Reliably Transmitting (NAT) to 65.105.249.89:15790:
NOTIFY sip:17528@192.168.0.74:15790 SIP/2.0
Via: SIP/2.0/UDP 209.251.54.82:5060;branch=z9hG4bK52f5d96e;rport
From: "Unavailable"<sip:17528@pbx.testcorp.com>;tag=c0574656
To: "Unavailable"<sip:17528@pbx.testcorp.com>;tag=c0574656
Contact: <sip:Unknown@209.251.54.82>
Call-ID: 2a41f3322b14033bZGU0ZWY4ZWFhYjE3NGFlNWQwMTNjNGQ3ZWNiYjI2Yjg.
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: terminated;reason=timeout
Content-Length: 0


---
pbx*CLI>
<--- SIP read from 65.105.249.89:15790 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.251.54.82:5060;branch=z9hG4bK52f5d96e;rport=5060
Contact: <sip:17528@192.168.0.74:15790>
To: "Unavailable"<sip:17528@pbx.testcorp.com>;tag=c0574656
From: "Unavailable"<sip:17528@pbx.testcorp.com>;tag=c0574656
Call-ID: 2a41f3322b14033bZGU0ZWY4ZWFhYjE3NGFlNWQwMTNjNGQ3ZWNiYjI2Yjg.
CSeq: 104 NOTIFY
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2a41f3322b14033bZGU0ZWY4ZWFhYjE3NGFlNWQwMTNjNGQ3ZWNiYjI2Yjg.' Method: SUBSCRIBE
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
pbx*CLI>
<--- SIP read from 76.180.27.4:17444 --->
INVITE sip:7168444444@pbx.testcorp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.42.8:17444;branch=z9hG4bK-d87543-fb58d266ae6cfe06-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:10500@76.180.27.4:17444>
To: "7168444444"<sip:7168444444@pbx.testcorp.com>
From: "Unavailable"<sip:10500@pbx.testcorp.com>;tag=a1376d4d
Call-ID: ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 443

v=0
o=- 9 2 IN IP4 192.168.42.8
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.42.8
t=0 0
m=audio 42104 RTP/AVP 107 100 106 6 0 105 18 3 5 101
a=alt:1 1 : iVIkNlgf aDnbfLoq 192.168.42.8 42104
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:E533EA13E80B4AD3B8D0C123A7292C53

<------------->
--- (12 headers 16 lines) ---
Sending to 76.180.27.4 : 17444 (NAT)
Using INVITE request as basis request - ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
pbx*CLI>
<--- Reliably Transmitting (NAT) to 76.180.27.4:17444 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.42.8:17444;branch=z9hG4bK-d87543-fb58d266ae6cfe06-1--d87543-;received=76.180.27.4;rport=17444
From: "Unavailable"<sip:10500@pbx.testcorp.com>;tag=a1376d4d
To: "7168444444"<sip:7168444444@pbx.testcorp.com>;tag=as3d7f47f7
Call-ID: ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16cf36ca"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.' in 32000 ms (Method: INVITE)
Found user '10500'
pbx*CLI>
<--- SIP read from 76.180.27.4:17444 --->
ACK sip:7168444444@pbx.testcorp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.42.8:17444;branch=z9hG4bK-d87543-fb58d266ae6cfe06-1--d87543-;rport
To: "7168444444"<sip:7168444444@pbx.testcorp.com>;tag=as3d7f47f7
From: "Unavailable"<sip:10500@pbx.testcorp.com>;tag=a1376d4d
Call-ID: ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 76.180.27.4:17444 --->
INVITE sip:7168444444@pbx.testcorp.com SIP/2.0
Via: SIP/2.0/UDP 192.168.42.8:17444;branch=z9hG4bK-d87543-ab411f2bb66ccf71-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:10500@76.180.27.4:17444>
To: "7168444444"<sip:7168444444@pbx.testcorp.com>
From: "Unavailable"<sip:10500@pbx.testcorp.com>;tag=a1376d4d
Call-ID: ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="10500",realm="asterisk",nonce="16cf36ca",uri="sip:7168444444@pbx.testcorp.com",response="b771fab42e62ed2e72e1346502ab8afb",algorithm=MD5
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 443

v=0
o=- 9 2 IN IP4 192.168.42.8
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.42.8
t=0 0
m=audio 42104 RTP/AVP 107 100 106 6 0 105 18 3 5 101
a=alt:1 1 : iVIkNlgf aDnbfLoq 192.168.42.8 42104
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:E533EA13E80B4AD3B8D0C123A7292C53

<------------->
--- (13 headers 16 lines) ---
Sending to 76.180.27.4 : 17444 (NAT)
Using INVITE request as basis request - ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
Found user '10500'
Found RTP audio format 107
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 192.168.42.8:42104
Got unsupported a:fmtp in SDP offer
Got unsupported a:fmtp in SDP offer
Found unknown media description format BV32 for ID 107
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x326 (gsm|ulaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.42.8:42104
Looking for 7168444444 in from-internal (domain pbx.testcorp.com)
list_route: hop: <sip:10500@76.180.27.4:17444>
pbx*CLI>
<--- Transmitting (NAT) to 76.180.27.4:17444 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.42.8:17444;branch=z9hG4bK-d87543-ab411f2bb66ccf71-1--d87543-;received=76.180.27.4;rport=17444
From: "Unavailable"<sip:10500@pbx.testcorp.com>;tag=a1376d4d
To: "7168444444"<sip:7168444444@pbx.testcorp.com>
Call-ID: ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7168444444@209.251.54.82>
Content-Length: 0


<------------>
    -- Executing [7168444444@from-internal:1] Macro("SIP/10500-054f80b0", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/10500-054f80b0", "AMPUSER=10500") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/10500-054f80b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/10500-054f80b0", "1|Set|REALCALLERIDNUM=10500") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/10500-054f80b0", "AMPUSER=10500") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/10500-054f80b0", "AMPUSERCIDNAME=Ken") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/10500-054f80b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/10500-054f80b0", "AMPUSERCID=10500") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/10500-054f80b0", "CALLERID(all)="Ken" <10500>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/10500-054f80b0", "REALCALLERIDNUM=10500") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/10500-054f80b0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/10500-054f80b0", "Using CallerID "Ken" <10500>") in new stack
    -- Executing [7168444444@from-internal:2] Set("SIP/10500-054f80b0", "_NODEST=") in new stack
    -- Executing [7168444444@from-internal:3] Macro("SIP/10500-054f80b0", "record-enable|10500|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/10500-054f80b0", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/10500-054f80b0", "recordingcheck|20100125-154810|1264452490.42409") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20100125-154810|1264452490.42409: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/10500-054f80b0", "") in new stack
    -- Executing [7168444444@from-internal:4] Macro("SIP/10500-054f80b0", "dialout-trunk|6|7168444444||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/10500-054f80b0", "DIAL_TRUNK=6") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/10500-054f80b0", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/10500-054f80b0", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/10500-054f80b0", "DIAL_NUMBER=7168444444") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/10500-054f80b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/10500-054f80b0", "OUTBOUND_GROUP=OUT_6") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/10500-054f80b0", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/10500-054f80b0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/10500-054f80b0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/10500-054f80b0", "outbound-callerid|6") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/10500-054f80b0", "0|SetCallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/10500-054f80b0", "0|Set|REALCALLERIDNUM=10500") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/10500-054f80b0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/10500-054f80b0", "USEROUTCID=8775399066") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/10500-054f80b0", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/10500-054f80b0", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/10500-054f80b0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/10500-054f80b0", "0|Set|CALLERID(all)=") in new stack
    -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/10500-054f80b0", "0?exit") in new stack
    -- Executing [s@macro-outbound-callerid:14] Set("SIP/10500-054f80b0", "CALLERID(all)=8775399066") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/10500-054f80b0", "0|SetCallerPres|prohib_passed_screen") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/10500-054f80b0", "0|AGI|fixlocalprefix") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/10500-054f80b0", "OUTNUM=7168444444") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/10500-054f80b0", "custom=SIP/voicetrading") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/10500-054f80b0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/10500-054f80b0", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/10500-054f80b0", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/10500-054f80b0", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/10500-054f80b0", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/10500-054f80b0", "SIP/voicetrading/7168444444|300|") in new stack
    -- Couldn't call voicetrading/7168444444
Scheduling destruction of SIP dialog '45cf2dec6f8f41274c64a7f4241b070a@209.251.54.82' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/10500-054f80b0", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/10500-054f80b0", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/10500-054f80b0", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
    -- Executing [7168444444@from-internal:5] Macro("SIP/10500-054f80b0", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/10500-054f80b0", "all-circuits-busy-now|noanswer") in new stack
Audio is at 209.251.54.82 port 17858
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<--- Transmitting (NAT) to 76.180.27.4:17444 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.42.8:17444;branch=z9hG4bK-d87543-ab411f2bb66ccf71-1--d87543-;received=76.180.27.4;rport=17444
From: "Unavailable"<sip:10500@pbx.testcorp.com>;tag=a1376d4d
To: "7168444444"<sip:7168444444@pbx.testcorp.com>;tag=as3a94547d
Call-ID: ee068b42ba765105N2UxYWQwNzBiN2ZmZjYyMWQ5ZDY0YTcxY2VjODc2ODQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7168444444@209.251.54.82>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3860 3860 IN IP4 209.251.54.82
s=session
c=IN IP4 209.251.54.82
t=0 0
m=audio 17858 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv