asterisk1*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" ;tag=as2d4cacea To: Contact: Call-ID: 7ed8edd67dbe5259696bf2937ea6f230@voipstunt.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Shop" ;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" ;tag=as2d4cacea To: 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" ;tag=as2d4cacea To: Contact: Call-ID: 7ed8edd67dbe5259696bf2937ea6f230@voipstunt.com CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Shop" ;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>