<--- SIP read from UDP:10.0.0.128:5060 --->
REFER sip:201@10.0.0.5:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bKf6a493cb4f5c93d12
Max-Forwards: 70
From: "216" <sip:55i@10.0.0.5:5060>;tag=cc566ea55c
To: <sip:201@10.0.0.5:5060;user=phone>;tag=as79c112b2
Call-ID: cd109d4de3acce4f
CSeq: 4912 REFER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="55i",realm="asterisk",nonce="26ac1f9a",uri="sip:201@10.0.0.5:5060",response="3016b872db86be5dfd7d81364f51dcf0",algorithm=MD5
Contact: "216" <sip:55i@10.0.0.128:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D197307>"
Refer-To: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk?Replaces=341c34210c95c85c5a26695d086e36f9%40pabx.devel.voipcortex.co.uk%3bto-tag%3das1ed3dea1%3bfrom-tag%3d1002760794>
Referred-By: <sip:55i@10.0.0.5>
Supported: path
User-Agent: Aastra 55i/3.2.2.2063
Content-Length: 0


<------------->
--- (16 headers 0 lines) ---
Call cd109d4de3acce4f got a SIP call transfer from caller: (REFER)!
SIP transfer to extension 0015651f831f@local by 55i@10.0.0.5

<--- Transmitting (no NAT) to 10.0.0.128:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bKf6a493cb4f5c93d12;received=10.0.0.128
From: "216" <sip:55i@10.0.0.5:5060>;tag=cc566ea55c
To: <sip:201@10.0.0.5:5060;user=phone>;tag=as79c112b2
Call-ID: cd109d4de3acce4f
CSeq: 4912 REFER
Server: Asterisk PBX 1.8.11.2-9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:201@10.0.0.5:5060>
Content-Length: 0


<------------>
set_destination: Parsing <sip:55i@10.0.0.128:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.0.0.128:5060
Reliably Transmitting (no NAT) to 10.0.0.128:5060:
NOTIFY sip:55i@10.0.0.128:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK1abad28c
Max-Forwards: 70
From: <sip:201@10.0.0.5:5060;user=phone>;tag=as79c112b2
To: "216" <sip:55i@10.0.0.5:5060>;tag=cc566ea55c
Contact: <sip:201@10.0.0.5:5060>
Call-ID: cd109d4de3acce4f
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.11.2-9
Event: refer;id=4912
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 16

SIP/2.0 200 OK

---
set_destination: Parsing <sip:55i@10.0.0.128:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.0.0.128:5060
Audio is at 19144
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.128:5060:
INVITE sip:55i@10.0.0.128:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK6383945f
Max-Forwards: 70
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Contact: <sip:0015651f831f@10.0.0.5:5060>
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.8.11.2-9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Remote-Party-ID: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 1084297811 1084297817 IN IP4 10.0.0.5
s=Asterisk PBX 1.8.11.2-9
c=IN IP4 10.0.0.5
t=0 0
m=audio 19144 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

---
Scheduling destruction of SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' in 32000 ms (Method: ACK)
Scheduling destruction of SIP dialog 'cd109d4de3acce4f' in 32000 ms (Method: REFER)

<--- SIP read from UDP:10.0.0.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK1abad28c
From: <sip:201@10.0.0.5:5060;user=phone>;tag=as79c112b2
To: "216" <sip:55i@10.0.0.5:5060>;tag=cc566ea55c
Call-ID: cd109d4de3acce4f
CSeq: 102 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 55i/3.2.2.2063
Supported: path
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:10.0.0.128:5060 --->
BYE sip:201@10.0.0.5:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bK5cb87f9c522291850
Max-Forwards: 70
From: "216" <sip:55i@10.0.0.5:5060>;tag=cc566ea55c
To: <sip:201@10.0.0.5:5060;user=phone>;tag=as79c112b2
Call-ID: cd109d4de3acce4f
CSeq: 4913 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="55i",realm="asterisk",nonce="26ac1f9a",uri="sip:201@10.0.0.5:5060",response="677cc5608b76a4dafa5ef87f04942d3d",algorithm=MD5
Supported: path
User-Agent: Aastra 55i/3.2.2.2063
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 10.0.0.128:5060 (no NAT)
Scheduling destruction of SIP dialog 'cd109d4de3acce4f' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.0.0.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bK5cb87f9c522291850;received=10.0.0.128
From: "216" <sip:55i@10.0.0.5:5060>;tag=cc566ea55c
To: <sip:201@10.0.0.5:5060;user=phone>;tag=as79c112b2
Call-ID: cd109d4de3acce4f
CSeq: 4913 BYE
Server: Asterisk PBX 1.8.11.2-9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:10.0.0.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK6383945f
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 106 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "216" <sip:55i@10.0.0.128:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D197307>"
Server: Aastra 55i/3.2.2.2063
Supported: path, replaces
Content-Type: application/sdp
Content-Length: 229

v=0
o=MxSIP 0 7 IN IP4 10.0.0.128
s=SIP Call
c=IN IP4 10.0.0.128
t=0 0
m=audio 3000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (13 headers 12 lines) ---
set_destination: Parsing <sip:55i@10.0.0.128:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.0.0.128:5060
Reliably Transmitting (no NAT) to 10.0.0.128:5060:
BYE sip:55i@10.0.0.128:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK5e7de6c7
Max-Forwards: 70
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 107 BYE
User-Agent: Asterisk PBX 1.8.11.2-9
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
ontent-Length: 0


---
Scheduling destruction of SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' in 32000 ms (Method: ACK)
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.128:3000
set_destination: Parsing <sip:55i@10.0.0.128:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.0.0.128:5060
Transmitting (no NAT) to 10.0.0.128:5060:
ACK sip:55i@10.0.0.128:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK1e809647
Max-Forwards: 70
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Contact: <sip:0015651f831f@10.0.0.5:5060>
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 106 ACK
User-Agent: Asterisk PBX 1.8.11.2-9
Content-Length: 0


---
set_destination: Parsing <sip:55i@10.0.0.128:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.0.0.128:5060
Reliably Transmitting (no NAT) to 10.0.0.128:5060:
BYE sip:55i@10.0.0.128:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK0ce74ee4
Max-Forwards: 70
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 108 BYE
User-Agent: Asterisk PBX 1.8.11.2-9
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
ontent-Length: 0


---
Scheduling destruction of SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' in 32000 ms (Method: ACK)

<--- SIP read from UDP:10.0.0.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK5e7de6c7
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 107 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 55i/3.2.2.2063
Supported: path
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:10.0.0.128:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK0ce74ee4
From: "Line 1" <sip:0015651f831f@pabx.devel.voipcortex.co.uk>;tag=as1ed3dea1
To: <sip:55i@10.0.0.128:5060;transport=udp>;tag=1002760794
Call-ID: 341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk
CSeq: 108 BYE
Server: Aastra 55i/3.2.2.2063
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '341c34210c95c85c5a26695d086e36f9@pabx.devel.voipcortex.co.uk' Method: ACK
[Jul  3 17:45:38] NOTICE[29358]: chan_sip.c:13368 sip_reg_timeout:    -- Registration for 'slicksip@10.0.0.250' timed out, trying again (Attempt #3)