<--- SIP read from 10.4.20.190:5060 --->
H
<------------->
--- (1 headers 0 lines) ---
Really destroying SIP dialog '53e0c-e87-0001e10a19b1@yourbox.org' Method: REGIST                                                                                                                        ER
[May 24 11:03:26] NOTICE[13038]: chan_sip.c:6954 sip_reregister:    -- Re-regist                                                                                                                        ration for  allsolutions@proxy01.sipphone.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 198.65.166.131:5060:
REGISTER sip:proxy01.sipphone.com SIP/2.0
Via: SIP/2.0/UDP 68.109.169.37:5060;branch=z9hG4bK444035a0;rport
From: <sip:allsolutions@proxy01.sipphone.com>;tag=as77b8977d
To: <sip:allsolutions@proxy01.sipphone.com>
Call-ID: 44feebbd39bea976092f0cfa036be65e@127.0.1.1
CSeq: 117 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="allsolutions", realm="proxy01.sipphone.com", alg                                                                                                                        orithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4655d475016b89b2a4e5393b6f07a                                                                                                                        b545e8f6886", response="eaa2f5ace170b4dee55bdbcecf6044bc", opaque=""
Expires: 120
Contact: <sip:allsolutions@68.109.169.37>
Event: registration
Content-Length: 0


---
Itchy*CLI>
<--- SIP read from 198.65.166.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.109.169.37:5060;branch=z9hG4bK444035a0;rport=63656
From: <sip:allsolutions@proxy01.sipphone.com>;tag=as77b8977d
To: <sip:allsolutions@proxy01.sipphone.com>;tag=21a483426c2cd5d9b85bffe6bba40a2e                                                                                                                        .b0ea
Call-ID: 44feebbd39bea976092f0cfa036be65e@127.0.1.1
CSeq: 117 REGISTER
Contact: <sip:allsolutions@68.109.169.37>;expires=120
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '44feebbd39bea976092f0cfa036be65e@127.0.1.1                                                                                                                        ' in 32000 ms (Method: REGISTER)
[May 24 11:03:26] NOTICE[13038]: chan_sip.c:11771 handle_response_register: Outb                                                                                                                        ound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregi                                                                                                                        stration in 105 s)
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
INVITE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK93040
To: <sip:1000@10.4.20.111>
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 356 INVITE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Contact: <sip:7101@10.4.20.184:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284

v=0
o=- 836899966 836899976 IN IP4 10.4.20.184
s=Uplink Skype2Sip
c=IN IP4 10.4.20.184
t=0 0
m=audio 10000 RTP/AVP 0 8 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


<------------->
--- (13 headers 14 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)
Using INVITE request as basis request - 1180028879-3040-TOPLAP@10.4.20.184

<--- Reliably Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK93040;received=10.4.20.184;rport                                                                                                                        =5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>;tag=as3add4983
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 356 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a666ba1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1180028879-3040-TOPLAP@10.4.20.184' in 320                                                                                                                        00 ms (Method: INVITE)
Found user '7101'
Retransmitting #1 (NAT) to 10.4.20.184:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK93040;received=10.4.20.184;rport                                                                                                                        =5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>;tag=as3add4983
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 356 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a666ba1"
Content-Length: 0


---
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
ACK sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK93040
To: <sip:1000@10.4.20.111>;tag=as3add4983
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 356 ACK
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from 10.4.20.184:5060 --->
INVITE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK103040
To: <sip:1000@10.4.20.111>
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 357 INVITE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Contact: <sip:7101@10.4.20.184:5060>
Proxy-Authorization: Digest username="7101",realm="asterisk",nonce="7a666ba1",ur                                                                                                                        i="sip:1000@10.4.20.111",response="b32de259e0d9104ee5272e5a548ac4df",opaque=""
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284

v=0
o=- 836899966 836899976 IN IP4 10.4.20.184
s=Uplink Skype2Sip
c=IN IP4 10.4.20.184
t=0 0
m=audio 10000 RTP/AVP 0 8 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


<------------->
--- (14 headers 14 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)
Using INVITE request as basis request - 1180028879-3040-TOPLAP@10.4.20.184
Found user '7101'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 13
Found RTP audio format 101
Peer audio RTP is at port 10.4.20.184:10000
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format GSM for ID 3
Found description format CN for ID 13
Found description format telephone-event for ID 101
Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex                                                                                                                        |ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/                                                                                                                        video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone                                                                                                                        -event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.4.20.184:10000
Looking for 1000 in fromskype (domain 10.4.20.111)
list_route: hop: <sip:7101@10.4.20.184:5060>

<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK103040;received=10.4.20.184;rpor                                                                                                                        t=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 357 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0


<------------>

<--- SIP read from 10.4.20.184:5060 --->
ACK sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK93040
To: <sip:1000@10.4.20.111>;tag=as3add4983
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 356 ACK
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- Executing [1000@fromskype:1] Dial("SIP/7101-081d6660", "Zap/2&SIP/101&SIP                                                                                                                        /102&SIP/103") in new stack
[May 24 11:03:30] WARNING[13077]: app_dial.c:1081 dial_exec_full: Unable to crea                                                                                                                        te channel of type 'Zap' (cause 0 - Unknown)
Really destroying SIP dialog '206e523072045b1604c2d8783fe428c4@127.0.1.1' Method                                                                                                                        : INVITE
[May 24 11:03:30] WARNING[13077]: app_dial.c:1081 dial_exec_full: Unable to crea                                                                                                                        te channel of type 'SIP' (cause 3 - No route to destination)
Really destroying SIP dialog '181fb3173cc3855c4e4269ba7a705901@127.0.1.1' Method                                                                                                                        : INVITE
[May 24 11:03:30] WARNING[13077]: app_dial.c:1081 dial_exec_full: Unable to crea                                                                                                                        te channel of type 'SIP' (cause 3 - No route to destination)
Audio is at 10.4.20.111 port 10010
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.4.20.190:5060:
INVITE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK3170eaf9;rport
From: "From Skype" <sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>
Contact: <sip:3133731337@10.4.20.111>
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 24 May 2007 18:03:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 5910 5910 IN IP4 10.4.20.111
s=session
c=IN IP4 10.4.20.111
t=0 0
m=audio 10010 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
    -- Called 103
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 100 Trying
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
From: "From Skype"<sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK3170eaf9;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 180 Ringing
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
From: "From Skype"<sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK3170eaf9;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/103-081d8950 is ringing
Itchy*CLI>
<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK103040;received=10.4.20.184;rpor                                                                                                                        t=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>;tag=as74244df7
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 357 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0


<------------>
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
H
<------------->
--- (1 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
From: "From Skype"<sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK3170eaf9;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Supported: timer,100rel
Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Type: application/sdp
Content-Length: 132

v=0
o=103 0 0 IN IP4 10.4.20.190
s=-
c=IN IP4 10.4.20.190
t=0 0
m=audio 8198 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000

<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.4.20.190:8198
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0                                                                                                                        x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone                                                                                                                        -event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.4.20.190:8198
list_route: hop: <sip:103@10.4.20.190:5060;user=phone>
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port                                                                                                                         to send to
set_destination: set destination to 10.4.20.190, port 5060
Transmitting (NAT) to 10.4.20.190:5060:
ACK sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK3cb4ef6d;rport
From: "From Skype" <sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
Contact: <sip:3133731337@10.4.20.111>
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/103-081d8950 answered SIP/7101-081d6660
Audio is at 10.4.20.111 port 10004
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Itchy*CLI>
<--- Reliably Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK103040;received=10.4.20.184;rpor                                                                                                                        t=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>;tag=as74244df7
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 357 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 5910 5910 IN IP4 10.4.20.111
s=session
c=IN IP4 10.4.20.111
t=0 0
m=audio 10004 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/7101-081d6660 and SIP/103-081d8950
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
ACK sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK113040
To: <sip:1000@10.4.20.111>;tag=as74244df7
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 357 ACK
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Proxy-Authorization: Digest username="7101",realm="asterisk",nonce="7a666ba1",ur                                                                                                                        i="sip:1000@10.4.20.111",response="b32de259e0d9104ee5272e5a548ac4df",opaque=""
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
H
<------------->
--- (1 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
BYE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK123040
To: <sip:1000@10.4.20.111>;tag=as74244df7
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 358 BYE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)

<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK123040;received=10.4.20.184;rpor                                                                                                                        t=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>;tag=as74244df7
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 358 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '734db83e575ad5da3fa724ae388d4a32@10.4.20.1                                                                                                                        11' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port                                                                                                                         to send to
set_destination: set destination to 10.4.20.190, port 5060
Reliably Transmitting (NAT) to 10.4.20.190:5060:
BYE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK63732395;rport
From: "From Skype" <sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
Contact: <sip:3133731337@10.4.20.111>
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (fromskype, 1000, 1) exited non-zero on 'SIP/7101-081d6660'
Really destroying SIP dialog '1180028879-3040-TOPLAP@10.4.20.184' Method: BYE
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
From: "From Skype"<sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK63732395;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '734db83e575ad5da3fa724ae388d4a32@10.4.20.111' Meth                                                                                                                        od: INVITE
Itchy*CLI> core verbose 3
Verbosity was 10 and is now 3
Itchy*CLI> sip nodebug
SIP Debugging Disabled
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
BYE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK123040
To: <sip:1000@10.4.20.111>;tag=as74244df7
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 358 BYE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)

<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK123040;received=10.4.20.184;rport=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7526
To: <sip:1000@10.4.20.111>;tag=as74244df7
Call-ID: 1180028879-3040-TOPLAP@10.4.20.184
CSeq: 358 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '734db83e575ad5da3fa724ae388d4a32@10.4.20.111' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port to send to
set_destination: set destination to 10.4.20.190, port 5060
Reliably Transmitting (NAT) to 10.4.20.190:5060:
BYE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK63732395;rport
From: "From Skype" <sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
Contact: <sip:3133731337@10.4.20.111>
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (fromskype, 1000, 1) exited non-zero on 'SIP/7101-081d6660'
Really destroying SIP dialog '1180028879-3040-TOPLAP@10.4.20.184' Method: BYE
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 734db83e575ad5da3fa724ae388d4a32@10.4.20.111
From: "From Skype"<sip:3133731337@10.4.20.111>;tag=as68f1ace8
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-985a79b3
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK63732395;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '734db83e575ad5da3fa724ae388d4a32@10.4.20.111' Method: INVITE