<--- SIP read from UDP://10.0.42.231:1024 --->
INVITE sip:204@10.10.23.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-oghu2s6emve6;rport
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1
X-Serialnumber: 000413401713
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.11
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 366

v=0
o=root 436322610 436322610 IN IP4 10.0.42.231
s=call
c=IN IP4 10.0.42.231
t=0 0
m=audio 50476 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
--- (19 headers 17 lines) ---
  == Using SIP RTP TOS bits 24
  == Using SIP RTP CoS mark 5
Sending to 10.0.42.231 : 1024 (no NAT)
Using INVITE request as basis request - 3c27ceecf38a-50kofsvqf788
Found user '202-000413401713-1' for '202-000413401713-1'

<--- Reliably Transmitting (no NAT) to 10.0.42.231:1024 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-oghu2s6emve6;received=10.0.42.231;rport=1024
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>;tag=as1273b7a6
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="660182df"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c27ceecf38a-50kofsvqf788' in 32000 ms (Method: INVITE)
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.231:1024 --->
ACK sip:204@10.10.23.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-oghu2s6emve6;rport
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>;tag=as1273b7a6
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.231:1024 --->
INVITE sip:204@10.10.23.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-uiy4i5waydqw;rport
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1
X-Serialnumber: 000413401713
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom820/8.2.11
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username="202-000413401713-1",realm="asterisk",nonce="660182df",uri="sip:204@10.10.23.4;user=phone",response="6c730d5be4844b9e40fcf59a37b5f3e0",algorithm=MD5
Content-Type: application/sdp
Content-Length: 366

v=0
o=root 436322610 436322610 IN IP4 10.0.42.231
s=call
c=IN IP4 10.0.42.231
t=0 0
m=audio 50476 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
--- (20 headers 17 lines) ---
Sending to 10.0.42.231 : 1024 (no NAT)
Using INVITE request as basis request - 3c27ceecf38a-50kofsvqf788
Found user '202-000413401713-1' for '202-000413401713-1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 99
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 10.0.42.231:50476
Found audio description format pcmu for ID 0
Found audio description format pcma for ID 8
Found audio description format g722 for ID 9
Found audio description format g726-32 for ID 99
Found audio description format gsm for ID 3
Found audio description format g729 for ID 18
Found audio description format g723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.42.231:50476
Looking for 204 in Internal (domain 10.10.23.4)
list_route: hop: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>

<--- Transmitting (no NAT) to 10.0.42.231:1024 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-uiy4i5waydqw;received=10.0.42.231;rport=1024
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:204@10.10.23.4>
Content-Length: 0


<------------>
    -- Executing [204@Internal:1] Set("SIP/202-000413401713-1-14085698", "CallFlowID=204") in new stack
    -- Executing [204@Internal:2] DumpChan("SIP/202-000413401713-1-14085698", "") in new stack
vpbx03*CLI>
Dumping Info For Channel: SIP/202-000413401713-1-14085698:
================================================================================
Info:
Name=               SIP/202-000413401713-1-14085698
Type=               SIP
UniqueID=           1254228283.54
CallerIDNum=        202
CallerIDName=       Snom 820
DNIDDigits=         204
RDNIS=              (N/A)
Language=           da
State=              Ring (4)
Rings=              0
NativeFormat=       0x8 (alaw)
WriteFormat=        0x8 (alaw)
ReadFormat=         0x8 (alaw)
RawWriteFormat=     0x8 (alaw)
RawReadFormat=      0x8 (alaw)
1stFileDescriptor=  19
Framesin=           0
Framesout=          0
TimetoHangup=       0
ElapsedTime=        0h0m0s
Context=            Internal
Extension=          204
Priority=           2
CallGroup=
PickupGroup=
Application=        DumpChan
Data=               (Empty)
Blocking_in=        (Not Blocking)

Variables:
CallFlowID=204
SIPCALLID=3c27ceecf38a-50kofsvqf788
SIPDOMAIN=10.10.23.4
SIPURI=sip:202-000413401713-1@10.0.42.231:1024
================================================================================
    -- Executing [204@Internal:3] Goto("SIP/202-000413401713-1-14085698", "DemoDial,s,1") in new stack
    -- Goto (DemoDial,s,1)
    -- Executing [s@DemoDial:1] NoOp("SIP/202-000413401713-1-14085698", "DemoDial") in new stack
    -- Executing [s@DemoDial:2] Dial("SIP/202-000413401713-1-14085698", "Local/204@DialAccount,10,") in new stack
    -- Called 204@DialAccount
    -- Executing [204@DialAccount:1] Set("Local/204@DialAccount-817c;2", "CallFlowID=204") in new stack
    -- Executing [204@DialAccount:2] ExecIf("Local/204@DialAccount-817c;2", "0?Hangup(UNALLOCATED)") in new stack
    -- Executing [204@DialAccount:3] GotoIf("Local/204@DialAccount-817c;2", "0?Dial") in new stack
    -- Executing [204@DialAccount:4] GotoIf("Local/204@DialAccount-817c;2", "0?Dial") in new stack
    -- Executing [204@DialAccount:5] GotoIf("Local/204@DialAccount-817c;2", "1?Dial") in new stack
    -- Goto (DialAccount,204,9)
    -- Executing [204@DialAccount:9] Dial("Local/204@DialAccount-817c;2", "Local/204-0004132311d4-1@DialLine/n") in new stack
    -- Executing [204-0004132311d4-1@DialLine:1] Dial("Local/204-0004132311d4-1@DialLine-8448;2", "SIP/204-0004132311d4-1") in new stack
  == Using SIP RTP TOS bits 24
  == Using SIP RTP CoS mark 5
Audio is at 10.10.23.4 port 10456
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.42.233:2048:
INVITE sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x SIP/2.0
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK0c27d4ca;rport
Max-Forwards: 70
From: "Snom 820" <sip:202@10.10.23.4>;tag=as0ee8ea48
To: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>
Contact: <sip:202@10.10.23.4>
Call-ID: 172a04e92cd4489d01eeae4c2292d58a@10.10.23.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.16-rc1
Date: Tue, 29 Sep 2009 12:44:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 336583191 336583191 IN IP4 10.10.23.4
s=Asterisk PBX 1.6.0.16-rc1
c=IN IP4 10.10.23.4
t=0 0
m=audio 10456 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 204-0004132311d4-1
    -- Called 204-0004132311d4-1@DialLine/n
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.233:2048 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK0c27d4ca;rport=5060
From: "Snom 820" <sip:202@10.10.23.4>;tag=as0ee8ea48
To: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>;tag=rc8nxd5to6
Call-ID: 172a04e92cd4489d01eeae4c2292d58a@10.10.23.4
CSeq: 102 INVITE
Contact: <sip:200@10.10.23.4;user=phone>
Diversion: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>;reason="unconditional"
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 10.0.42.233
Transmitting (no NAT) to 10.0.42.233:2048:
ACK sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x SIP/2.0
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK0c27d4ca;rport
Max-Forwards: 70
From: "Snom 820" <sip:202@10.10.23.4>;tag=as0ee8ea48
To: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>;tag=rc8nxd5to6
Contact: <sip:202@10.10.23.4>
Call-ID: 172a04e92cd4489d01eeae4c2292d58a@10.10.23.4
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.16-rc1
Content-Length: 0


---
    -- Now forwarding Local/204-0004132311d4-1@DialLine-8448;2 to 'Local/200@Internal' (thanks to SIP/204-0004132311d4-1-f8002958)
    -- Executing [200@Internal:1] Set("Local/200@Internal-2be0;2", "CallFlowID=200") in new stack
    -- Local/204-0004132311d4-1@DialLine-8448;1 stopped sounds
    -- Local/204@DialAccount-817c;1 stopped sounds
    -- Executing [200@Internal:2] DumpChan("Local/200@Internal-2be0;2", "") in new stack
vpbx03*CLI>
Dumping Info For Channel: Local/200@Internal-2be0;2:
================================================================================
Info:
Name=               Local/200@Internal-2be0;2
Type=               Local
UniqueID=           1254228283.61
CallerIDNum=        202
CallerIDName=       Snom 820
DNIDDigits=         (N/A)
RDNIS=              204-0004132311d4-1
Language=           en
State=              Ring (4)
Rings=              0
NativeFormat=       0x8 (alaw)
WriteFormat=        0x8 (alaw)
ReadFormat=         0x8 (alaw)
RawWriteFormat=     0x8 (alaw)
RawReadFormat=      0x8 (alaw)
1stFileDescriptor=  -1
Framesin=           0
Framesout=          0
TimetoHangup=       0
ElapsedTime=        0h0m0s
Context=            Internal
Extension=          200
Priority=           2
CallGroup=
PickupGroup=
Application=        DumpChan
Data=               (Empty)
Blocking_in=        (Not Blocking)

Variables:
CallFlowID=200
================================================================================
    -- Executing [200@Internal:3] Goto("Local/200@Internal-2be0;2", "DemoDial,s,1") in new stack
    -- Goto (DemoDial,s,1)
    -- Executing [s@DemoDial:1] NoOp("Local/200@Internal-2be0;2", "DemoDial") in new stack
    -- Executing [s@DemoDial:2] Dial("Local/200@Internal-2be0;2", "Local/200@DialAccount,10,") in new stack
    -- Called 200@DialAccount
    -- Executing [200@DialAccount:1] Set("Local/200@DialAccount-c0d6;2", "CallFlowID=200") in new stack
    -- Executing [200@DialAccount:2] ExecIf("Local/200@DialAccount-c0d6;2", "0?Hangup(UNALLOCATED)") in new stack
    -- Executing [200@DialAccount:3] GotoIf("Local/200@DialAccount-c0d6;2", "0?Dial") in new stack
    -- Executing [200@DialAccount:4] GotoIf("Local/200@DialAccount-c0d6;2", "0?Dial") in new stack
    -- Executing [200@DialAccount:5] GotoIf("Local/200@DialAccount-c0d6;2", "1?Dial") in new stack
    -- Goto (DialAccount,200,9)
    -- Executing [200@DialAccount:9] Dial("Local/200@DialAccount-c0d6;2", "Local/200-0016b6915c2f-1@DialLine/n") in new stack
    -- Called 200-0016b6915c2f-1@DialLine/n
    -- Executing [200-0016b6915c2f-1@DialLine:1] Dial("Local/200-0016b6915c2f-1@DialLine-0cee;2", "SIP/200-0016b6915c2f-1") in new stack
  == Using SIP RTP TOS bits 24
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '179d2c2844139618567d6b9215fd2ddf@10.10.23.4' Method: INVITE
[Sep 29 14:44:43] WARNING[9678]: app_dial.c:1499 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [200-0016b6915c2f-1@DialLine:2] Goto("Local/200-0016b6915c2f-1@DialLine-0cee;2", "DialStatus,s,1") in new stack
    -- Goto (DialStatus,s,1)
    -- Executing [s@DialStatus:1] ExecIf("Local/200-0016b6915c2f-1@DialLine-0cee;2", "0?Hangup") in new stack
    -- Executing [s@DialStatus:2] ExecIf("Local/200-0016b6915c2f-1@DialLine-0cee;2", "0?Busy(5)") in new stack
    -- Executing [s@DialStatus:3] ExecIf("Local/200-0016b6915c2f-1@DialLine-0cee;2", "0?Hangup") in new stack
    -- Executing [s@DialStatus:4] ExecIf("Local/200-0016b6915c2f-1@DialLine-0cee;2", "0?Hangup") in new stack
    -- Executing [s@DialStatus:5] ExecIf("Local/200-0016b6915c2f-1@DialLine-0cee;2", "0?Busy(5)") in new stack
    -- Executing [s@DialStatus:6] ExecIf("Local/200-0016b6915c2f-1@DialLine-0cee;2", "1?Hangup(1)") in new stack
  == Spawn extension (DialStatus, s, 6) exited non-zero on 'Local/200-0016b6915c2f-1@DialLine-0cee;2'
    -- Executing [h@DialStatus:1] Set("Local/200-0016b6915c2f-1@DialLine-0cee;2", "CDR(hangupcause)=1") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'Local/200@DialAccount-c0d6;2' status is 'CHANUNAVAIL'
    -- Local/200@DialAccount-c0d6;1 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@DemoDial:3] GotoIf("Local/200@Internal-2be0;2", "0?Busy") in new stack
    -- Executing [s@DemoDial:4] Hangup("Local/200@Internal-2be0;2", "") in new stack
  == Spawn extension (DemoDial, s, 4) exited non-zero on 'Local/200@Internal-2be0;2'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [204-0004132311d4-1@DialLine:2] Goto("Local/204-0004132311d4-1@DialLine-8448;2", "DialStatus,s,1") in new stack
    -- Goto (DialStatus,s,1)
    -- Executing [s@DialStatus:1] ExecIf("Local/204-0004132311d4-1@DialLine-8448;2", "0?Hangup") in new stack
    -- Executing [s@DialStatus:2] ExecIf("Local/204-0004132311d4-1@DialLine-8448;2", "0?Busy(5)") in new stack
    -- Executing [s@DialStatus:3] ExecIf("Local/204-0004132311d4-1@DialLine-8448;2", "0?Hangup") in new stack
    -- Executing [s@DialStatus:4] ExecIf("Local/204-0004132311d4-1@DialLine-8448;2", "0?Hangup") in new stack
    -- Executing [s@DialStatus:5] ExecIf("Local/204-0004132311d4-1@DialLine-8448;2", "0?Busy(5)") in new stack
    -- Executing [s@DialStatus:6] ExecIf("Local/204-0004132311d4-1@DialLine-8448;2", "1?Hangup(1)") in new stack
  == Spawn extension (DialStatus, s, 6) exited non-zero on 'Local/204-0004132311d4-1@DialLine-8448;2'
    -- Executing [h@DialStatus:1] Set("Local/204-0004132311d4-1@DialLine-8448;2", "CDR(hangupcause)=1") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'Local/204@DialAccount-817c;2' status is 'CHANUNAVAIL'
    -- Local/204@DialAccount-817c;1 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@DemoDial:3] GotoIf("SIP/202-000413401713-1-14085698", "0?Busy") in new stack
    -- Executing [s@DemoDial:4] Hangup("SIP/202-000413401713-1-14085698", "") in new stack
  == Spawn extension (DemoDial, s, 4) exited non-zero on 'SIP/202-000413401713-1-14085698'
Scheduling destruction of SIP dialog '3c27ceecf38a-50kofsvqf788' in 32000 ms (Method: INVITE)
vpbx03*CLI>
<--- Reliably Transmitting (no NAT) to 10.0.42.231:1024 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-uiy4i5waydqw;received=10.0.42.231;rport=1024
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>;tag=as48a292bd
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.231:1024 --->
ACK sip:204@10.10.23.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-uiy4i5waydqw;rport
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=s8gebdg2mc
To: <sip:204@10.10.23.4;user=phone>;tag=as48a292bd
Call-ID: 3c27ceecf38a-50kofsvqf788
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '172a04e92cd4489d01eeae4c2292d58a@10.10.23.4' Method: INVITE
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.231:1024 --->
REGISTER sip:10.10.23.4 SIP/2.0
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-43bj96h0av2s;rport
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=kjw2697eih
To: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>
Call-ID: 3c26701e8a2a-zrkpsicqqgwf
CSeq: 5996 REGISTER
Max-Forwards: 70
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:530eb2bd-d208-4f70-ba2d-39076f46ef09>";audio;mobility="fixed";duplex="full";description="snom820";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom820/8.2.11
Supported: gruu
Allow-Events: dialog
X-Real-IP: 10.0.42.231
Expires: 60
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.42.231 : 1024 (no NAT)

<--- Transmitting (no NAT) to 10.0.42.231:1024 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-43bj96h0av2s;received=10.0.42.231;rport=1024
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=kjw2697eih
To: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=as3972630c
Call-ID: 3c26701e8a2a-zrkpsicqqgwf
CSeq: 5996 REGISTER
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32930f19"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c26701e8a2a-zrkpsicqqgwf' in 32000 ms (Method: REGISTER)
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.231:1024 --->
REGISTER sip:10.10.23.4 SIP/2.0
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-uamc5s0s3j1a;rport
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=kjw2697eih
To: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>
Call-ID: 3c26701e8a2a-zrkpsicqqgwf
CSeq: 5997 REGISTER
Max-Forwards: 70
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:530eb2bd-d208-4f70-ba2d-39076f46ef09>";audio;mobility="fixed";duplex="full";description="snom820";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom820/8.2.11
Supported: gruu
Allow-Events: dialog
X-Real-IP: 10.0.42.231
Authorization: Digest username="202-000413401713-1",realm="asterisk",nonce="32930f19",uri="sip:10.10.23.4",response="8b5e4061badeadc7de6419f9715a3e45",algorithm=MD5
Expires: 60
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Sending to 10.0.42.231 : 1024 (no NAT)
Reliably Transmitting (no NAT) to 10.0.42.231:1024:
OPTIONS sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e SIP/2.0
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK246cfa2d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.23.4>;tag=as472afc6e
To: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>
Contact: <sip:asterisk@10.10.23.4>
Call-ID: 092f4ec20314e8dc585fee07697ddf57@10.10.23.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.16-rc1
Date: Tue, 29 Sep 2009 12:44:43 GMT
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
vpbx03*CLI>
<--- Transmitting (no NAT) to 10.0.42.231:1024 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.42.231:1024;branch=z9hG4bK-uamc5s0s3j1a;received=10.0.42.231;rport=1024
From: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=kjw2697eih
To: "202 Snom 820" <sip:202-000413401713-1@10.10.23.4>;tag=as3972630c
Call-ID: 3c26701e8a2a-zrkpsicqqgwf
CSeq: 5997 REGISTER
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e;expires=60
Date: Tue, 29 Sep 2009 12:44:43 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c26701e8a2a-zrkpsicqqgwf' in 32000 ms (Method: REGISTER)
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.231:1024 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK246cfa2d;rport=5060
From: "asterisk" <sip:asterisk@10.10.23.4>;tag=as472afc6e
To: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>
Call-ID: 092f4ec20314e8dc585fee07697ddf57@10.10.23.4
CSeq: 102 OPTIONS
Contact: <sip:202-000413401713-1@10.0.42.231:1024;line=z1ot3r2e>;reg-id=1
User-Agent: snom820/8.2.11
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '092f4ec20314e8dc585fee07697ddf57@10.10.23.4' Method: OPTIONS
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.233:2048 --->
REGISTER sip:10.10.23.4 SIP/2.0
Via: SIP/2.0/UDP 10.0.42.233:2048;branch=z9hG4bK-42914t5l6hee;rport
From: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>;tag=83e04fs0ym
To: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>
Call-ID: 3c267019c627-j16cdwiqikwg
CSeq: 5963 REGISTER
Max-Forwards: 70
Contact: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:7992da1b-ec1b-4d2c-829e-3e206fc113b3>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom360/7.3.14
Supported: gruu
Allow-Events: dialog
X-Real-IP: 10.0.42.233
Expires: 60
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.42.233 : 2048 (no NAT)

<--- Transmitting (no NAT) to 10.0.42.233:2048 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.42.233:2048;branch=z9hG4bK-42914t5l6hee;received=10.0.42.233;rport=2048
From: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>;tag=83e04fs0ym
To: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>;tag=as751adedf
Call-ID: 3c267019c627-j16cdwiqikwg
CSeq: 5963 REGISTER
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d00d4fe"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c267019c627-j16cdwiqikwg' in 32000 ms (Method: REGISTER)
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.233:2048 --->
REGISTER sip:10.10.23.4 SIP/2.0
Via: SIP/2.0/UDP 10.0.42.233:2048;branch=z9hG4bK-ze47wmcivdn8;rport
From: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>;tag=83e04fs0ym
To: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>
Call-ID: 3c267019c627-j16cdwiqikwg
CSeq: 5964 REGISTER
Max-Forwards: 70
Contact: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:7992da1b-ec1b-4d2c-829e-3e206fc113b3>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom360/7.3.14
Supported: gruu
Allow-Events: dialog
X-Real-IP: 10.0.42.233
Authorization: Digest username="204-0004132311d4-1",realm="asterisk",nonce="3d00d4fe",uri="sip:10.10.23.4",response="2632cdaac3c4eb196342318ddfc04043",algorithm=MD5
Expires: 60
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Sending to 10.0.42.233 : 2048 (no NAT)
Reliably Transmitting (no NAT) to 10.0.42.233:2048:
OPTIONS sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x SIP/2.0
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK46a21e74;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.23.4>;tag=as051d2348
To: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>
Contact: <sip:asterisk@10.10.23.4>
Call-ID: 499c7c7746d30ff14ae258e52102b5a3@10.10.23.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.16-rc1
Date: Tue, 29 Sep 2009 12:44:44 GMT
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
vpbx03*CLI>
<--- Transmitting (no NAT) to 10.0.42.233:2048 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.42.233:2048;branch=z9hG4bK-ze47wmcivdn8;received=10.0.42.233;rport=2048
From: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>;tag=83e04fs0ym
To: "204 Snom 360" <sip:204-0004132311d4-1@10.10.23.4>;tag=as751adedf
Call-ID: 3c267019c627-j16cdwiqikwg
CSeq: 5964 REGISTER
User-Agent: Asterisk PBX 1.6.0.16-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x;expires=60
Date: Tue, 29 Sep 2009 12:44:44 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c267019c627-j16cdwiqikwg' in 32000 ms (Method: REGISTER)
vpbx03*CLI>
<--- SIP read from UDP://10.0.42.233:2048 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.23.4:5060;branch=z9hG4bK46a21e74;rport=5060
From: "asterisk" <sip:asterisk@10.10.23.4>;tag=as051d2348
To: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>
Call-ID: 499c7c7746d30ff14ae258e52102b5a3@10.10.23.4
CSeq: 102 OPTIONS
Contact: <sip:204-0004132311d4-1@10.0.42.233:2048;line=5x3je26x>;reg-id=1
User-Agent: snom360/7.3.14
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '499c7c7746d30ff14ae258e52102b5a3@10.10.23.4' Method: OPTIONS
Really destroying SIP dialog '3c27ced0d3d9-n877cgyqyx6l' Method: ACK