rendered paste bodyasterisk1*CLI> sip debug peer voipstunt
SIP Debugging Enabled for IP: 194.120.0.202:5060
-- Executing Macro("SIP/503-09d9b150", "dialout-trunk|3|6026923282||") in new stack
-- Executing Set("SIP/503-09d9b150", "DIAL_TRUNK=3") in new stack
-- Executing Set("SIP/503-09d9b150", "DIAL_NUMBER=6026923282") in new stack
-- Executing Set("SIP/503-09d9b150", "ROUTE_PASSWD=") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing Set("SIP/503-09d9b150", "GROUP()=OUT_3") in new stack
-- Executing Macro("SIP/503-09d9b150", "user-callerid") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "0?report") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "0?start") in new stack
-- Executing Set("SIP/503-09d9b150", "REALCALLERIDNUM=503") in new stack
-- Executing NoOp("SIP/503-09d9b150", "REALCALLERIDNUM is 503") in new stack
-- Executing Set("SIP/503-09d9b150", "AMPUSER=503") in new stack
-- Executing Set("SIP/503-09d9b150", "AMPUSERCIDNAME=Shop") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "0?report") in new stack
-- Executing Set("SIP/503-09d9b150", "CALLERID(all)=Shop <503>") in new stack
-- Executing NoOp("SIP/503-09d9b150", "Using CallerID "Shop" <503>") in new stack
-- Executing Macro("SIP/503-09d9b150", "record-enable|503|OUT") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/503-09d9b150", "recordingcheck|20061028-143737|1162071457.37") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20061028-143737|1162071457.37: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/503-09d9b150", "No recording needed") in new stack
-- Executing Macro("SIP/503-09d9b150", "outbound-callerid|3") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/503-09d9b150", "REALCALLERIDNUM is 503") in new stack
-- Executing Set("SIP/503-09d9b150", "USEROUTCID=") in new stack
-- Executing Set("SIP/503-09d9b150", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/503-09d9b150", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing GotoIf("SIP/503-09d9b150", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,13)
-- Executing GotoIf("SIP/503-09d9b150", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,17)
-- Executing NoOp("SIP/503-09d9b150", "CallerID set to "Shop" <503>") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,12)
-- Executing AGI("SIP/503-09d9b150", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Added prefix. New number: 0116026923282
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/503-09d9b150", "OUTNUM=0116026923282") in new stack
-- Executing Set("SIP/503-09d9b150", "custom=SIP/Voipstunt") in new stack
-- Executing GotoIf("SIP/503-09d9b150", "0?customtrunk") in new stack
-- Executing Dial("SIP/503-09d9b150", "SIP/Voipstunt/0116026923282|120|r") in new stack
We're at 192.168.0.25 port 14904
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 14 lines
Reliably Transmitting (NAT) to 194.120.0.202:5060:
INVITE sip:0116026923282@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK4866eb76;rport
From: "Shop" <sip:n7okn@voipstunt.com>;tag=as2d4cacea
To: <sip:0116026923282@sip.voipstunt.com>
Contact: <sip:n7okn@192.168.0.25>
Call-ID: 7ed8edd67dbe5259696bf2937ea6f230@voipstunt.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Shop" <sip:503@voipstunt.com>;privacy=off;screen=no
Date: Sat, 28 Oct 2006 21:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 308
v=0
o=root 8901 8901 IN IP4 192.168.0.25
s=session
c=IN IP4 192.168.0.25
t=0 0
m=audio 14904 RTP/AVP 0 18 3 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called Voipstunt/0116026923282
asterisk1*CLI>
<-- SIP read from 194.120.0.202:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK4866eb76;rport
From: "Shop" <sip:n7okn@voipstunt.com>;tag=as2d4cacea
To: <sip:0116026923282@sip.voipstunt.com>
Contact: sip:0116026923282@194.120.0.202:5060
Call-ID: 7ed8edd67dbe5259696bf2937ea6f230@voipstunt.com
CSeq: 102 INVITE
Server: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (NAT) to 194.120.0.202:5060:
ACK sip:0116026923282@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK4866eb76;rport
From: "Shop" <sip:n7okn@voipstunt.com>;tag=as2d4cacea
To: <sip:0116026923282@sip.voipstunt.com>
Contact: <sip:n7okn@192.168.0.25>
Call-ID: 7ed8edd67dbe5259696bf2937ea6f230@voipstunt.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Shop" <sip:503@voipstunt.com>;privacy=off;screen=no
Content-Length: 0
---
-- SIP/Voipstunt-09da0690 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/503-09d9b150", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/503-09d9b150", "Dial failed due to CONGESTION") in new stack
-- Executing Macro("SIP/503-09d9b150", "outisbusy|") in new stack
-- Executing Playback("SIP/503-09d9b150", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
Destroying call '7ed8edd67dbe5259696bf2937ea6f230@voipstunt.com'
-- Executing Playback("SIP/503-09d9b150", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/503-09d9b150", "hangupcall") in new stack
-- Executing ResetCDR("SIP/503-09d9b150", "w") in new stack
-- Executing NoCDR("SIP/503-09d9b150", "") in new stack
-- Executing Wait("SIP/503-09d9b150", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/503-09d9b150' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/503-09d9b150' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/503-09d9b150'
Destroying call '0c6bce58557fda71095fa8f7605fe166@127.0.0.1'
asterisk1*CLI>