rendered paste body<--- Reliably Transmitting (NAT) to 41.221.5.18:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK70615d99;received=41.221.5.18;rport=5060
From: "+27105905701" <sip:+27105905701@41.221.5.18>;tag=as4fae6db5
To: <sip:0105905701@41.86.105.230>;tag=as0cf44ea8
Call-ID: 12ebf25d31f010ba740e8ac51fc95762@41.221.5.18
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.4.43)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
[Jan 10 22:23:09] NOTICE[28920]: chan_sip.c:15765 handle_request_invite: Call from 'andrewcolin' to extension '0105905701' rejected because extension not found.
Scheduling destruction of SIP dialog '12ebf25d31f010ba740e8ac51fc95762@41.221.5.18' in 6400 ms (Method: INVITE)
<--- SIP read from 41.221.5.18:5060 --->
ACK sip:0105905701@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK70615d99;rport
From: "+27105905701" <sip:+27105905701@41.221.5.18>;tag=as4fae6db5
To: <sip:0105905701@41.86.105.230>;tag=as0cf44ea8
Contact: <sip:+27105905701@41.221.5.18>
Call-ID: 12ebf25d31f010ba740e8ac51fc95762@41.221.5.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '12ebf25d31f010ba740e8ac51fc95762@41.221.5.18' Method: ACK
<--- SIP read from 41.221.5.18:5060 --->
INVITE sip:0105905701@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK1a421887;rport
From: "+27105905701" <sip:+27105905701@41.221.5.18>;tag=as63c91f77
To: <sip:0105905701@41.86.105.230>
Contact: <sip:+27105905701@41.221.5.18>
Call-ID: 1dd1d5f10c3078943f40882021551430@41.221.5.18
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jan 2012 20:23:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 3160 3160 IN IP4 41.221.5.18
s=session
c=IN IP4 41.221.5.18
t=0 0
m=audio 13686 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 41.221.5.18 : 5060 (NAT)
Using INVITE request as basis request - 1dd1d5f10c3078943f40882021551430@41.221.5.18
Found peer 'andrewcolin'
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 41.221.5.18:13686
Looking for 0105905701 in external (domain 41.86.105.230)
<--- Reliably Transmitting (NAT) to 41.221.5.18:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK1a421887;received=41.221.5.18;rport=5060
From: "+27105905701" <sip:+27105905701@41.221.5.18>;tag=as63c91f77
To: <sip:0105905701@41.86.105.230>;tag=as7194a511
Call-ID: 1dd1d5f10c3078943f40882021551430@41.221.5.18
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.4.43)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
[Jan 10 22:23:09] NOTICE[28920]: chan_sip.c:15765 handle_request_invite: Call from 'andrewcolin' to extension '0105905701' rejected because extension not found.
Scheduling destruction of SIP dialog '1dd1d5f10c3078943f40882021551430@41.221.5.18' in 6400 ms (Method: INVITE)
<--- SIP read from 41.221.5.18:5060 --->
ACK sip:0105905701@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.221.5.18:5060;branch=z9hG4bK1a421887;rport
From: "+27105905701" <sip:+27105905701@41.221.5.18>;tag=as63c91f77
To: <sip:0105905701@41.86.105.230>;tag=as7194a511
Contact: <sip:+27105905701@41.221.5.18>
Call-ID: 1dd1d5f10c3078943f40882021551430@41.221.5.18
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1dd1d5f10c3078943f40882021551430@41.221.5.18' Method: ACK
-- IAX2/bitco-2950 is circuit-busy
-- Hungup 'IAX2/bitco-2950'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:21] NoOp("SIP/900-0000005d", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
-- Executing [s@macro-dialout-trunk:22] Goto("SIP/900-0000005d", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/900-0000005d", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/900-0000005d", "34|1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/900-0000005d", "continue|1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/900-0000005d", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/900-0000005d", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/900-0000005d", "CALLERID(number)=900") in new stack
-- Executing [0105905701@from-internal:6] Macro("SIP/900-0000005d", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/900-0000005d", "") in new stack
Audio is at 41.86.105.230 port 10450
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 41.135.75.17:5061 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 41.135.75.17:5061;branch=z9hG4bK-7cef9f0e;received=41.135.75.17
From: 900 <sip:900@41.86.105.230>;tag=e3cbbe759271ff7ao1
To: <sip:0105905701@41.86.105.230>;tag=as0668c01d
Call-ID: fdfd4a51-a8ec854e@192.168.1.2
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.4.43)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:0105905701@41.86.105.230>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 28890 28890 IN IP4 41.86.105.230
s=session
c=IN IP4 41.86.105.230
t=0 0
m=audio 10450 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Executing [s@macro-outisbusy:2] Playback("SIP/900-0000005d", "all-circuits-busy-now|noanswer") in new stack
[Jan 10 22:23:09] WARNING[4066]: file.c:665 ast_openstream_full: File all-circuits-busy-now does not exist in any format
[Jan 10 22:23:09] WARNING[4066]: file.c:995 ast_streamfile: Unable to open all-circuits-busy-now (format 0x100 (g729)): No such file or directory
[Jan 10 22:23:09] WARNING[4066]: app_playback.c:440 playback_exec: ast_streamfile failed on SIP/900-0000005d for all-circuits-busy-now|noanswer
-- Executing [s@macro-outisbusy:3] Playback("SIP/900-0000005d", "pls-try-call-later|noanswer") in new stack
[Jan 10 22:23:09] WARNING[4066]: file.c:665 ast_openstream_full: File pls-try-call-later does not exist in any format
[Jan 10 22:23:09] WARNING[4066]: file.c:995 ast_streamfile: Unable to open pls-try-call-later (format 0x100 (g729)): No such file or directory
[Jan 10 22:23:09] WARNING[4066]: app_playback.c:440 playback_exec: ast_streamfile failed on SIP/900-0000005d for pls-try-call-later|noanswer
-- Executing [s@macro-outisbusy:4] Macro("SIP/900-0000005d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/900-0000005d", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/900-0000005d", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,5)
-- Executing [s@macro-hangupcall:5] GotoIf("SIP/900-0000005d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] Hangup("SIP/900-0000005d", "") in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/900-0000005d' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/900-0000005d' in macro 'outisbusy'
== Spawn extension (from-internal, 0105905701, 6) exited non-zero on 'SIP/900-0000005d'
-- Executing [h@from-internal:1] Hangup("SIP/900-0000005d", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/900-0000005d'
Scheduling destruction of SIP dialog 'fdfd4a51-a8ec854e@192.168.1.2' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 41.135.75.17:5061 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 41.135.75.17:5061;branch=z9hG4bK-7cef9f0e;received=41.135.75.17
From: 900 <sip:900@41.86.105.230>;tag=e3cbbe759271ff7ao1
To: <sip:0105905701@41.86.105.230>;tag=as0668c01d
Call-ID: fdfd4a51-a8ec854e@192.168.1.2
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.4.43)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
== End MixMonitor Recording SIP/900-0000005d
<--- SIP read from 41.135.75.17:5061 --->
ACK sip:0105905701@41.86.105.230 SIP/2.0
Via: SIP/2.0/UDP 41.135.75.17:5061;branch=z9hG4bK-7cef9f0e
From: 900 <sip:900@41.86.105.230>;tag=e3cbbe759271ff7ao1
To: <sip:0105905701@41.86.105.230>;tag=as0668c01d
Call-ID: fdfd4a51-a8ec854e@192.168.1.2
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="900",realm="asterisk",nonce="3c4524ae",uri="sip:0105905701@41.86.105.230",algorithm=MD5,response="0506cbecaeb154cb8c91efb0bb56c6ec"
Contact: 900 <sip:900@41.135.75.17:5061>
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
tornado*CLI>