-- Executing [s@macro-user-callerid:7] Set("SIP/2546-08eed708", "AMPUSERCID=2546") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/2546-08eed708", "CALLERID(all)="Foster Brian D" <2546>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/2546-08eed708", "REALCALLERIDNUM=2546") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/2546-08eed708", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/2546-08eed708", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/2546-08eed708", "Using CallerID "Foster Brian D" <2546>") in new stack
    -- Executing [17772255307@from-internal:2] Set("SIP/2546-08eed708", "_NODEST=") in new stack
    -- Executing [17772255307@from-internal:3] Macro("SIP/2546-08eed708", "record-enable|2546|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/2546-08eed708", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/2546-08eed708", "recordingcheck|20100817-063307|1282019587.526") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20100817-063307|1282019587.526: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/2546-08eed708", "") in new stack
    -- Executing [17772255307@from-internal:4] Macro("SIP/2546-08eed708", "dialout-trunk|2|17772255307||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/2546-08eed708", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2546-08eed708", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2546-08eed708", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/2546-08eed708", "DIAL_NUMBER=17772255307") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/2546-08eed708", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/2546-08eed708", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2546-08eed708", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2546-08eed708", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/2546-08eed708", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/2546-08eed708", "outbound-callerid|2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2546-08eed708", "0|SetCallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/2546-08eed708", "0|Set|REALCALLERIDNUM=2546") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/2546-08eed708", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/2546-08eed708", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/2546-08eed708", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/2546-08eed708", "TRUNKOUTCID=13174291069") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/2546-08eed708", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/2546-08eed708", "1|Set|CALLERID(all)=13174291069") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/2546-08eed708", "0|Set|CALLERID(all)=") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/2546-08eed708", "0|SetCallerPres|prohib_passed_screen") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/2546-08eed708", "1|AGI|fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

<--- SIP read from 204.11.192.39:5080 --->
SIP/2.0 200 Ok
v: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK545c35e1;rport=5060;received=82.59.12.239
f: <sip:17772180395@callcentric.com>;tag=as40090eed
t: <sip:17772180395@callcentric.com>
i: 03afa696652757ce3d78bfdd531007bb@127.0.0.1
CSeq: 115 REGISTER
m: <sip:17772180395@127.0.0.1>;expires=70
l: 0


<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '03afa696652757ce3d78bfdd531007bb@127.0.0.1' in 32000 ms (Method: REGISTER)
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/2546-08eed708", "OUTNUM=17772255307") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/2546-08eed708", "custom=SIP/callcentric") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2546-08eed708", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/2546-08eed708", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2546-08eed708", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2546-08eed708", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2546-08eed708", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/2546-08eed708", "SIP/callcentric/17772255307|300|") in new stack
Audio is at 127.0.0.1 port 17964
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 204.11.192.34:5080:
INVITE sip:17772255307@callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6c5acaa2;rport
From: "13174291069" <sip:17772180395@callcentric.com>;tag=as1084bdd5
To: <sip:17772255307@callcentric.com:5080>
Contact: <sip:17772180395@127.0.0.1>
Call-ID: 3d38d51c0d9f1a5f48a377d61a0c5759@callcentric.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 17 Aug 2010 04:33:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 2757 2757 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 17964 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called callcentric/17772255307
bdfoster*CLI>
<--- SIP read from 204.11.192.34:5080 --->
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6c5acaa2;rport=5060;received=82.59.12.239
f: "13174291069" <sip:17772180395@callcentric.com>;tag=as1084bdd5
t: <sip:17772255307@callcentric.com:5080>
i: 3d38d51c0d9f1a5f48a377d61a0c5759@callcentric.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="119a6fd7db507556775435a786559efa", opaque="", stale=TRUE, algorithm=MD5
l: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 204.11.192.34:5080:
ACK sip:17772255307@callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6c5acaa2;rport
From: "13174291069" <sip:17772180395@callcentric.com>;tag=as1084bdd5
To: <sip:17772255307@callcentric.com:5080>
Contact: <sip:17772180395@127.0.0.1>
Call-ID: 3d38d51c0d9f1a5f48a377d61a0c5759@callcentric.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/callcentric-08a970b8 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/2546-08eed708", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/2546-08eed708", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/2546-08eed708", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing [17772255307@from-internal:5] Macro("SIP/2546-08eed708", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/2546-08eed708", "all-circuits-busy-now|noanswer") in new stack
Audio is at 10.0.1.5 port 17268
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 10.0.1.4:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.4:5060;branch=z9hG4bK-11f69fe0;received=10.0.1.4
From: "Brian Foster" <sip:2546@10.0.1.5>;tag=11079bf0b413dc0co0
To: <sip:17772255307@10.0.1.5>;tag=as44e3477f
Call-ID: cb88b4c0-e531197c@10.0.1.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17772255307@10.0.1.5>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 2757 2757 IN IP4 10.0.1.5
s=session
c=IN IP4 10.0.1.5
t=0 0
m=audio 17268 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

<------------>
    -- <SIP/2546-08eed708> Playing 'all-circuits-busy-now' (language 'en')
Really destroying SIP dialog '3d38d51c0d9f1a5f48a377d61a0c5759@callcentric.com' Method: INVITE