac540",uri="sip:192.168.5.200",algorithm=MD5,response="96511398d8deb8c2c5c5fa246b0fd023"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-d2627b53;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003616 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK13b30550;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as66ee0015
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 314f5899378482d35fad08f80d2286fd@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-d2627b53;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003616 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:46 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as66ee0015
Call-ID: 314f5899378482d35fad08f80d2286fd@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK13b30550;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:46] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '314f5899378482d35fad08f80d2286fd@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:47] VERBOSE[2713] chan_sip.c: Retransmitting #6 (NAT) to 78.141.179.70:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK30f76c2b;rport
Max-Forwards: 70
From: <sip:99051000113331@sip.skype.com>;tag=as3c2c3989
To: <sip:99051000113331@sip.skype.com>
Call-ID: 66e6d27578caf3e3161eefc547e1339e@193.200.173.1
CSeq: 495 REGISTER
User-Agent: Asterisk PBX 1.6.2.10
Authorization: Digest username="99051000113331", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com", nonce="4cb42add00017688f69421f5e656119d300014e781135ff8", response="56f0455b61cb433953a5e329d5bb1039"
Expires: 600
Contact: <sip:s@82.144.205.16>
Content-Length: 0


---
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-50efcf85;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003617 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="2f1ac540",uri="sip:192.168.5.200",algorithm=MD5,response="96511398d8deb8c2c5c5fa246b0fd023"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-50efcf85;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003617 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:48] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-50efcf85;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003617 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f9fb801", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-a84f0228;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003618 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="0f9fb801",uri="sip:192.168.5.200",algorithm=MD5,response="a86241c9ba86e1747f95f3e4e51a0806"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-a84f0228;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003618 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK7dabb392;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as35284367
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 586fd63f232ba9ae170b969b3a2c986f@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-a84f0228;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003618 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:48 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as35284367
Call-ID: 586fd63f232ba9ae170b969b3a2c986f@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK7dabb392;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:48] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '586fd63f232ba9ae170b969b3a2c986f@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:48] WARNING[2713] chan_sip.c: Maximum retries exceeded on transmission OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct 12 13:10:48] WARNING[2713] chan_sip.c: Hanging up call OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg. - no reply to our critical packet (see doc/sip-retransmit.txt).
[Oct 12 13:10:48] VERBOSE[29403] pbx.c: == Spawn extension (ivr-5, s, 12) exited non-zero on 'SIP/666-00000162'
[Oct 12 13:10:48] VERBOSE[29403] pbx.c: -- Executing [h@ivr-5:1] Hangup("SIP/666-00000162", "") in new stack
[Oct 12 13:10:48] VERBOSE[29403] pbx.c: == Spawn extension (ivr-5, h, 1) exited non-zero on 'SIP/666-00000162'
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-d1a57402;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003619 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="0f9fb801",uri="sip:192.168.5.200",algorithm=MD5,response="a86241c9ba86e1747f95f3e4e51a0806"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-d1a57402;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003619 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:49] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-d1a57402;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003619 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="567f6ee7", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog 'OGVhNTdmY2RkODQ4ZDMxOGQwZGFjNjc4Yjk2ZjAxMjg.' Method: INVITE
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-72562a09;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003620 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="567f6ee7",uri="sip:192.168.5.200",algorithm=MD5,response="c853b3144fa2b1d528dd3d2a14943730"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-72562a09;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003620 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK4bcdc766;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as49ae7169
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 6ee959bf02ac355c608f2d67794bf195@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-72562a09;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003620 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:49 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as49ae7169
Call-ID: 6ee959bf02ac355c608f2d67794bf195@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK4bcdc766;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:49] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '6ee959bf02ac355c608f2d67794bf195@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-560928f1;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003621 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="567f6ee7",uri="sip:192.168.5.200",algorithm=MD5,response="c853b3144fa2b1d528dd3d2a14943730"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-560928f1;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003621 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:51] NOTICE[2713] chan_sip.c: Correct auth, but based on stale nonce received from '"90" <sip:90@192.168.5.200>'
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-560928f1;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003621 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="254dd8c5", stale=true
Content-Length: 0


<------------>
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
REGISTER sip:192.168.5.200 SIP/2.0
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-37ad8e48;rport
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003622 REGISTER
Max-Forwards: 70
Authorization: Digest username="90",realm="asterisk",nonce="254dd8c5",uri="sip:192.168.5.200",algorithm=MD5,response="bc436517a9ed251f0428ff46caf76c2c"
Contact: "90" <sip:90@82.207.47.143:10085>;expires=300
Warning: 399 spa "Full Cone NAT Detected"
User-Agent: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: --- (15 headers 0 lines) ---
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Sending to 192.168.1.1 : 5060 (NAT)
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-37ad8e48;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003622 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
OPTIONS sip:90@82.207.47.143:10085 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK2b1ef011;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as068bc5d5
To: <sip:90@82.207.47.143:10085>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 1ac6f75e1439a83b367dabc60821aee9@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.207.47.143:10085;branch=z9hG4bK-37ad8e48;received=192.168.1.1;rport=5060
From: "90" <sip:90@192.168.5.200>;tag=d64be1ad85387785o2
To: "90" <sip:90@192.168.5.200>;tag=as7b95af12
Call-ID: a89f114-b603d59d@192.168.1.1
CSeq: 1003622 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:90@82.207.47.143:10085>;expires=300
Date: Tue, 12 Oct 2010 10:10:51 GMT
Content-Length: 0


<------------>
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Scheduling destruction of SIP dialog 'a89f114-b603d59d@192.168.1.1' in 32000 ms (Method: REGISTER)
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
To: <sip:90@82.207.47.143:10085>;tag=c1bcaacd404940e5i2
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as068bc5d5
Call-ID: 1ac6f75e1439a83b367dabc60821aee9@82.144.205.16
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK2b1ef011;received=192.168.5.200;rport=5060
Server: Linksys/SPA9000-6.1.5
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces


<------------->
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: --- (11 headers 0 lines) ---
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '1ac6f75e1439a83b367dabc60821aee9@82.144.205.16' Method: OPTIONS
[Oct 12 13:10:51] NOTICE[2713] chan_sip.c: -- Registration for '99051000113331@sip.skype.com' timed out, trying again (Attempt #106)
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: REGISTER 11 headers, 0 lines
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 78.141.179.70:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK02537f39;rport
Max-Forwards: 70
From: <sip:99051000113331@sip.skype.com>;tag=as1a763c94
To: <sip:99051000113331@sip.skype.com>
Call-ID: 66e6d27578caf3e3161eefc547e1339e@193.200.173.1
CSeq: 496 REGISTER
User-Agent: Asterisk PBX 1.6.2.10
Authorization: Digest username="99051000113331", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com", nonce="4cb42add00017688f69421f5e656119d300014e781135ff8", response="56f0455b61cb433953a5e329d5bb1039"
Expires: 600
Contact: <sip:s@82.144.205.16>
Content-Length: 0


---
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Really destroying SIP dialog '66e6d27578caf3e3161eefc547e1339e@193.200.173.1' Method: REGISTER
[Oct 12 13:10:51] VERBOSE[2713] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.176:60392:
OPTIONS sip:666@192.168.1.176:60392;rinstance=19e19ceddef67e14 SIP/2.0
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK368cd7a0;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@82.144.205.16>;tag=as44eb67aa
To: <sip:666@192.168.1.176:60392;rinstance=19e19ceddef67e14>
Contact: <sip:Unknown@82.144.205.16>
Call-ID: 47fd75fa4e62b8667a15a39b41c428a0@82.144.205.16
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.10
Date: Tue, 12 Oct 2010 10:10:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Oct 12 13:10:52] VERBOSE[2713] chan_sip.c:
<--- SIP read from UDP:192.168.1.176:60392 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.144.205.16:5060;branch=z9hG4bK368cd7a0;rport=5060;received=192.168.5.200
Contact: <sip:192.168.1.176:60392>
To: <sip:666@192.168.1.176:60392;rinstance=19e19ceddef67e14>;tag=51266a1f
From: "Unknown"<sip:Unknown@82.144.205.16>;tag=as44eb67aa
Call-ID: 47fd75fa4e62b8667a15a39b41c428a0@82.144.205.16
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


Это с момента звонка и до последнего упоминания о номере звонившего.
Я звонил с номера "666" IP:192.168.1.176 на номер 7777 (эмулятор входящего вызова)
2010-10-12 14:22

1 2>
Войдите чтобы оставить сообщение!
Добавить страницу в закладки:  Delicious Google Slashdot Yahoo Yandex.ru Reddit Digg Technorati Bobrdobr.ru Newsland.ru Smi2.ru Rumarkz.ru Vaau.ru Memori.ru Rucity.com Moemesto.ru News2.ru Mister-Wong.ru Myscoop.ru 100zakladok.ru