asterisk*CLI> console dial 950956141@from-next
    -- Executing [950956141@from-next:1] NoOp("Console/default", "") in new stack
    -- Executing [950956141@from-next:2] SIPAddHeader("Console/default", "Remote-Party-ID:<sip:950479369@provider.com\;user=phone>\;privacy=off\;party=calling") in new stack
    -- Executing [950956141@from-next:3] Dial("Console/default", "SIP/NEXTC/950956141,,r") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 192.168.5.100 port 10420
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 193.22.119.20:5060:
INVITE sip:950956141@provider.com SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;rport
Max-Forwards: 70
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Contact: <sip:account@192.168.5.100>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Date: Wed, 27 Apr 2011 15:20:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Remote-Party-ID: <sip:950479369@provider.com;user=phone>;privacy=off;party=calling
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1031639655 1031639655 IN IP4 192.168.5.100
s=Asterisk PBX 1.6.0.28
c=IN IP4 192.168.5.100
t=0 0
m=audio 10420 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called NEXTC/950956141
Really destroying SIP dialog '717ebb64653e00f77c83516c637666fa@127.0.1.1' Method: REGISTER
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;rport=5060;received=333.444.555.666
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=ba3d51acad53eeb51d56ab2459dbff7b.b6e2
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="provider.com", nonce="1111"
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 193.22.119.20:5060:
ACK sip:950956141@provider.com SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;rport
Max-Forwards: 70
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=ba3d51acad53eeb51d56ab2459dbff7b.b6e2
Contact: <sip:account@192.168.5.100>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.28
Content-Length: 0

---
Audio is at 192.168.5.100 port 10420
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 193.22.119.20:5060:
INVITE sip:950956141@provider.com SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport
Max-Forwards: 70
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Contact: <sip:account@192.168.5.100>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Proxy-Authorization: Digest username="account", realm="provider.com", algorithm=MD5, uri="sip:950956141@provider.com", nonce="111", response="2222"
Date: Wed, 27 Apr 2011 15:20:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Remote-Party-ID: <sip:950479369@provider.com;user=phone>;privacy=off;party=calling
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1031639655 1031639656 IN IP4 192.168.5.100
s=Asterisk PBX 1.6.0.28
c=IN IP4 192.168.5.100
t=0 0
m=audio 10420 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=9bbe2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-33e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=9bbe2c31
To: sip:333.444.555.666:5060;tag=as1e8a6558
Call-ID: 415ba657-33e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-33e61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=1ebe2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-b5e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=1ebe2c31
To: sip:333.444.555.666:5060;tag=as517672ba
Call-ID: 415ba657-b5e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-b5e61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=5ebe2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-f5e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=5ebe2c31
To: sip:333.444.555.666:5060;tag=as49ee3853
Call-ID: 415ba657-f5e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-f5e61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=02ce2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-a9e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=02ce2c31
To: sip:333.444.555.666:5060;tag=as31fd5d89
Call-ID: 415ba657-a9e61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-a9e61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=26ce2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-cde61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=26ce2c31
To: sip:333.444.555.666:5060;tag=as42b7cf41
Call-ID: 415ba657-cde61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-cde61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=56ce2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-fde61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=56ce2c31
To: sip:333.444.555.666:5060;tag=as284ac6ac
Call-ID: 415ba657-fde61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-fde61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=98ce2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-30f61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=98ce2c31
To: sip:333.444.555.666:5060;tag=as085ce297
Call-ID: 415ba657-30f61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-30f61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=b9ce2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-51f61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=b9ce2c31
To: sip:333.444.555.666:5060;tag=as64a7d3a9
Call-ID: 415ba657-51f61026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-51f61026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=6cde2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-04071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=6cde2c31
To: sip:333.444.555.666:5060;tag=as138d3f54
Call-ID: 415ba657-04071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-04071026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=0ede2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-a5071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=0ede2c31
To: sip:333.444.555.666:5060;tag=as35a3efbc
Call-ID: 415ba657-a5071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-a5071026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=aede2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-46071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=aede2c31
To: sip:333.444.555.666:5060;tag=as0dd2d71d
Call-ID: 415ba657-46071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-46071026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=dede2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-76071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=dede2c31
To: sip:333.444.555.666:5060;tag=as2d0b1b44
Call-ID: 415ba657-76071026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-76071026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=7eee2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-16171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=7eee2c31
To: sip:333.444.555.666:5060;tag=as5d13cba8
Call-ID: 415ba657-16171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-16171026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=40fe2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-e7171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=40fe2c31
To: sip:333.444.555.666:5060;tag=as25e98403
Call-ID: 415ba657-e7171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-e7171026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=a1fe2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-49171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=a1fe2c31
To: sip:333.444.555.666:5060;tag=as2cffdfa8
Call-ID: 415ba657-49171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-49171026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
OPTIONS sip:333.444.555.666:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0
From: sip:pinger@provider.com;tag=76fe2c31
To: sip:333.444.555.666:5060
Call-ID: 415ba657-1e171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Looking for s in from-sip-external (domain 333.444.555.666)

<--- Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20:5060;branch=0;received=193.22.119.20
From: sip:pinger@provider.com;tag=76fe2c31
To: sip:333.444.555.666:5060;tag=as284c508c
Call-ID: 415ba657-1e171026-0c8082@193.22.119.20
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.5.100>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '415ba657-1e171026-0c8082@193.22.119.20' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '3dfc2140626bcd5c173dea8b7d1218ec@127.0.1.1' Method: REGISTER
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
SIP/2.0 100 Runing CPL script...
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport=5060;received=333.444.555.666
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 INVITE
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
SIP/2.0 100 Runing CPL script...
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport=5060;received=333.444.555.666
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 INVITE
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
INVITE sip:s@192.168.5.100 SIP/2.0
Record-Route: <sip:193.22.119.20;lr=on;ftag=as636f7e93>
Record-Route: <sip:193.22.119.20;lr=on;ftag=as636f7e93>
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.e7840977.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.d7840977.0
Via: SIP/2.0/UDP 192.168.5.100:5060;received=333.444.555.666;branch=z9hG4bK628c3d36;rport=5060
Max-Forwards: 31
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Contact: <sip:account@333.444.555.666:5060;nat=yes;nat=yes>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 INVITE
Date: Wed, 27 Apr 2011 15:20:42 GMT
Remote-Party-ID: <sip:950479369@provider.com;user=phone>;privacy=off;party=calling
Content-Type: application/sdp
Content-Length: 330
P-RTP-Proxy: YES
P-Asserted-Identity: <sip:950479369@provider.com;user=phone>
X-CPLFROM: account

v=0
o=root 1031639655 1031639656 IN IP4 192.168.5.100
s=Asterisk PBX 1.6.0.28
c=IN IP4 193.22.119.2
t=0 0
m=audio 63334 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
--- (19 headers 15 lines) ---

<--- Transmitting (NAT) to 193.22.119.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.e7840977.0;received=193.22.119.20
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.d7840977.0
Via: SIP/2.0/UDP 192.168.5.100:5060;received=333.444.555.666;branch=z9hG4bK628c3d36;rport=5060
Record-Route: <sip:193.22.119.20;lr=on;ftag=as636f7e93>
Record-Route: <sip:193.22.119.20;lr=on;ftag=as636f7e93>
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:account@192.168.5.100>
Content-Length: 0

<------------>
    -- Now forwarding Console/default to 'Local/s@from-provider' (thanks to SIP/NEXTC-0000048a)
Scheduling destruction of SIP dialog '0b0737290c8c655f1ca7fe5c3b308b3a@provider.com' in 6400 ms (Method: INVITE)
[Apr 27 17:20:42] WARNING[17514]: chan_sip.c:15559 func_header_read: This function can only be used on SIP channels.
Reliably Transmitting (NAT) to 193.22.119.20:5060:
CANCEL sip:950956141@provider.com SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport
Max-Forwards: 70
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 1.6.0.28
Content-Length: 0

---
    -- Executing [s@from-provider:1] NoOp("Local/s@from-provider-c91f;2", "") in new stack
[Apr 27 17:20:42] WARNING[17514]: chan_sip.c:15559 func_header_read: This function can only be used on SIP channels.
Scheduling destruction of SIP dialog '0b0737290c8c655f1ca7fe5c3b308b3a@provider.com' in 6400 ms (Method: INVITE)
    -- Executing [s@from-provider:2] Set("Local/s@from-provider-c91f;2", "dialed=") in new stack
    -- Executing [s@from-provider:3] Goto("Local/s@from-provider-c91f;2", "ext-did,,1") in new stack
    -- Goto (ext-did,s,1)
[Apr 27 17:20:42] WARNING[17514]: pbx.c:3781 __ast_pbx_run: Channel 'Local/s@from-provider-c91f;2' sent into invalid extension 's' in context 'ext-did', but no invalid handler
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [950956141@from-next:4] Hangup("Console/default", "") in new stack
  == Spawn extension (from-next, 950956141, 4) exited non-zero on 'Console/default'
  --- <("<) --- Hangup on Console --- (>")> ---
Really destroying SIP dialog '225dc8ab4963b9c0030343da710c040e@127.0.1.1' Method: REGISTER
asterisk*CLI> 
<--- SIP read from UDP://192.168.31.101:5060 --->

<------------->
Really destroying SIP dialog '4bdef3090f8df2673fe4f43073eb7595@127.0.1.1' Method: REGISTER
Retransmitting #1 (NAT) to 193.22.119.20:5060:
CANCEL sip:950956141@provider.com SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport
Max-Forwards: 70
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 1.6.0.28
Content-Length: 0

---
Really destroying SIP dialog '7ae54185564cfa6126b99a1530084f65@127.0.1.1' Method: REGISTER
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport=5060;received=333.444.555.666
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 CANCEL
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
CANCEL sip:s@192.168.5.100 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.e7840977.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.d7840977.0
Via: SIP/2.0/UDP 192.168.5.100:5060;received=333.444.555.666;branch=z9hG4bK628c3d36;rport=5060
Max-Forwards: 31
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 CANCEL
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 193.22.119.20 : 5060 (NAT)
Scheduling destruction of SIP dialog '0b0737290c8c655f1ca7fe5c3b308b3a@provider.com' in 6400 ms (Method: CANCEL)

<--- Reliably Transmitting (NAT) to 193.22.119.20:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>

<--- Transmitting (NAT) to 193.22.119.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.e7840977.0;received=193.22.119.20
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK14b2.d7840977.0
Via: SIP/2.0/UDP 192.168.5.100:5060;received=333.444.555.666;branch=z9hG4bK628c3d36;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog '1782a17244444ab40d24014327cb6b9e@127.0.1.1' Method: REGISTER
Really destroying SIP dialog '2a754bde4f0638e4249ee5905c243592@127.0.1.1' Method: REGISTER
asterisk*CLI> 
<--- SIP read from UDP://193.22.119.20:5060 --->
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK628c3d36;rport=5060;received=333.444.555.666
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 103 CANCEL
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (NAT) to 193.22.119.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '540b8664675933455eef7c5801e6ee6d@127.0.1.1' Method: REGISTER
Retransmitting #2 (NAT) to 193.22.119.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #3 (NAT) to 193.22.119.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '415ba657-06a61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-88a61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-c8a61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-7ca61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-90b61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-c0b61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-03b61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-24b61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-d6c61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-78c61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-19c61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-49c61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-e8d61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-bad61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-1cd61026-2a8082@193.22.119.20' Method: OPTIONS
Really destroying SIP dialog '415ba657-e0e61026-2a8082@193.22.119.20' Method: OPTIONS
asterisk*CLI> consol
<--- SIP read from UDP://192.168.31.108:5060 --->

<------------->
Retransmitting #4 (NAT) to 193.22.119.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (NAT) to 10.0.0.43:5060:
OPTIONS sip:102@10.0.0.43:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK11c32d7a;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.4>;tag=as19893121
To: <sip:102@10.0.0.43:5060>
Contact: <sip:Unknown@10.0.0.4>
Call-ID: 4747e4210f3dca5e466f730042d6c8fa@10.0.0.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Date: Wed, 27 Apr 2011 15:20:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
asterisk*CLI> console han
<--- SIP read from UDP://10.0.0.43:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK11c32d7a;rport=5060
From: "Unknown" <sip:Unknown@10.0.0.4>;tag=as19893121
To: <sip:102@10.0.0.43:5060>;tag=3601982364
Call-ID: 4747e4210f3dca5e466f730042d6c8fa@10.0.0.4
CSeq: 102 OPTIONS
Contact: <sip:102@10.0.0.43:5060>
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '4747e4210f3dca5e466f730042d6c8fa@10.0.0.4' Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.0.44:5060:
OPTIONS sip:104@10.0.0.44:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK006277c5;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.4>;tag=as037095bc
To: <sip:104@10.0.0.44:5060>
Contact: <sip:Unknown@10.0.0.4>
Call-ID: 180e6d2d5001b9c91745134826a159a5@10.0.0.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Date: Wed, 27 Apr 2011 15:20:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
asterisk*CLI> console hang
<--- SIP read from UDP://10.0.0.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK006277c5;rport=5060
From: "Unknown" <sip:Unknown@10.0.0.4>;tag=as037095bc
To: <sip:104@10.0.0.44:5060>;tag=2207677857
Call-ID: 180e6d2d5001b9c91745134826a159a5@10.0.0.4
CSeq: 102 OPTIONS
Contact: <sip:104@10.0.0.44:5060>
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '180e6d2d5001b9c91745134826a159a5@10.0.0.4' Method: OPTIONS
asterisk*CLI> console hangup
<--- SIP read from UDP://192.168.31.96:5060 --->

<------------->
Retransmitting #5 (NAT) to 193.22.119.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<------------->
--- (13 headers 0 lines) ---
Sending to 10.0.0.52 : 5060 (NAT)
Reliably Transmitting (NAT) to 10.0.0.52:5060:
OPTIONS sip:201@10.0.0.52:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5a494b87;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.4>;tag=as013ebf5e
To: <sip:201@10.0.0.52:5060>
Contact: <sip:Unknown@10.0.0.4>
Call-ID: 417f38b50d3a6642741bd240472822dd@10.0.0.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.28
Date: Wed, 27 Apr 2011 15:20:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<------------>
Scheduling destruction of SIP dialog '1985409740@10_0_0_52' in 32000 ms (Method: REGISTER)
asterisk*CLI> console hangup
<--- SIP read from UDP://10.0.0.52:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5a494b87;rport=5060
From: "Unknown" <sip:Unknown@10.0.0.4>;tag=as013ebf5e
To: <sip:201@10.0.0.52:5060>;tag=2975443970
Call-ID: 417f38b50d3a6642741bd240472822dd@10.0.0.4
CSeq: 102 OPTIONS
Contact: <sip:201@10.0.0.52:5060>
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '417f38b50d3a6642741bd240472822dd@10.0.0.4' Method: OPTIONS
asterisk*CLI> console hangup
No call to hang up
Command 'console hangup' failed.
Retransmitting #6 (NAT) to 193.22.119.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK7b9a1845;received=193.22.119.20;rport=5060
From: "MyName Here" <sip:account@provider.com>;tag=as636f7e93
To: <sip:950956141@provider.com>;tag=0dc66d454840fa7d0b618c9f0fdb83b7-d1fa
Call-ID: 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.28
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
asterisk*CLI> exit
<--- SIP read from UDP://192.168.31.57:5060 --->

<------------->
asterisk*CLI> exit
<--- SIP read from UDP://192.168.31.63:5060 --->

<------------->
asterisk*CLI> 
<--- SIP read from UDP://192.168.31.79:5060 --->

<------------->
asterisk*CLI> sip set debug off
SIP Debugging Disabled
[Apr 27 17:20:52] WARNING[3064]: chan_sip.c:2961 retrans_pkt: Maximum retries exceeded on transmission 0b0737290c8c655f1ca7fe5c3b308b3a@provider.com for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
asterisk*CLI>