[root@localhost ~]# asterisk -r Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.0.9 currently running on localhost (pid = 23639) Verbosity is at least 10 localhost*CLI> sip set debug on SIP Debugging enabled localhost*CLI> <--- SIP read from UDP://10.20.20.62:5060 ---> INVITE sip:7000@10.20.20.25:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.20.20.62;branch=z9hG4bKfd60efc75894B0DC From: "Greg Nutt" ;tag=7B2AD619-818A3948 To: CSeq: 1 INVITE Call-ID: efc8325d-32a442a3-3821af9a@10.20.20.62 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.2.0392 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 268 v=0 o=- 1242330512 1242330512 IN IP4 10.20.20.62 s=Polycom IP Phone c=IN IP4 10.20.20.62 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 12 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Sending to 10.20.20.62 : 5060 (NAT) Using INVITE request as basis request - efc8325d-32a442a3-3821af9a@10.20.20.62 Found user '5221' for '5221' 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 10.20.20.62:2222 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.20.20.62:2222 Looking for 7000 in DLPN_DialPlan1 (domain 10.20.20.25) list_route: hop: <--- Transmitting (NAT) to 10.20.20.62:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.20.20.62;branch=z9hG4bKfd60efc75894B0DC;received=10.20.20.62 From: "Greg Nutt" ;tag=7B2AD619-818A3948 To: Call-ID: efc8325d-32a442a3-3821af9a@10.20.20.62 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [7000@DLPN_DialPlan1:1] Macro("SIP/5221-090d3d00", "trunkdial-failover-0.3,SIP/NUTT_LOCAL/7000,,NUTT_LOCAL,") in new stack -- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/5221-090d3d00", "0?1-fmsetcid,1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/5221-090d3d00", "0?1-setgbobname,1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/5221-090d3d00", "CALLERID(num)=") in new stack -- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf("SIP/5221-090d3d00", "0?1-dial,1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:5] Set("SIP/5221-090d3d00", "CALLERID(all)=") in new stack -- Executing [s@macro-trunkdial-failover-0.3:6] Goto("SIP/5221-090d3d00", "1-dial,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/5221-090d3d00", "SIP/NUTT_LOCAL/7000") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Audio is at 142.46.193.202 port 16240 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 206.248.146.138:5060: INVITE sip:7000@206.248.146.138 SIP/2.0 Via: SIP/2.0/UDP 142.46.193.202:5060;branch=z9hG4bK75e12728 Max-Forwards: 70 From: "asterisk" ;tag=as2100555a To: Contact: Call-ID: 65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 15 May 2009 08:09:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 345 v=0 o=root 1040231192 1040231192 IN IP4 142.46.193.202 s=Asterisk PBX 1.6.0.9 c=IN IP4 142.46.193.202 t=0 0 m=audio 16240 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called NUTT_LOCAL/7000 <--- SIP read from UDP://127.0.0.1:5060 ---> INVITE sip:7000@206.248.146.138 SIP/2.0 Via: SIP/2.0/UDP 142.46.193.202:5060;branch=z9hG4bK75e12728 Max-Forwards: 70 From: "asterisk" ;tag=as2100555a To: Contact: Call-ID: 65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 15 May 2009 08:09:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 345 v=0 o=root 1040231192 1040231192 IN IP4 142.46.193.202 s=Asterisk PBX 1.6.0.9 c=IN IP4 142.46.193.202 t=0 0 m=audio 16240 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 15 lines) --- <--- Transmitting (NAT) to 0.0.0.0:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 142.46.193.202:5060;branch=z9hG4bK75e12728;received=127.0.0.1 From: "asterisk" ;tag=as2100555a To: ;tag=as2100555a Call-ID: 65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 <------------> Scheduling destruction of SIP dialog '65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202' in 32000 ms (Method: INVITE) <--- SIP read from UDP://127.0.0.1:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 142.46.193.202:5060;branch=z9hG4bK75e12728;received=127.0.0.1 From: "asterisk" ;tag=as2100555a To: ;tag=as2100555a Call-ID: 65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 <-------------> --- (12 headers 0 lines) --- -- Got SIP response 482 "Loop Detected" back from 0.0.0.0 Transmitting (NAT) to 127.0.0.1:5060: ACK sip:7000@206.248.146.138 SIP/2.0 Via: SIP/2.0/UDP 142.46.193.202:5060;branch=z9hG4bK75e12728 Max-Forwards: 70 From: "asterisk" ;tag=as2100555a To: ;tag=as2100555a Contact: Call-ID: 65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- <--- SIP read from UDP://127.0.0.1:5060 ---> ACK sip:7000@206.248.146.138 SIP/2.0 Via: SIP/2.0/UDP 142.46.193.202:5060;branch=z9hG4bK75e12728 Max-Forwards: 70 From: "asterisk" ;tag=as2100555a To: ;tag=as2100555a Contact: Call-ID: 65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Now forwarding SIP/5221-090d3d00 to 'Local/7000@DID_NUTT_LOCAL' (thanks to SIP/NUTT_LOCAL-091034d0) [May 15 04:09:44] NOTICE[30637]: chan_local.c:654 local_alloc: No such extension/context 7000@DID_NUTT_LOCAL creating local channel [May 15 04:09:44] NOTICE[30637]: app_dial.c:513 do_forward: Unable to create local channel for call forward to 'Local/7000@DID_NUTT_LOCAL' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/5221-090d3d00", "0 > 0 ?1-CHANUNAVAIL,1:1-out,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-out,1) -- Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup("SIP/5221-090d3d00", "") in new stack == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/5221-090d3d00' in macro 'trunkdial-failover-0.3' == Spawn extension (DLPN_DialPlan1, 7000, 1) exited non-zero on 'SIP/5221-090d3d00' Scheduling destruction of SIP dialog 'efc8325d-32a442a3-3821af9a@10.20.20.62' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 10.20.20.62:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.20.20.62;branch=z9hG4bKfd60efc75894B0DC;received=10.20.20.62 From: "Greg Nutt" ;tag=7B2AD619-818A3948 To: ;tag=as51c7378c Call-ID: efc8325d-32a442a3-3821af9a@10.20.20.62 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> localhost*CLI> <--- SIP read from UDP://10.20.20.62:5060 ---> ACK sip:7000@10.20.20.25:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.20.62;branch=z9hG4bKfd60efc75894B0DC From: "Greg Nutt" ;tag=7B2AD619-818A3948 To: ;tag=as51c7378c CSeq: 1 ACK Call-ID: efc8325d-32a442a3-3821af9a@10.20.20.62 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.2.0392 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '65b268180ec7f49d716e8fe15d41a4f4@142.46.193.202' Method: ACK localhost*CLI> sip set debug off SIP Debugging Disabled localhost*CLI>