<-- SIP read from 10.0.0.103:5060:
INVITE sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-1b363930
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
Expires: 240
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 239841 239841 IN IP4 10.0.0.103
s=-
c=IN IP4 10.0.0.103
t=0 0
m=audio 16388 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (14 headers 18 lines) ---
Using INVITE request as basis request - d4938ff0-dad10c30@10.0.0.103
Sending to 10.0.0.103 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-1b363930;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>;tag=as37bd59c5
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27bec94b"
Content-Length: 0


---
Scheduling destruction of call 'd4938ff0-dad10c30@10.0.0.103' in 15000 ms
Found user 'chrivers'

<-- SIP read from 10.0.0.103:5060:
ACK sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-1b363930
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>;tag=as37bd59c5
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0


--- (10 headers 0 lines) ---

<-- SIP read from 10.0.0.103:5060:
INVITE sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-cc50400c
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="27bec94b",uri="sip:1234@10.0.0.3",algorithm=MD5,response="a6342334397e9dee1928747b28f7d810"
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
Expires: 240
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 239841 239841 IN IP4 10.0.0.103
s=-
c=IN IP4 10.0.0.103
t=0 0
m=audio 16388 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 18 lines) ---
Using INVITE request as basis request - d4938ff0-dad10c30@10.0.0.103
Sending to 10.0.0.103 : 5060 (non-NAT)
Found user 'chrivers'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.103:16388
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 1234 in demo (domain 10.0.0.3)
list_route: hop: <sip:chrivers@10.0.0.103:5060>
Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-cc50400c;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1234@10.0.0.3>
Content-Length: 0


---
    -- Executing Playback("SIP/chrivers-0818f320", "transfer|skip") in new stack
    -- Executing Macro("SIP/chrivers-0818f320", "stdexten|1234|Console/dsp") in new stack
    -- Executing Dial("SIP/chrivers-0818f320", "Console/dsp|20") in new stack
Nov  6 02:27:12 WARNING[8845]: channel.c:2597 ast_request: No channel type registered for 'Console'
Nov  6 02:27:12 NOTICE[8845]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Console' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Goto("SIP/chrivers-0818f320", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
    -- Executing Goto("SIP/chrivers-0818f320", "s-NOANSWER|1") in new stack
    -- Goto (macro-stdexten,s-NOANSWER,1)
    -- Executing VoiceMail("SIP/chrivers-0818f320", "u1234") in new stack
We're at 10.0.0.3 port 10658
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-cc50400c;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>;tag=as363d643b
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1234@10.0.0.3>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 8776 8776 IN IP4 10.0.0.3
s=session
c=IN IP4 10.0.0.3
t=0 0
m=audio 10658 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Playing 'vm-theperson' (language 'en')

<-- SIP read from 10.0.0.103:5060:
ACK sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-4580bac8
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>;tag=as363d643b
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="27bec94b",uri="sip:1234@10.0.0.3",algorithm=MD5,response="a6342334397e9dee1928747b28f7d810"
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0


--- (11 headers 0 lines) ---

*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
<-- SIP read from 10.0.0.103:5060:
BYE sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-167e211f
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>;tag=as363d643b
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="27bec94b",uri="sip:1234@10.0.0.3",algorithm=MD5,response="b73760e0454aa85e909a454ef7afe704"
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0


--- (10 headers 0 lines) ---
Sending to 10.0.0.103 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-167e211f;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=ca37bcf0eb32c330o0
To: <sip:1234@10.0.0.3>;tag=as363d643b
Call-ID: d4938ff0-dad10c30@10.0.0.103
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1234@10.0.0.3>
Content-Length: 0
X-Asterisk-HangupCause: Channel not implemented


---
  == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/chrivers-0818f320' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/chrivers-0818f320'
Destroying call 'd4938ff0-dad10c30@10.0.0.103'

*CLI>