a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #6 (no NAT) to 216.58.0.51:5060: REGISTER sip:216.58.0.51 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK03b075e1;rport Max-Forwards: 70 From: ;tag=as01b3a028 To: Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1 CSeq: 165 REGISTER User-Agent: FPBX-2.8.1(1.6.2.15) Expires: 120 Contact: Content-Length: 0 --- REGISTER 10 headers, 0 lines Reliably Transmitting (no NAT) to 216.58.0.51:5060: REGISTER sip:216.58.0.51 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6b864217;rport Max-Forwards: 70 From: ;tag=as77a328cb To: Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1 CSeq: 166 REGISTER User-Agent: FPBX-2.8.1(1.6.2.15) Expires: 120 Contact: Content-Length: 0 --- Really destroying SIP dialog '7f310bc557a68d50120b4df406a1cb25@127.0.0.1' Method: REGISTER Retransmitting #1 (no NAT) to 216.58.0.51:5060: REGISTER sip:216.58.0.51 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6b864217;rport Max-Forwards: 70 From: ;tag=as77a328cb To: Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1 CSeq: 166 REGISTER User-Agent: FPBX-2.8.1(1.6.2.15) Expires: 120 Contact: Content-Length: 0 --- Retransmitting #2 (no NAT) to 216.58.0.51:5060: REGISTER sip:216.58.0.51 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6b864217;rport Max-Forwards: 70 From: ;tag=as77a328cb To: Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1 CSeq: 166 REGISTER User-Agent: FPBX-2.8.1(1.6.2.15) Expires: 120 Contact: Content-Length: 0 --- == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/203-0000067e", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack -- Executing [s@macro-dialout-trunk:21] Goto("SIP/203-0000067e", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/203-0000067e", "RC=111") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/203-0000067e", "111,1") in new stack -- Goto (macro-dialout-trunk,111,1) -- Executing [111@macro-dialout-trunk:1] Goto("SIP/203-0000067e", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/203-0000067e", "1?noreport") in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/203-0000067e", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 111 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:4] Set("SIP/203-0000067e", "CALLERID(number)=203") in new stack -- Executing [4168853558@from-internal:7] Macro("SIP/203-0000067e", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/203-0000067e", "") in new stack Audio is at 10.10.10.7 port 16182 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 10.10.10.12:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-8e22ef22;received=10.10.10.12 From: "Itamar Desk" ;tag=fbf6965bb4d0b75eo0 To: "14168853558" ;tag=as09bfab37 Call-ID: 3d947bdf-fb3e6faa@10.10.10.12 CSeq: 102 INVITE Server: FPBX-2.8.1(1.6.2.15) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 1722232481 1722232481 IN IP4 10.10.10.7 s=Asterisk PBX 1.6.2.15 c=IN IP4 10.10.10.7 t=0 0 m=audio 16182 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=ptime:20 a=sendrecv <------------> -- Executing [s@macro-outisbusy:2] Playback("SIP/203-0000067e", "all-circuits-busy-now,noanswer") in new stack -- Playing 'all-circuits-busy-now.gsm' (language 'en') Really destroying SIP dialog '2439b7946a79860616f8f33068549a03@cia.com' Method: INVITE -- Executing [s@macro-outisbusy:3] Playback("SIP/203-0000067e", "pls-try-call-later,noanswer") in new stack -- Playing 'pls-try-call-later.gsm' (language 'en') Retransmitting #3 (no NAT) to 216.58.0.51:5060: REGISTER sip:216.58.0.51 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6b864217;rport Max-Forwards: 70 From: ;tag=as77a328cb To: Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1 CSeq: 166 REGISTER User-Agent: FPBX-2.8.1(1.6.2.15) Expires: 120 Contact: Content-Length: 0 --- -- Executing [s@macro-outisbusy:4] Macro("SIP/203-0000067e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/203-0000067e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/203-0000067e", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/203-0000067e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/203-0000067e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/203-0000067e' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/203-0000067e' in macro 'outisbusy' == Spawn extension (from-internal, 4168853558, 7) exited non-zero on 'SIP/203-0000067e' -- Executing [h@from-internal:1] Macro("SIP/203-0000067e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/203-0000067e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/203-0000067e", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/203-0000067e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/203-0000067e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/203-0000067e' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/203-0000067e' Scheduling destruction of SIP dialog '3d947bdf-fb3e6faa@10.10.10.12' in 6400 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 10.10.10.12:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-8e22ef22;received=10.10.10.12 From: "Itamar Desk" ;tag=fbf6965bb4d0b75eo0 To: "14168853558" ;tag=as09bfab37 Call-ID: 3d947bdf-fb3e6faa@10.10.10.12 CSeq: 102 INVITE Server: FPBX-2.8.1(1.6.2.15) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:10.10.10.12:5060 ---> ACK sip:4168853558@10.10.10.7 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-8e22ef22 From: "Itamar Desk" ;tag=fbf6965bb4d0b75eo0 To: "14168853558" ;tag=as09bfab37 Call-ID: 3d947bdf-fb3e6faa@10.10.10.12 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="203",realm="asterisk",nonce="227831b1",uri="sip:4168853558@10.10.10.7",algorithm=MD5,response="54fbcbcafb92ae528cdc62918523121a" Contact: "Itamar Desk" User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI>