--- (14 headers 14 lines)---
Using INVITE request as basis request - rsjaxvshzylyvwa@192.168.0.163
Sending to 192.168.0.163 : 5060 (NAT)
Found user 'twinkle'
Found RTP audio format 3
Found RTP audio format 99
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.163:8000
Found description format GSM
Found description format speex
Found description format speex
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 0654306488 in default (domain 192.168.0.1)
list_route: hop: <sip:twinkle@192.168.0.163>
Transmitting (NAT) to 192.168.0.163:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.163;branch=z9hG4bKehvdftdh;received=192.168.0.163;rport=5060
From: "Gerben Meijer" <sip:twinkle@192.168.0.1>;tag=ckngk
To: <sip:0654306488@192.168.0.1>
Call-ID: rsjaxvshzylyvwa@192.168.0.163
CSeq: 178 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0654306488@192.168.0.1>
Content-Length: 0


---
    -- Executing SetAccount("SIP/twinkle-08194aa0", "31767110244") in new stack
    -- Executing SetCallerID("SIP/twinkle-08194aa0", "31767110244") in new stack
    -- Executing SetCIDName("SIP/twinkle-08194aa0", "Gerben Meijer") in new stack
    -- Executing SetCIDNum("SIP/twinkle-08194aa0", "31767110244") in new stack
    -- Executing Dial("SIP/twinkle-08194aa0", "SIP/0654306488@31767110244") in new stack
Destroying call '5ba55a6a31a1fcf233a91068711caa0d@127.0.0.1'
Nov  9 02:18:24 NOTICE[9056]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("SIP/twinkle-08194aa0", "") in new stack
  == Spawn extension (default, 0654306488, 6) exited non-zero on 'SIP/twinkle-08194aa0'
Nov  9 02:18:24 ERROR[9056]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or directory
Reliably Transmitting (NAT) to 192.168.0.163:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.163;branch=z9hG4bKehvdftdh;received=192.168.0.163;rport=5060
From: "Gerben Meijer" <sip:twinkle@192.168.0.1>;tag=ckngk
To: <sip:0654306488@192.168.0.1>;tag=as65f2a4b9
Call-ID: rsjaxvshzylyvwa@192.168.0.163
CSeq: 178 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0654306488@192.168.0.1>
Content-Length: 0


---
spirit*CLI> 
<-- SIP read from 192.168.0.163:5060: 
ACK sip:0654306488@192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.163;rport;branch=z9hG4bKehvdftdh
Max-Forwards: 70
Proxy-Authorization: Digest username="twinkle",realm="asterisk",nonce="2048cc1f",uri="sip:0654306488@192.168.0.1",response="effaee0193739ccdd94a12aa28de02ef",algorithm=MD5
To: <sip:0654306488@192.168.0.1>;tag=as65f2a4b9
From: "Gerben Meijer" <sip:twinkle@192.168.0.1>;tag=ckngk
Call-ID: rsjaxvshzylyvwa@192.168.0.163
CSeq: 178 ACK
User-Agent: Twinkle/0.8.1
Content-Length: 0


--- (10 headers 0 lines)---
Destroying call 'rsjaxvshzylyvwa@192.168.0.163'
spirit*CLI>