SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.210:5060;branch=z9hG4bK57beccbb;rport=5060 From: "LBF" <sip:[email protected]>;tag=as6fee7365 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 103 INVITE Content-Length: 0 --- (7 headers 0 lines) --- SIP/2.0 404 Not Found From: "LBF" <sip:[email protected]>;tag=as6fee7365 To: <sip:[email protected]>;tag=SDup8s799-005ffc6a-0017-0486-0000-0000 Via: SIP/2.0/UDP 192.168.0.210:5060;branch=z9hG4bK57beccbb;rport=5060 Call-ID: [email protected] CSeq: 103 INVITE Reason: Q.850; cause=1 Content-Length: 0 --- (8 headers 0 lines) --- Transmitting (NAT) to 216.115.69.144:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 98.196.176.153:5060;branch=z9hG4bK57beccbb;rport Max-Forwards: 70 From: "LBF" <sip:[email protected]>;tag=as6fee7365 To: <sip:[email protected]>;tag=SDup8s799-005ffc6a-0017-0486-0000-0000 Contact: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 103 ACK User-Agent: FPBX-2.11.0(1.8.20.1) Content-Length: 0 --- Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/1003-00000020", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/1003-00000020", "0?continue,1:s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1003-00000020", "RC=1") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1003-00000020", "1,1") in new stack -- Goto (macro-dialout-trunk,1,1) -- Executing [1@macro-dialout-trunk:1] Goto("SIP/1003-00000020", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1003-00000020", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 1 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:2] Set("SIP/1003-00000020", "CALLERID(number)=1003") in new stack -- Executing [7137911414@from-internal:6] Macro("SIP/1003-00000020", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/1003-00000020", "") in new stack Audio is at 13216 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.18:49538;branch=z9hG4bKPjKZe8St6PbKBYBpger-pryrb3EFBPN0hA;received=192.168.0.18;rport=49538 From: <sip:[email protected]>;tag=rtvJApVE-lTfSPLhEsMImvWcB5VDZA9B To: <sip:[email protected]>;tag=as6ce27e3d Call-ID: DdX33HFp6Oq7BF34eMN68CdWOaC6TuYt CSeq: 4380 INVITE Server: FPBX-2.11.0(1.8.20.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp Require: timer Content-Length: 261 v=0 o=root 353950120 353950120 IN IP4 192.168.0.210 s=Asterisk PBX 1.8.20.1 c=IN IP4 192.168.0.210 t=0 0 m=audio 13216 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] GotoIf("SIP/1003-00000020", "0?emergency,1") in new stack -- Executing [s@macro-outisbusy:3] GotoIf("SIP/1003-00000020", "0?intracompany,1") in new stack -- Executing [s@macro-outisbusy:4] Playback("SIP/1003-00000020", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack -- <SIP/1003-00000020> Playing 'all-circuits-busy-now.ulaw' (language 'en') -- <SIP/1003-00000020> Playing 'pls-try-call-later.ulaw' (language 'en') Really destroying SIP dialog '[email protected]' Method: OPTIONS -- Executing [s@macro-outisbusy:5] Congestion("SIP/1003-00000020", "20") in new stack SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.0.18:49538;branch=z9hG4bKPjKZe8St6PbKBYBpger-pryrb3EFBPN0hA;received=192.168.0.18;rport=49538 From: <sip:[email protected]>;tag=rtvJApVE-lTfSPLhEsMImvWcB5VDZA9B To: <sip:[email protected]>;tag=as6ce27e3d Call-ID: DdX33HFp6Oq7BF34eMN68CdWOaC6TuYt CSeq: 4380 INVITE Server: FPBX-2.11.0(1.8.20.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 Content-Length: 0 [2013-11-11 16:01:54] WARNING[3061]: channel.c:4747 ast_prod: Prodding channel 'SIP/1003-00000020' failed [2013-11-11 16:01:54] WARNING[3061]: channel.c:4747 ast_prod: Prodding channel 'SIP/1003-00000020' failed == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/1003-00000020' in macro 'outisbusy' == Spawn extension (from-internal, 7137911414, 6) exited non-zero on 'SIP/1003-00000020' -- Executing [h@from-internal:1] Hangup("SIP/1003-00000020", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1003-00000020' ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.18:49538;rport;branch=z9hG4bKPjKZe8St6PbKBYBpger-pryrb3EFBPN0hA Max-Forwards: 70 From: <sip:[email protected]>;tag=rtvJApVE-lTfSPLhEsMImvWcB5VDZA9B To: <sip:[email protected]>;tag=as6ce27e3d Call-ID: DdX33HFp6Oq7BF34eMN68CdWOaC6TuYt CSeq: 4380 ACK Route: <sip:192.168.0.210:5060;transport=udp;lr> Content-Length: 0
You nned to setup your flowroute trunk properly. Currently it is not accepting your call to 7137911414, check your work.