asterisk1*CLI> sip debug SIP Debugging enabled asterisk1*CLI> <-- SIP read from 192.168.2.167:5060: INVITE sip:713092872137@192.168.2.6;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK4ed386b354825728 From: "Adam Shake" ;tag=519BB939-561A42AC To: CSeq: 1 INVITE Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1170843165 1170843165 IN IP4 192.168.2.167 s=Polycom IP Phone c=IN IP4 192.168.2.167 t=0 0 a=sendrecv m=audio 2240 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Sending to 192.168.2.167 : 5060 (non-NAT) Reliably Transmitting (NAT) to 192.168.2.167:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK4ed386b354825728;received=192.168.2.167 From: "Adam Shake" ;tag=519BB939-561A42AC To: ;tag=as77dba809 Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44e0293b" Content-Length: 0 --- Scheduling destruction of call 'aebd6655-3e46ebd7-f1173f7a@192.168.2.167' in 15000 ms Found user '3101' asterisk1*CLI> <-- SIP read from 192.168.2.167:5060: ACK sip:713092872137@192.168.2.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK4ed386b354825728 From: "Adam Shake" ;tag=519BB939-561A42AC To: ;tag=as77dba809 CSeq: 1 ACK Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- asterisk1*CLI> <-- SIP read from 192.168.2.167:5060: INVITE sip:713092872137@192.168.2.6;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B From: "Adam Shake" ;tag=519BB939-561A42AC To: CSeq: 2 INVITE Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="3101", realm="asterisk", nonce="44e0293b", uri="sip:713092872137@192.168.2.6;user=phone", response="8a565ee944d6e131141968283cb9415c", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1170843165 1170843165 IN IP4 192.168.2.167 s=Polycom IP Phone c=IN IP4 192.168.2.167 t=0 0 a=sendrecv m=audio 2240 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 =rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Sending to 192.168.2.167 : 5060 (NAT) Found user '3101' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.167:2240 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 713092872137 in from-internal (domain 192.168.2.6;user=phone) list_route: hop: Transmitting (NAT) to 192.168.2.167:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167 From: "Adam Shake" ;tag=519BB939-561A42AC To: Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/3101-09c10640", "dialout-trunk|1|13092872137||") in new stack -- Executing Set("SIP/3101-09c10640", "DIAL_TRUNK=1") in new stack -- Executing Set("SIP/3101-09c10640", "_NODEST=") in new stack -- Executing Set("SIP/3101-09c10640", "DIAL_NUMBER=13092872137") in new stack -- Executing Set("SIP/3101-09c10640", "ROUTE_PASSWD=") in new stack -- Executing Set("SIP/3101-09c10640", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/3101-09c10640", "GROUP()=OUT_1") in new stack -- Executing Macro("SIP/3101-09c10640", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/3101-09c10640", "user-callerid: device 3101") in new stack -- Executing GotoIf("SIP/3101-09c10640", "0?report") in new stack -- Executing GotoIf("SIP/3101-09c10640", "0?start") in new stack -- Executing Set("SIP/3101-09c10640", "REALCALLERIDNUM=3101") in new stack -- Executing NoOp("SIP/3101-09c10640", "REALCALLERIDNUM is 3101") in new stack -- Executing Set("SIP/3101-09c10640", "AMPUSER=3101") in new stack -- Executing Set("SIP/3101-09c10640", "AMPUSERCIDNAME=Adam Shake") in new stack -- Executing GotoIf("SIP/3101-09c10640", "0?report") in new stack -- Executing Set("SIP/3101-09c10640", "CALLERID(all)=Adam Shake <3101>") in new stack -- Executing Set("SIP/3101-09c10640", "REALCALLERIDNUM=3101") in new stack -- Executing NoOp("SIP/3101-09c10640", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/3101-09c10640", "Using CallerID "Adam Shake" <3101>") in new stack -- Executing Macro("SIP/3101-09c10640", "record-enable|3101|OUT") in new stack -- Executing GotoIf("SIP/3101-09c10640", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/3101-09c10640", "recordingcheck|20070207-214756|1170906476.2") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070207-214756|1170906476.2: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/3101-09c10640", "No recording needed") in new stack -- Executing GotoIf("SIP/3101-09c10640", "0?skipoutcid") in new stack -- Executing Set("SIP/3101-09c10640", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing Macro("SIP/3101-09c10640", "outbound-callerid|1") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/3101-09c10640", "REALCALLERIDNUM is 3101") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/3101-09c10640", "USEROUTCID=") in new stack -- Executing Set("SIP/3101-09c10640", "EMERGENCYCID=") in new stack -- Executing Set("SIP/3101-09c10640", "TRUNKOUTCID=") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/3101-09c10640", "1?usercid") in new stack -- Goto (macro-outbound-callerid,s,18) -- Executing GotoIf("SIP/3101-09c10640", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/3101-09c10640", "CallerID set to "Adam Shake" <3101>") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,16) -- Executing DeadAGI("SIP/3101-09c10640", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/3101-09c10640", "OUTNUM=13092872137") in new stack -- Executing Set("SIP/3101-09c10640", "custom=ZAP/g0") in new stack -- Executing GotoIf("SIP/3101-09c10640", "0?customtrunk") in new stack -- Executing Dial("SIP/3101-09c10640", "ZAP/g0/13092872137|300|") in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/3101-09c10640", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/3101-09c10640", "Dial failed due to CONGESTION - failing through to other trunks") in new stack -- Executing Macro("SIP/3101-09c10640", "outisbusy|") in new stack -- Executing Playback("SIP/3101-09c10640", "all-circuits-busy-now|noanswer") in new stack We're at 192.168.2.6 port 16906 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 192.168.2.167:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167 From: "Adam Shake" ;tag=519BB939-561A42AC To: ;tag=as19fb306a Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 2536 2536 IN IP4 192.168.2.6 s=session c=IN IP4 192.168.2.6 t=0 0 m=audio 16906 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/3101-09c10640", "pls-try-call-later|noanswer") in new stack -- Playing 'pls-try-call-later' (language 'en') -- Executing Macro("SIP/3101-09c10640", "hangupcall") in new stack -- Executing ResetCDR("SIP/3101-09c10640", "w") in new stack -- Executing NoCDR("SIP/3101-09c10640", "") in new stack -- Executing GotoIf("SIP/3101-09c10640", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("SIP/3101-09c10640", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("SIP/3101-09c10640", "5") in new stack asterisk1*CLI> <-- SIP read from 192.168.2.167:5060: CANCEL sip:713092872137@192.168.2.6;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B From: "Adam Shake" ;tag=519BB939-561A42AC To: CSeq: 2 CANCEL Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Proxy-Authorization: Digest username="3101", realm="asterisk", nonce="44e0293b", uri="sip:713092872137@192.168.2.6;user=phone", response="8a565ee944d6e131141968283cb9415c", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- Sending to 192.168.2.167 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.2.167:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167 From: "Adam Shake" ;tag=519BB939-561A42AC To: ;tag=as19fb306a Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 X-Asterisk-HangupCause: Circuit/channel congestion --- Transmitting (NAT) to 192.168.2.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167 From: "Adam Shake" ;tag=519BB939-561A42AC To: ;tag=as19fb306a Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Circuit/channel congestion --- == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3101-09c10640' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3101-09c10640' in macro 'outisbusy' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3101-09c10640' asterisk1*CLI> <-- SIP read from 192.168.2.167:5060: ACK sip:713092872137@192.168.2.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B From: "Adam Shake" ;tag=519BB939-561A42AC To: ;tag=as19fb306a CSeq: 2 ACK Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- Destroying call 'aebd6655-3e46ebd7-f1173f7a@192.168.2.167' 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.2.167:5060: OPTIONS sip:3101@192.168.2.167 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.6:5060;branch=z9hG4bK40aa3608;rport From: "Unknown" ;tag=as7e8cbfdc To: Contact: Call-ID: 07a98df94048464426ecac8770e27f2a@192.168.2.6 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 03:48:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.2.167:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.6:5060;branch=z9hG4bK40aa3608;rport From: "Unknown" ;tag=as7e8cbfdc To: ;tag=57441B67-4D7B718A CSeq: 102 OPTIONS Call-ID: 07a98df94048464426ecac8770e27f2a@192.168.2.6 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 --- (10 headers 0 lines) --- Destroying call '07a98df94048464426ecac8770e27f2a@192.168.2.6' asterisk1*CLI>