voip2*CLI> sip set debug peer username
SIP Debugging Enabled for IP: XXX.XX.XXX.XX:5060
    -- Accepting call from '6045551212' to '6045553000' on channel 0/1, span 1
    -- Executing Macro("DAHDI/1-1", "local_dids_match,6045553000,19972157")
    -- Executing [s@macro-local_dids_match:1] NoOp("DAHDI/1-1", "Advanced a2b sequence exten : 6045553000 ") in new stack
    -- Executing [s@macro-local_dids_match:2] Set("DAHDI/1-1", "__DID=6045553000") in new stack
    -- Executing [s@macro-local_dids_match:3] Gosub("DAHDI/1-1", "check_incoming_cost,6045553000,1") in new stack
    -- Executing [6045553000@check_incoming_cost:1] ExecIf("DAHDI/1-1", "0?Return") in new stack
    -- Executing [6045553000@check_incoming_cost:2] Set("DAHDI/1-1", "tariff=") in new stack
    -- Executing [6045553000@check_incoming_cost:3] ExecIf("DAHDI/1-1", "1?Return") in new stack
    -- Executing [s@macro-local_dids_match:4] Goto("DAHDI/1-1", "local_dids_match_2,6045553000,1") in new stack
    -- Goto (local_dids_match_2,6045553000,1)
    -- Executing [6045553000@local_dids_match_2:1] AGI("DAHDI/1-1", "a2billing.php,2,did") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php

[ Snip lots of output from a2billing.php ]

    -- AGI Script Executing Application: (DIAL) Options: (SIP/username,30,Hi)
  == Using SIP RTP CoS mark 5
Audio is at <serverip> port 18684
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to XXX.XX.XXX.XX:5060:
INVITE sip:username@XXX.XX.XXX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP <serverip>:5060;branch=z9hG4bK2bdc9bf9;rport
Max-Forwards: 70
From: "Lightspeed" <sip:6045551212@<serverip>>;tag=as092c5a5b
To: <sip:username@XXX.XX.XXX.XX:5060>
Contact: <sip:6045551212@<serverip>>
Call-ID: 456aa35b743a8d6464b8ddfa39db2271@<serverip>
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.17
Date: Tue, 19 Apr 2011 16:14:29 GMT
Session-Expires: 600
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1028241397 1028241397 IN IP4 <serverip>
s=Asterisk PBX 1.6.2.17
c=IN IP4 <serverip>
t=0 0
m=audio 18684 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called username

<--- SIP read from UDP:XXX.XX.XXX.XX:5060 --->
SIP/2.0 180 Ringing
From: "Lightspeed"<sip:6045551212@<serverip>>;tag=as092c5a5b
To: <sip:username@XXX.XX.XXX.XX:5060>;tag=10f1cf8-cdfaafcc-13c4-45028-479-6209c896-479
Call-ID: 456aa35b743a8d6464b8ddfa39db2271@<serverip>
CSeq: 102 INVITE
Via: SIP/2.0/UDP <serverip>:5060;rport=5060;branch=z9hG4bK2bdc9bf9
Supported: replaces
User-Agent: Thomson TG784 Build 8.2.7.7
Contact: <sip:username@XXX.XX.XXX.XX:5060>
X-Serialnumber: CP0845NT2LZ
Accept: application/dtmf-relay, x-application/dtmf-relay, application/sdp
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, INFO, OPTIONS, UPDATE
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
    -- SIP/username-00000137 is ringing
    -- Nobody picked up in 30000 ms
Scheduling destruction of SIP dialog '456aa35b743a8d6464b8ddfa39db2271@<serverip>' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to XXX.XX.XXX.XX:5060:
CANCEL sip:username@XXX.XX.XXX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP <serverip>:5060;branch=z9hG4bK2bdc9bf9;rport
Max-Forwards: 70
From: "Lightspeed" <sip:6045551212@<serverip>>;tag=as092c5a5b
To: <sip:username@XXX.XX.XXX.XX:5060>
Call-ID: 456aa35b743a8d6464b8ddfa39db2271@<serverip>
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.17
Content-Length: 0


---
Scheduling destruction of SIP dialog '456aa35b743a8d6464b8ddfa39db2271@<serverip>' in 32000 ms (Method: INVITE)
 a2billing.php,2,did: file:Class.A2Billing.php - line:1298 - uniqueid:1303229669.574 - DIAL SIP/username|30|Hi
 a2billing.php,2,did: file:Class.A2Billing.php - line:1310 - uniqueid:1303229669.574 - [SIP/username Friend][followme=1]:[ANSWEREDTIME=-DIALSTATUS=NOANSWER]
 a2billing.php,2,did: file:Class.A2Billing.php - line:1419 - uniqueid:1303229669.574 - [STATUS] CHANNEL (NOANSWER) - GOTO VOICEMAIL (9283688628)
    -- AGI Script Executing Application: (VoiceMail) Options: (9283688628,u)
    -- <DAHDI/1-1> Playing '/var/spool/asterisk/voicemail/default/9283688628/unavail.slin' (language 'en')

<--- SIP read from UDP:XXX.XX.XXX.XX:5060 --->
SIP/2.0 200 OK
From: "Lightspeed"<sip:6045551212@<serverip>>;tag=as092c5a5b
To: <sip:username@XXX.XX.XXX.XX:5060>;tag=10f1cf8-cdfaafcc-13c4-45028-479-6209c896-479
Call-ID: 456aa35b743a8d6464b8ddfa39db2271@<serverip>
CSeq: 102 CANCEL
Via: SIP/2.0/UDP <serverip>:5060;rport=5060;branch=z9hG4bK2bdc9bf9
Supported: replaces,100rel
User-Agent: Thomson TG784 Build 8.2.7.7
X-Serialnumber: CP0845NT2LZ
Accept: application/dtmf-relay, x-application/dtmf-relay, application/sdp
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, INFO, OPTIONS, UPDATE
Content-Length: 0


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

<--- SIP read from UDP:XXX.XX.XXX.XX:5060 --->
SIP/2.0 487 Request Terminated
From: "Lightspeed"<sip:6045551212@<serverip>>;tag=as092c5a5b
To: <sip:username@XXX.XX.XXX.XX:5060>;tag=10f1cf8-cdfaafcc-13c4-45028-479-6209c896-479
Call-ID: 456aa35b743a8d6464b8ddfa39db2271@<serverip>
CSeq: 102 INVITE
Via: SIP/2.0/UDP <serverip>:5060;rport=5060;branch=z9hG4bK2bdc9bf9
Supported: replaces,100rel
User-Agent: Thomson TG784 Build 8.2.7.7
X-Serialnumber: CP0845NT2LZ
Accept: application/dtmf-relay, x-application/dtmf-relay, application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to XXX.XX.XXX.XX:5060:
ACK sip:username@XXX.XX.XXX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP <serverip>:5060;branch=z9hG4bK2bdc9bf9;rport
Max-Forwards: 70
From: "Lightspeed" <sip:6045551212@<serverip>>;tag=as092c5a5b
To: <sip:username@XXX.XX.XXX.XX:5060>;tag=10f1cf8-cdfaafcc-13c4-45028-479-6209c896-479
Contact: <sip:6045551212@<serverip>>
Call-ID: 456aa35b743a8d6464b8ddfa39db2271@<serverip>
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.17
Content-Length: 0


---
Really destroying SIP dialog '456aa35b743a8d6464b8ddfa39db2271@<serverip>' Method: INVITE
    -- <DAHDI/1-1> Playing 'vm-intro.ulaw' (language 'en')
    -- <DAHDI/1-1> Playing 'beep.ulaw' (language 'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/9283688628/tmp/TGqidE format: wav49, 0xae97410
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/9283688628/tmp/TGqidE format: gsm, 0xa203988
    -- x=2, open writing:  /var/spool/asterisk/voicemail/default/9283688628/tmp/TGqidE format: wav, 0xa1e8c38
    -- Channel 0/1, span 1 got hangup request, cause 16
    -- User hung up
Scheduling destruction of SIP dialog '1ba97c382282fcf70cf1d352395a7809@<serverip>' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to XXX.XX.XXX.XX:5060:
NOTIFY sip:username@XXX.XX.XXX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP <serverip>:5060;branch=z9hG4bK1dced2a2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<serverip>>;tag=as4f3ef65f
To: <sip:username@XXX.XX.XXX.XX:5060>
Contact: <sip:asterisk@<serverip>>
Call-ID: 1ba97c382282fcf70cf1d352395a7809@<serverip>
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.6.2.17
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: yes
Message-Account: sip:default@<serverip>
Voice-Message: 2/0 (0/0)

---
 a2billing.php,2,did: file:Class.A2Billing.php - line:747 - uniqueid:1303229669.574 - [CARD STATUS UPDATE]
 a2billing.php,2,did: file:Class.A2Billing.php - line:756 - uniqueid:1303229669.574 - [QUERY USING CARD UPDATE::> UPDATE cc_card SET inuse=inuse-1, credit=credit+0.25 WHERE username='9283688628']
    -- <DAHDI/1-1>AGI Script a2billing.php completed, returning -1
    -- Hungup 'DAHDI/1-1'

<--- SIP read from UDP:XXX.XX.XXX.XX:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
From: "asterisk"<sip:asterisk@<serverip>>;tag=as4f3ef65f
To: <sip:username@XXX.XX.XXX.XX:5060>;tag=10f2098-cdfaafcc-13c4-45028-4ac-29d39170-4ac
Call-ID: 1ba97c382282fcf70cf1d352395a7809@<serverip>
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP <serverip>:5060;rport=5060;branch=z9hG4bK1dced2a2
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, INFO, OPTIONS, UPDATE
User-Agent: Thomson TG784 Build 8.2.7.7
X-Serialnumber: CP0845NT2LZ
Accept: application/dtmf-relay, x-application/dtmf-relay, application/sdp
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '1ba97c382282fcf70cf1d352395a7809@<serverip>' Method: NOTIFY