Script started on mer 08 ott 2008 08:49:48 CEST aviano:/# asterisk -r Asterisk 1.4.22, 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. ========================================================================= [Oct 8 08:49:54] == Parsing '/etc/asterisk/extconfig.conf': [Oct 8 08:49:54] Found [Oct 8 08:49:54] Connected to Asterisk 1.4.22 currently running on aviano (pid = 3444) aviano*CLI> Verbosity is at least 10 [Kaviano*CLI> sip debug aviano*CLI> SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. [Kaviano*CLI> sip set debug on aviano*CLI> Usage: sip set debug Enables dumping of SIP packets for debugging purposes sip set debug ip Enables dumping of SIP packets to and from host. sip set debug peer Enables dumping of SIP packets to and from host. Require peer to be registered. [Kaviano*CLI> sip set debug on [2008-10-08 08:50:14] NOTICE[3456]: chan_sip.c:7520 sip_reregister: -- Re-registration for 8888@trieste.tnet.it [Kaviano*CLI> sip set debug on [2008-10-08 08:50:14] REGISTER 13 headers, 0 lines [Kaviano*CLI> sip set debug on [2008-10-08 08:50:14] Reliably Transmitting (no NAT) to 172.25.18.68:5060: REGISTER sip:trieste.tnet.it SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK24eeed69;rport From: ;tag=as43100ad8 To: Call-ID: 66cee1f06e04e4b71735106b34dca9f0@172.25.18.66 CSeq: 110 REGISTER User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Authorization: Digest username="8888", realm="trieste.tnet.it", algorithm=MD5, uri="sip:trieste.tnet.it", nonce="48ec5817000001812b62da9100c5abd04d0a0d6860e6c98e", response="847e8c95f799e2013f8a3b75f585a43a" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Kaviano*CLI> sip set debug on [2008-10-08 08:50:14] <--- SIP read from 172.25.18.68:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK24eeed69;rport=5060 From: ;tag=as43100ad8 To: ;tag=dec0d061b99cee799a1c831932c39493.19c3 Call-ID: 66cee1f06e04e4b71735106b34dca9f0@172.25.18.66 CSeq: 110 REGISTER WWW-Authenticate: Digest realm="trieste.tnet.it", nonce="48ec595200000183cf66e4f6d7aff5c0658795f73cba18e9", stale=true Server: Kamailio (1.4.1-notls (i386/linux)) Content-Length: 0 <-------------> [2008-10-08 08:50:14] --- (9 headers 0 lines) --- [2008-10-08 08:50:14] Responding to challenge, registration to domain/host name trieste.tnet.it [2008-10-08 08:50:14] REGISTER 13 headers, 0 lines [2008-10-08 08:50:14] Reliably Transmitting (no NAT) to 172.25.18.68:5060: REGISTER sip:trieste.tnet.it SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK61245e90;rport From: ;tag=as0d4b070e To: Call-ID: 66cee1f06e04e4b71735106b34dca9f0@172.25.18.66 CSeq: 111 REGISTER User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Authorization: Digest username="8888", realm="trieste.tnet.it", algorithm=MD5, uri="sip:trieste.tnet.it", nonce="48ec595200000183cf66e4f6d7aff5c0658795f73cba18e9", response="be2d2e3b1dad6d1e43d128a75b8e6ae9" Expires: 120 Contact: Event: registration Content-Length: 0 --- [Kaviano*CLI> sip set debug on [2008-10-08 08:50:14] <--- SIP read from 172.25.18.68:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK61245e90;rport=5060 From: ;tag=as0d4b070e To: ;tag=dec0d061b99cee799a1c831932c39493.9b96 Call-ID: 66cee1f06e04e4b71735106b34dca9f0@172.25.18.66 CSeq: 111 REGISTER Contact: ;expires=120 Server: Kamailio (1.4.1-notls (i386/linux)) Content-Length: 0 <-------------> [2008-10-08 08:50:14] --- (9 headers 0 lines) --- [2008-10-08 08:50:14] Scheduling destruction of SIP dialog '66cee1f06e04e4b71735106b34dca9f0@172.25.18.66' in 32000 ms (Method: REGISTER) [2008-10-08 08:50:14] NOTICE[3456]: chan_sip.c:12682 handle_response_register: Outbound Registration: Expiry for trieste.tnet.it is 120 sec (Scheduling reregistration in 105 s) [Kaviano*CLI> sip set debug on[K[K [2008-10-08 08:50:19] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK7f7e8ca0;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Date: Wed, 08 Oct 2008 06:50:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3385 IN IP4 172.25.18.67 s=session c=IN IP4 172.25.18.67 t=0 0 m=audio 36912 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:19] --- (14 headers 12 lines) --- [2008-10-08 08:50:19] Sending to 172.25.18.67 : 5060 (NAT) [2008-10-08 08:50:19] Using INVITE request as basis request - 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 [2008-10-08 08:50:19] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK7f7e8ca0;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as3136923b Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="callsrv", nonce="695f92e0" Content-Length: 0 <------------> [2008-10-08 08:50:19] Scheduling destruction of SIP dialog '721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67' in 32000 ms (Method: INVITE) [2008-10-08 08:50:19] Found user 'aviano' [2008-10-08 08:50:19] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK7f7e8ca0;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as3136923b Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK2d715653;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Proxy-Authorization: Digest username="aviano", realm="callsrv", algorithm=MD5, uri="sip:0461842559@172.25.18.66", nonce="695f92e0", response="c3b2a744cee1ebd6d379a4d23f03ea13" Date: Wed, 08 Oct 2008 06:50:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3386 IN IP4 172.25.18.67 s=session c=IN IP4 172.25.18.67 t=0 0 m=audio 36912 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:19] --- (15 headers 12 lines) --- [2008-10-08 08:50:19] Sending to 172.25.18.67 : 5060 (NAT) [2008-10-08 08:50:19] Using INVITE request as basis request - 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 [2008-10-08 08:50:19] Found user 'aviano' [2008-10-08 08:50:19] Found RTP audio format 8 [2008-10-08 08:50:19] Found RTP audio format 101 [2008-10-08 08:50:19] Peer audio RTP is at port 172.25.18.67:36912 [2008-10-08 08:50:19] Found audio description format PCMA for ID 8 [2008-10-08 08:50:19] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:19] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:19] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:19] Peer audio RTP is at port 172.25.18.67:36912 [2008-10-08 08:50:19] Looking for 0461842559 in from-kamailio (domain 172.25.18.66) [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] list_route: hop: [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] <--- Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK2d715653;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [0461842559@from-kamailio:1] NoOp("SIP/aviano-0828ab70", "") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [0461842559@from-kamailio:2] Goto("SIP/aviano-0828ab70", "internals-Cluster|tnn2559|1") in new stack [2008-10-08 08:50:19] -- Goto (internals-Cluster,tnn2559,1) [2008-10-08 08:50:19] -- Executing [tnn2559@internals-Cluster:1] Macro("SIP/aviano-0828ab70", "customdial|tnn2559|45|1|tnn2559@voicemail-NOVAS||10") in new stack [2008-10-08 08:50:19] -- Executing [s@macro-customdial:1] NoOp("SIP/aviano-0828ab70", "") in new stack [2008-10-08 08:50:19] -- Executing [s@macro-customdial:2] Gosub("SIP/aviano-0828ab70", "s-setvariables|1") in new stack [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:1] Set("SIP/aviano-0828ab70", "NomeUtente=tnn2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:2] ExecIf("SIP/aviano-0828ab70", "1|Set|_TimeOuttnn2559=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:3] Set("SIP/aviano-0828ab70", "TempoLimite=45") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:4] Set("SIP/aviano-0828ab70", "NumCalls=1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:5] Set("SIP/aviano-0828ab70", "VoiceMail=tnn2559@voicemail-NOVAS") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:6] Set("SIP/aviano-0828ab70", "NumRicaduta=""") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:7] Set("SIP/aviano-0828ab70", "NumRicaduta=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:8] ExecIf("SIP/aviano-0828ab70", "1|Set|_PICKUPMARK=10") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:9] ExecIf("SIP/aviano-0828ab70", "0|Set|_Distretto=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:10] Set("SIP/aviano-0828ab70", "NumAliases=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:11] Set("SIP/aviano-0828ab70", "Segreteria=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:12] Set("SIP/aviano-0828ab70", "Tecnologia=SIP") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:13] Set("SIP/aviano-0828ab70", "TIMEOUT(absolute)=7200") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Channel will hangup at 2008-10-08 08:50:19 UTC. [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:14] ExecIf("SIP/aviano-0828ab70", "1|Set|_OrigCalled=2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:15] ExecIf("SIP/aviano-0828ab70", "1|Set|_OrigCaller=aviano") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:16] Set("SIP/aviano-0828ab70", "CalledNum=2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:17] Set("SIP/aviano-0828ab70", "CallerNum=aviano") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:18] Set("SIP/aviano-0828ab70", "DialString=SIP/tnn2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:19] ExecIf("SIP/aviano-0828ab70", "0|Set|CALLERID(name)=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:20] ExecIf("SIP/aviano-0828ab70", "0|Set|CALLERID(num)=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:21] ExecIf("SIP/aviano-0828ab70", "0|Set|CALLERID(num)=NP") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:22] ExecIf("SIP/aviano-0828ab70", "1|Set|ExternalCaller=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:23] ExecIf("SIP/aviano-0828ab70", "1|Set|Richiamata=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:24] Set("SIP/aviano-0828ab70", "VoicemailParam=u") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:25] ExecIf("SIP/aviano-0828ab70", "1|Set|_NoHangtnn2559=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:26] Set("SIP/aviano-0828ab70", "WelcomeMsg=""") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s-setvariables@macro-customdial:27] Return("SIP/aviano-0828ab70", "") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:3] GotoIf("SIP/aviano-0828ab70", "1?dnd") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Goto (macro-customdial,s,14) [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:14] GotoIf("SIP/aviano-0828ab70", "0?s-BUSY|1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:15] GotoIf("SIP/aviano-0828ab70", "1?aliases") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Goto (macro-customdial,s,18) [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:18] GotoIf("SIP/aviano-0828ab70", "1?curcalls") in new stack [2008-10-08 08:50:19] -- Goto (macro-customdial,s,24) [2008-10-08 08:50:19] -- Executing [s@macro-customdial:24] GotoIf("SIP/aviano-0828ab70", "1?chiama") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Goto (macro-customdial,s,26) [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:26] Set("SIP/aviano-0828ab70", "ControlloLibero=OK") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:27] GotoIf("SIP/aviano-0828ab70", "1?dialutente") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Goto (macro-customdial,s,32) [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- Executing [s@macro-customdial:32] Dial("SIP/aviano-0828ab70", "SIP/tnn2559|45|g") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] Audio is at 172.25.18.66 port 31332 [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] Adding codec 0x8 (alaw) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] Adding non-codec 0x1 (telephone-event) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] Reliably Transmitting (no NAT) to 172.25.18.83:2048: INVITE sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK2b920229;rport From: "Lorenzo Grosselli" ;tag=as7107edd2 To: Contact: Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Date: Wed, 08 Oct 2008 06:50:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3444 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 31332 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2008-10-08 08:50:19] -- Called tnn2559 [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK2b920229;rport=5060 From: "Lorenzo Grosselli" ;tag=as7107edd2 To: "tnn2559" ;tag=o0q76c9fzh Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] -- SIP/tnn2559-082722b0 is ringing [Kaviano*CLI> sip set debug [2008-10-08 08:50:19] <--- Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK2d715653;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:20] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK2b920229;rport=5060 From: "Lorenzo Grosselli" ;tag=as7107edd2 To: "tnn2559" ;tag=o0q76c9fzh Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:20] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:20] -- SIP/tnn2559-082722b0 is ringing [Kaviano*CLI> sip set debug [2008-10-08 08:50:21] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK2b920229;rport=5060 From: "Lorenzo Grosselli" ;tag=as7107edd2 To: "tnn2559" ;tag=o0q76c9fzh Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:21] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:21] -- SIP/tnn2559-082722b0 is ringing [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK2b920229;rport=5060 From: "Lorenzo Grosselli" ;tag=as7107edd2 To: "tnn2559" ;tag=o0q76c9fzh Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 220 v=0 o=root 1815803457 1815803458 IN IP4 172.25.18.83 s=call c=IN IP4 172.25.18.83 t=0 0 m=audio 33680 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] --- (13 headers 11 lines) --- [2008-10-08 08:50:22] Found RTP audio format 8 [2008-10-08 08:50:22] Found RTP audio format 101 [2008-10-08 08:50:22] Peer audio RTP is at port 172.25.18.83:33680 [2008-10-08 08:50:22] Found audio description format pcma for ID 8 [2008-10-08 08:50:22] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:22] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:22] Peer audio RTP is at port 172.25.18.83:33680 [2008-10-08 08:50:22] list_route: hop: [2008-10-08 08:50:22] set_destination: Parsing for address/port to send to [2008-10-08 08:50:22] set_destination: set destination to 172.25.18.83, port 2048 [2008-10-08 08:50:22] Transmitting (no NAT) to 172.25.18.83:2048: ACK sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK37f2ddfd;rport From: "Lorenzo Grosselli" ;tag=as7107edd2 To: ;tag=o0q76c9fzh Contact: Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] -- SIP/tnn2559-082722b0 answered SIP/aviano-0828ab70 [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Audio is at 172.25.18.66 port 37890 [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Adding codec 0x8 (alaw) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Adding non-codec 0x1 (telephone-event) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK2d715653;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3444 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 37890 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] -- Native bridging SIP/aviano-0828ab70 and SIP/tnn2559-082722b0 [2008-10-08 08:50:22] set_destination: Parsing for address/port to send to [2008-10-08 08:50:22] set_destination: set destination to 172.25.18.83, port 2048 [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Audio is at 172.25.18.66 port 31332 [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Adding codec 0x8 (alaw) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Adding non-codec 0x1 (telephone-event) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] Reliably Transmitting (no NAT) to 172.25.18.83:2048: INVITE sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK02bb3830;rport From: "Lorenzo Grosselli" ;tag=as7107edd2 To: ;tag=o0q76c9fzh Contact: Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3445 IN IP4 172.25.18.67 s=session c=IN IP4 172.25.18.67 t=0 0 m=audio 36912 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK2fdaaf55;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:22] --- (10 headers 0 lines) --- [2008-10-08 08:50:22] set_destination: Parsing for address/port to send to [2008-10-08 08:50:22] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:22] Audio is at 172.25.18.66 port 37890 [2008-10-08 08:50:22] Adding codec 0x8 (alaw) to SDP [2008-10-08 08:50:22] Adding non-codec 0x1 (telephone-event) to SDP [2008-10-08 08:50:22] Reliably Transmitting (no NAT) to 172.25.18.67:5060: INVITE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3445 IN IP4 172.25.18.83 s=session c=IN IP4 172.25.18.83 t=0 0 m=audio 33680 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2008-10-08 08:50:22] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3387 IN IP4 172.25.18.60 s=session c=IN IP4 172.25.18.60 t=0 0 m=audio 16428 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:22] --- (13 headers 12 lines) --- [2008-10-08 08:50:22] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [2008-10-08 08:50:22] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [2008-10-08 08:50:22] --- (12 headers 0 lines) --- [2008-10-08 08:50:22] set_destination: Parsing for address/port to send to [2008-10-08 08:50:22] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:22] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:22] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:22] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:22] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK02bb3830;rport=5060 From: "Lorenzo Grosselli" ;tag=as7107edd2 To: "tnn2559" ;tag=o0q76c9fzh Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 220 v=0 o=root 1815803457 1815803459 IN IP4 172.25.18.83 s=call c=IN IP4 172.25.18.83 t=0 0 m=audio 33680 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:22] --- (13 headers 11 lines) --- [2008-10-08 08:50:22] Found RTP audio format 8 [2008-10-08 08:50:22] Found RTP audio format 101 [2008-10-08 08:50:22] Peer audio RTP is at port 172.25.18.83:33680 [2008-10-08 08:50:22] Found audio description format pcma for ID 8 [2008-10-08 08:50:22] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:22] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:22] Peer audio RTP is at port 172.25.18.83:33680 [2008-10-08 08:50:22] set_destination: Parsing for address/port to send to [2008-10-08 08:50:22] set_destination: set destination to 172.25.18.83, port 2048 [2008-10-08 08:50:22] Transmitting (no NAT) to 172.25.18.83:2048: ACK sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK1c8cf7a4;rport From: "Lorenzo Grosselli" ;tag=as7107edd2 To: ;tag=o0q76c9fzh Contact: Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 103 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:23] Retransmitting #1 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:23] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:23] --- (10 headers 0 lines) --- [2008-10-08 08:50:23] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [2008-10-08 08:50:23] --- (12 headers 0 lines) --- [2008-10-08 08:50:23] set_destination: Parsing for address/port to send to [2008-10-08 08:50:23] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:23] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:24] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:24] --- (12 headers 0 lines) --- [2008-10-08 08:50:24] set_destination: Parsing for address/port to send to [2008-10-08 08:50:24] set_destination: set destination to 172.25.18.67, port 5060 [Kaviano*CLI> sip set debug [2008-10-08 08:50:24] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:24] Retransmitting #2 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:24] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:24] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:26] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [2008-10-08 08:50:26] --- (12 headers 0 lines) --- [2008-10-08 08:50:26] set_destination: Parsing for address/port to send to [2008-10-08 08:50:26] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:26] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK65c19f6f;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:26] Retransmitting #3 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:26] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:26] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] <--- SIP read from 172.25.18.83:2048 ---> BYE sip:aviano@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-gdpr2k7coau9;rport From: "tnn2559" ;tag=o0q76c9fzh To: "Lorenzo Grosselli" ;tag=as7107edd2 Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 1 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom300/7.1.30 RTP-RxStat: Total_Rx_Pkts=283,Rx_Pkts=283,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=282,Tx_Pkts=282,Remote_Tx_Pkts=251 Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] --- (12 headers 0 lines) --- [2008-10-08 08:50:28] Sending to 172.25.18.83 : 2048 (NAT) [2008-10-08 08:50:28] <--- Transmitting (NAT) to 172.25.18.83:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-gdpr2k7coau9;received=172.25.18.83;rport=2048 From: "tnn2559" ;tag=o0q76c9fzh To: "Lorenzo Grosselli" ;tag=as7107edd2 Call-ID: 5e3805250555289d60a263f93a5beac2@172.25.18.66 CSeq: 1 BYE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] set_destination: Parsing for address/port to send to [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] set_destination: set destination to 172.25.18.67, port 5060 [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] Audio is at 172.25.18.66 port 37890 [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] Adding codec 0x8 (alaw) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] Adding non-codec 0x1 (telephone-event) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] Reliably Transmitting (no NAT) to 172.25.18.67:5060: INVITE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK335d5643;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3446 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 37890 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] -- Executing [s@macro-customdial:33] Goto("SIP/aviano-0828ab70", "s-ANSWER|1") in new stack [2008-10-08 08:50:28] -- Goto (macro-customdial,s-ANSWER,1) [2008-10-08 08:50:28] -- Executing [s-ANSWER@macro-customdial:1] NoOp("SIP/aviano-0828ab70", "") in new stack [2008-10-08 08:50:28] -- Executing [s-ANSWER@macro-customdial:2] Set("SIP/aviano-0828ab70", "CDR(userfield)=HANGUP CALLED") in new stack [2008-10-08 08:50:28] -- Executing [s-ANSWER@macro-customdial:3] Hangup("SIP/aviano-0828ab70", "16") in new stack [2008-10-08 08:50:28] == Spawn extension (macro-customdial, s-ANSWER, 3) exited non-zero on 'SIP/aviano-0828ab70' in macro 'customdial' [2008-10-08 08:50:28] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK335d5643;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [2008-10-08 08:50:28] --- (11 headers 0 lines) --- [2008-10-08 08:50:28] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK335d5643;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3388 IN IP4 172.25.18.60 s=session c=IN IP4 172.25.18.60 t=0 0 m=audio 16428 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:28] --- (12 headers 12 lines) --- [2008-10-08 08:50:28] Found RTP audio format 8 [2008-10-08 08:50:28] Found RTP audio format 101 [2008-10-08 08:50:28] Peer audio RTP is at port 172.25.18.60:16428 [2008-10-08 08:50:28] Found audio description format PCMA for ID 8 [2008-10-08 08:50:28] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:28] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:28] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:28] Peer audio RTP is at port 172.25.18.60:16428 [2008-10-08 08:50:28] set_destination: Parsing for address/port to send to [2008-10-08 08:50:28] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:28] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK0cffdf4f;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 103 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:28] set_destination: Parsing for address/port to send to [2008-10-08 08:50:28] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:28] Audio is at 172.25.18.66 port 37890 [2008-10-08 08:50:28] Adding codec 0x8 (alaw) to SDP [2008-10-08 08:50:28] Adding non-codec 0x1 (telephone-event) to SDP [2008-10-08 08:50:28] Reliably Transmitting (no NAT) to 172.25.18.67:5060: INVITE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK468b6d6b;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3447 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 37890 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2008-10-08 08:50:28] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3389 IN IP4 172.25.18.60 s=session c=IN IP4 172.25.18.60 t=0 0 m=audio 16428 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:28] --- (13 headers 12 lines) --- [2008-10-08 08:50:28] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [2008-10-08 08:50:28] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK468b6d6b;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [2008-10-08 08:50:28] --- (12 headers 0 lines) --- [2008-10-08 08:50:28] set_destination: Parsing for address/port to send to [2008-10-08 08:50:28] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:28] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK468b6d6b;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:28] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;rport From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Contact: Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:28] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] == Spawn extension (macro-customdial, s-ANSWER, 3) exited non-zero on 'SIP/aviano-0828ab70' [2008-10-08 08:50:28] -- Executing [h@macro-customdial:1] NoOp("SIP/aviano-0828ab70", "CALLBACK_HANGUP") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:2] NoOp("SIP/aviano-0828ab70", "CALLERID(num) - CHANNEL - CalledNum - DIALEDPEERNUMBER - BLINDTRANSFER - OrigCaller") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:3] NoOp("SIP/aviano-0828ab70", "aviano - SIP/aviano-0828ab70 - 2559 - tnn2559 - - aviano") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:4] GotoIf("SIP/aviano-0828ab70", "0?fine") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:5] ExecIf("SIP/aviano-0828ab70", "0|Set|CalledNum=tnn2") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:6] ExecIf("SIP/aviano-0828ab70", "0|Set|BlindNum=") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:7] GotoIf("SIP/aviano-0828ab70", "0?fine") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:8] GotoIf("SIP/aviano-0828ab70", "0?caller") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:9] GotoIf("SIP/aviano-0828ab70", "0?called") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:10] Set("SIP/aviano-0828ab70", "CalledNum=tnn2") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:11] GotoIf("SIP/aviano-0828ab70", "0?caller") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:12] GotoIf("SIP/aviano-0828ab70", "1?pren2") in new stack [2008-10-08 08:50:28] -- Goto (macro-customdial,h,19) [2008-10-08 08:50:28] -- Executing [h@macro-customdial:19] GotoIf("SIP/aviano-0828ab70", "1?caller") in new stack [2008-10-08 08:50:28] -- Goto (macro-customdial,h,23) [2008-10-08 08:50:28] -- Executing [h@macro-customdial:23] GotoIf("SIP/aviano-0828ab70", "0?blind") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:24] Set("SIP/aviano-0828ab70", "CallerNum=avia") in new stack [2008-10-08 08:50:28] -- Executing [h@macro-customdial:25] GotoIf("SIP/aviano-0828ab70", "1?pren4") in new stack [2008-10-08 08:50:28] -- Goto (macro-customdial,h,32) [2008-10-08 08:50:28] -- Executing [h@macro-customdial:32] GotoIf("SIP/aviano-0828ab70", "1?blind") in new stack [2008-10-08 08:50:28] -- Goto (macro-customdial,h,36) [2008-10-08 08:50:28] -- Executing [h@macro-customdial:36] GotoIf("SIP/aviano-0828ab70", "1?fine") in new stack [2008-10-08 08:50:28] -- Goto (macro-customdial,h,48) [2008-10-08 08:50:28] -- Executing [h@macro-customdial:48] NoOp("SIP/aviano-0828ab70", "Fine Verifica CallBack") in new stack [2008-10-08 08:50:28] Scheduling destruction of SIP dialog '721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67' in 32000 ms (Method: ACK) [2008-10-08 08:50:28] set_destination: Parsing for address/port to send to [2008-10-08 08:50:28] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:28] Reliably Transmitting (no NAT) to 172.25.18.67:5060: BYE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK692f6927;rport From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 BYE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK692f6927;received=172.25.18.66;rport=5060 From: ;tag=as2b97de9c To: "Lorenzo Grosselli" ;tag=as17c42656 Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 BYE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [2008-10-08 08:50:28] --- (11 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:28] Really destroying SIP dialog '5e3805250555289d60a263f93a5beac2@172.25.18.66' Method: BYE [Kaviano*CLI> sip set debug [2008-10-08 08:50:29] Retransmitting #1 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:30] Retransmitting #2 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:30] Retransmitting #4 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:32] Retransmitting #3 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:34] Retransmitting #5 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] <--- SIP read from 172.25.18.83:2048 ---> REGISTER sip:172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-30dplqdgzde8;rport From: "tnn2559" ;tag=h5f7llyzym To: "tnn2559" Call-ID: 3c2670175393-vqxvde04go2x CSeq: 426 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 172.25.18.83 Expires: 600 Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] --- (14 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] Using latest REGISTER request as basis request [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] Sending to 172.25.18.83 : 2048 (NAT) [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] <--- Transmitting (no NAT) to 172.25.18.83:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-30dplqdgzde8;received=172.25.18.83;rport=2048 From: "tnn2559" ;tag=h5f7llyzym To: "tnn2559" Call-ID: 3c2670175393-vqxvde04go2x CSeq: 426 REGISTER User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] <--- Transmitting (no NAT) to 172.25.18.83:2048 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-30dplqdgzde8;received=172.25.18.83;rport=2048 From: "tnn2559" ;tag=h5f7llyzym To: "tnn2559" ;tag=as47120c77 Call-ID: 3c2670175393-vqxvde04go2x CSeq: 426 REGISTER User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="callsrv", nonce="37c51cb2" Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:35] Scheduling destruction of SIP dialog '3c2670175393-vqxvde04go2x' in 32000 ms (Method: REGISTER) [Kaviano*CLI> sip set debug [2008-10-08 08:50:36] <--- SIP read from 172.25.18.83:2048 ---> REGISTER sip:172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-tex4wf7hshmi;rport From: "tnn2559" ;tag=h5f7llyzym To: "tnn2559" Call-ID: 3c2670175393-vqxvde04go2x CSeq: 427 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/7.1.30 Supported: gruu Allow-Events: dialog X-Real-IP: 172.25.18.83 Authorization: Digest username="tnn2559",realm="callsrv",nonce="37c51cb2",uri="sip:172.25.18.66",response="061611bdfad3c6a9f7fe54fdf26d0275",algorithm=MD5 Expires: 600 Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:36] --- (15 headers 0 lines) --- [2008-10-08 08:50:36] Using latest REGISTER request as basis request [2008-10-08 08:50:36] Sending to 172.25.18.83 : 2048 (NAT) [2008-10-08 08:50:36] <--- Transmitting (no NAT) to 172.25.18.83:2048 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-tex4wf7hshmi;received=172.25.18.83;rport=2048 From: "tnn2559" ;tag=h5f7llyzym To: "tnn2559" Call-ID: 3c2670175393-vqxvde04go2x CSeq: 427 REGISTER User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:36] <--- Transmitting (no NAT) to 172.25.18.83:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-tex4wf7hshmi;received=172.25.18.83;rport=2048 From: "tnn2559" ;tag=h5f7llyzym To: "tnn2559" ;tag=as47120c77 Call-ID: 3c2670175393-vqxvde04go2x CSeq: 427 REGISTER User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 600 Contact: ;expires=600 Date: Wed, 08 Oct 2008 06:50:36 GMT Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:36] Scheduling destruction of SIP dialog '3c2670175393-vqxvde04go2x' in 32000 ms (Method: REGISTER) [Kaviano*CLI> sip set debug [2008-10-08 08:50:36] Retransmitting #4 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:38] Retransmitting #6 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK078f5795;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] Retransmitting #5 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK02d2b3cd;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Date: Wed, 08 Oct 2008 06:50:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3385 IN IP4 172.25.18.67 s=session c=IN IP4 172.25.18.67 t=0 0 m=audio 34508 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] --- (14 headers 12 lines) --- [2008-10-08 08:50:40] Sending to 172.25.18.67 : 5060 (NAT) [2008-10-08 08:50:40] Using INVITE request as basis request - 63623e8230ea1c267015915a7b77284b@172.25.18.67 [2008-10-08 08:50:40] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK02d2b3cd;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as01db12ad Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="callsrv", nonce="7770c105" Content-Length: 0 <------------> [2008-10-08 08:50:40] Scheduling destruction of SIP dialog '63623e8230ea1c267015915a7b77284b@172.25.18.67' in 32000 ms (Method: INVITE) [2008-10-08 08:50:40] Found user 'aviano' [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK02d2b3cd;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as01db12ad Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:40] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK51e4e86f;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Proxy-Authorization: Digest username="aviano", realm="callsrv", algorithm=MD5, uri="sip:0461842559@172.25.18.66", nonce="7770c105", response="fb9b3139070af08a3187c511be819e01" Date: Wed, 08 Oct 2008 06:50:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3386 IN IP4 172.25.18.67 s=session c=IN IP4 172.25.18.67 t=0 0 m=audio 34508 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:40] --- (15 headers 12 lines) --- [2008-10-08 08:50:40] Sending to 172.25.18.67 : 5060 (NAT) [2008-10-08 08:50:40] Using INVITE request as basis request - 63623e8230ea1c267015915a7b77284b@172.25.18.67 [2008-10-08 08:50:40] Found user 'aviano' [2008-10-08 08:50:40] Found RTP audio format 8 [2008-10-08 08:50:40] Found RTP audio format 101 [2008-10-08 08:50:40] Peer audio RTP is at port 172.25.18.67:34508 [2008-10-08 08:50:40] Found audio description format PCMA for ID 8 [2008-10-08 08:50:40] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:40] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:40] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:40] Peer audio RTP is at port 172.25.18.67:34508 [2008-10-08 08:50:40] Looking for 0461842559 in from-kamailio (domain 172.25.18.66) [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] list_route: hop: [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK51e4e86f;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [0461842559@from-kamailio:1] NoOp("SIP/aviano-b7302078", "") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [0461842559@from-kamailio:2] Goto("SIP/aviano-b7302078", "internals-Cluster|tnn2559|1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Goto (internals-Cluster,tnn2559,1) [2008-10-08 08:50:40] -- Executing [tnn2559@internals-Cluster:1] Macro("SIP/aviano-b7302078", "customdial|tnn2559|45|1|tnn2559@voicemail-NOVAS||10") in new stack [2008-10-08 08:50:40] -- Executing [s@macro-customdial:1] NoOp("SIP/aviano-b7302078", "") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s@macro-customdial:2] Gosub("SIP/aviano-b7302078", "s-setvariables|1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:1] Set("SIP/aviano-b7302078", "NomeUtente=tnn2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:2] ExecIf("SIP/aviano-b7302078", "1|Set|_TimeOuttnn2559=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:3] Set("SIP/aviano-b7302078", "TempoLimite=45") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:4] Set("SIP/aviano-b7302078", "NumCalls=1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:5] Set("SIP/aviano-b7302078", "VoiceMail=tnn2559@voicemail-NOVAS") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:6] Set("SIP/aviano-b7302078", "NumRicaduta=""") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:7] Set("SIP/aviano-b7302078", "NumRicaduta=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:8] ExecIf("SIP/aviano-b7302078", "1|Set|_PICKUPMARK=10") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:9] ExecIf("SIP/aviano-b7302078", "0|Set|_Distretto=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:10] Set("SIP/aviano-b7302078", "NumAliases=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:11] Set("SIP/aviano-b7302078", "Segreteria=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:12] Set("SIP/aviano-b7302078", "Tecnologia=SIP") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:13] Set("SIP/aviano-b7302078", "TIMEOUT(absolute)=7200") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Channel will hangup at 2008-10-08 08:50:40 UTC. [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:14] ExecIf("SIP/aviano-b7302078", "1|Set|_OrigCalled=2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:15] ExecIf("SIP/aviano-b7302078", "1|Set|_OrigCaller=aviano") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:16] Set("SIP/aviano-b7302078", "CalledNum=2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:17] Set("SIP/aviano-b7302078", "CallerNum=aviano") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:18] Set("SIP/aviano-b7302078", "DialString=SIP/tnn2559") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:19] ExecIf("SIP/aviano-b7302078", "0|Set|CALLERID(name)=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:20] ExecIf("SIP/aviano-b7302078", "0|Set|CALLERID(num)=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:21] ExecIf("SIP/aviano-b7302078", "0|Set|CALLERID(num)=NP") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:22] ExecIf("SIP/aviano-b7302078", "1|Set|ExternalCaller=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:23] ExecIf("SIP/aviano-b7302078", "1|Set|Richiamata=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:24] Set("SIP/aviano-b7302078", "VoicemailParam=u") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:25] ExecIf("SIP/aviano-b7302078", "1|Set|_NoHangtnn2559=0") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:26] Set("SIP/aviano-b7302078", "WelcomeMsg=""") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s-setvariables@macro-customdial:27] Return("SIP/aviano-b7302078", "") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s@macro-customdial:3] GotoIf("SIP/aviano-b7302078", "1?dnd") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Goto (macro-customdial,s,14) [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s@macro-customdial:14] GotoIf("SIP/aviano-b7302078", "0?s-BUSY|1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s@macro-customdial:15] GotoIf("SIP/aviano-b7302078", "1?aliases") in new stack [2008-10-08 08:50:40] -- Goto (macro-customdial,s,18) [2008-10-08 08:50:40] -- Executing [s@macro-customdial:18] GotoIf("SIP/aviano-b7302078", "1?curcalls") in new stack [2008-10-08 08:50:40] -- Goto (macro-customdial,s,24) [2008-10-08 08:50:40] -- Executing [s@macro-customdial:24] GotoIf("SIP/aviano-b7302078", "1?chiama") in new stack [2008-10-08 08:50:40] -- Goto (macro-customdial,s,26) [2008-10-08 08:50:40] -- Executing [s@macro-customdial:26] Set("SIP/aviano-b7302078", "ControlloLibero=OK") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s@macro-customdial:27] GotoIf("SIP/aviano-b7302078", "1?dialutente") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Goto (macro-customdial,s,32) [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Executing [s@macro-customdial:32] Dial("SIP/aviano-b7302078", "SIP/tnn2559|45|g") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] Audio is at 172.25.18.66 port 34200 [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] Adding codec 0x8 (alaw) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] Adding non-codec 0x1 (telephone-event) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] Reliably Transmitting (no NAT) to 172.25.18.83:2048: INVITE sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK40819be5;rport From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: Contact: Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Date: Wed, 08 Oct 2008 06:50:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3444 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 34200 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- Called tnn2559 [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK40819be5;rport=5060 From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: "tnn2559" ;tag=rpivw02s6q Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- SIP/tnn2559-082722b0 is ringing [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK51e4e86f;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK40819be5;rport=5060 From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: "tnn2559" ;tag=rpivw02s6q Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:40] -- SIP/tnn2559-082722b0 is ringing [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK40819be5;rport=5060 From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: "tnn2559" ;tag=rpivw02s6q Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 218 v=0 o=root 250668101 250668102 IN IP4 172.25.18.83 s=call c=IN IP4 172.25.18.83 t=0 0 m=audio 35572 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:41] --- (13 headers 11 lines) --- [2008-10-08 08:50:41] Found RTP audio format 8 [2008-10-08 08:50:41] Found RTP audio format 101 [2008-10-08 08:50:41] Peer audio RTP is at port 172.25.18.83:35572 [2008-10-08 08:50:41] Found audio description format pcma for ID 8 [2008-10-08 08:50:41] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:41] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:41] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:41] Peer audio RTP is at port 172.25.18.83:35572 [2008-10-08 08:50:41] list_route: hop: [2008-10-08 08:50:41] set_destination: Parsing for address/port to send to [2008-10-08 08:50:41] set_destination: set destination to 172.25.18.83, port 2048 [2008-10-08 08:50:41] Transmitting (no NAT) to 172.25.18.83:2048: ACK sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK0f8e3a42;rport From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: ;tag=rpivw02s6q Contact: Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] -- SIP/tnn2559-082722b0 answered SIP/aviano-b7302078 [2008-10-08 08:50:41] Audio is at 172.25.18.66 port 30478 [2008-10-08 08:50:41] Adding codec 0x8 (alaw) to SDP [2008-10-08 08:50:41] Adding non-codec 0x1 (telephone-event) to SDP [2008-10-08 08:50:41] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK51e4e86f;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3444 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 30478 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2008-10-08 08:50:41] -- Native bridging SIP/aviano-b7302078 and SIP/tnn2559-082722b0 [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] set_destination: Parsing for address/port to send to [2008-10-08 08:50:41] set_destination: set destination to 172.25.18.83, port 2048 [2008-10-08 08:50:41] Audio is at 172.25.18.66 port 34200 [2008-10-08 08:50:41] Adding codec 0x8 (alaw) to SDP [2008-10-08 08:50:41] Adding non-codec 0x1 (telephone-event) to SDP [2008-10-08 08:50:41] Reliably Transmitting (no NAT) to 172.25.18.83:2048: INVITE sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK73a8bf2c;rport From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: ;tag=rpivw02s6q Contact: Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3445 IN IP4 172.25.18.67 s=session c=IN IP4 172.25.18.67 t=0 0 m=audio 34508 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK6009f620;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:41] --- (10 headers 0 lines) --- [2008-10-08 08:50:41] set_destination: Parsing for address/port to send to [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:41] Audio is at 172.25.18.66 port 30478 [2008-10-08 08:50:41] Adding codec 0x8 (alaw) to SDP [2008-10-08 08:50:41] Adding non-codec 0x1 (telephone-event) to SDP [2008-10-08 08:50:41] Reliably Transmitting (no NAT) to 172.25.18.67:5060: INVITE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3445 IN IP4 172.25.18.83 s=session c=IN IP4 172.25.18.83 t=0 0 m=audio 35572 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2008-10-08 08:50:41] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3387 IN IP4 172.25.18.60 s=session c=IN IP4 172.25.18.60 t=0 0 m=audio 16430 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:41] --- (13 headers 12 lines) --- [2008-10-08 08:50:41] <--- Reliably Transmitting (no NAT) to 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [2008-10-08 08:50:41] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] --- (12 headers 0 lines) --- [2008-10-08 08:50:41] set_destination: Parsing for address/port to send to [2008-10-08 08:50:41] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:41] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:41] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:41] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:41] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK73a8bf2c;rport=5060 From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: "tnn2559" ;tag=rpivw02s6q Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 218 v=0 o=root 250668101 250668103 IN IP4 172.25.18.83 s=call c=IN IP4 172.25.18.83 t=0 0 m=audio 35572 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:41] --- (13 headers 11 lines) --- [2008-10-08 08:50:41] Found RTP audio format 8 [2008-10-08 08:50:41] Found RTP audio format 101 [2008-10-08 08:50:41] Peer audio RTP is at port 172.25.18.83:35572 [2008-10-08 08:50:41] Found audio description format pcma for ID 8 [2008-10-08 08:50:41] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:41] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:41] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:41] Peer audio RTP is at port 172.25.18.83:35572 [2008-10-08 08:50:41] set_destination: Parsing for address/port to send to [2008-10-08 08:50:41] set_destination: set destination to 172.25.18.83, port 2048 [2008-10-08 08:50:41] Transmitting (no NAT) to 172.25.18.83:2048: ACK sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK1563d584;rport From: "Lorenzo Grosselli" ;tag=as7bb3a0e6 To: ;tag=rpivw02s6q Contact: Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 103 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] WARNING[3456]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 for seqno 104 (Critical Response) -- See doc/sip-retransmit.txt. [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] --- (12 headers 0 lines) --- [2008-10-08 08:50:42] set_destination: Parsing for address/port to send to [2008-10-08 08:50:42] set_destination: set destination to 172.25.18.67, port 5060 [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] Retransmitting #1 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:42] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:43] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:43] --- (12 headers 0 lines) --- [2008-10-08 08:50:43] set_destination: Parsing for address/port to send to [2008-10-08 08:50:43] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:43] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:43] Retransmitting #2 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:43] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [2008-10-08 08:50:43] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:44] Retransmitting #6 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK66b14b49;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as17c42656 To: ;tag=as2b97de9c Call-ID: 721d422c2daad2b23f5f4b8e09c4ee46@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] Scheduling destruction of SIP dialog '586e7b6b5bde8c9f412fd4a55dc4dca4@172.25.18.66' in 32000 ms (Method: NOTIFY) [2008-10-08 08:50:45] Reliably Transmitting (no NAT) to 172.25.18.83:2048: NOTIFY sip:tnn2559@172.25.18.83:2048;line=ul089bda SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK7a53b9f2;rport From: "asterisk" ;tag=as188b8986 To: Contact: Call-ID: 586e7b6b5bde8c9f412fd4a55dc4dca4@172.25.18.66 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 87 Messages-Waiting: no Message-Account: sip:100@172.25.18.66 Voice-Message: 0/0 (0/0) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [2008-10-08 08:50:45] --- (12 headers 0 lines) --- [2008-10-08 08:50:45] set_destination: Parsing for address/port to send to [2008-10-08 08:50:45] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:45] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK550e0209;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 102 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:45] Retransmitting #3 (no NAT) to 172.25.18.67:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK650aa571;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] --- (10 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] <--- SIP read from 172.25.18.83:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK7a53b9f2;rport=5060 From: "asterisk" ;tag=as188b8986 To: Call-ID: 586e7b6b5bde8c9f412fd4a55dc4dca4@172.25.18.66 CSeq: 102 NOTIFY Content-Length: 0 <-------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] --- (7 headers 0 lines) --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:45] Really destroying SIP dialog '586e7b6b5bde8c9f412fd4a55dc4dca4@172.25.18.66' Method: NOTIFY [Kaviano*CLI> sip set debug [2008-10-08 08:50:46] Really destroying SIP dialog '66cee1f06e04e4b71735106b34dca9f0@172.25.18.66' Method: REGISTER [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] <--- SIP read from 172.25.18.83:2048 ---> BYE sip:aviano@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-z8xcmeg3t8k5;rport From: "tnn2559" ;tag=rpivw02s6q To: "Lorenzo Grosselli" ;tag=as7bb3a0e6 Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 1 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom300/7.1.30 RTP-RxStat: Total_Rx_Pkts=286,Rx_Pkts=286,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=286,Tx_Pkts=286,Remote_Tx_Pkts=251 Content-Length: 0 <-------------> [2008-10-08 08:50:47] --- (12 headers 0 lines) --- [2008-10-08 08:50:47] Sending to 172.25.18.83 : 2048 (NAT) [2008-10-08 08:50:47] <--- Transmitting (NAT) to 172.25.18.83:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.83:2048;branch=z9hG4bK-z8xcmeg3t8k5;received=172.25.18.83;rport=2048 From: "tnn2559" ;tag=rpivw02s6q To: "Lorenzo Grosselli" ;tag=as7bb3a0e6 Call-ID: 77d147ef581617122306c63d53f713b7@172.25.18.66 CSeq: 1 BYE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] set_destination: Parsing for address/port to send to [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] set_destination: set destination to 172.25.18.67, port 5060 [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] Audio is at 172.25.18.66 port 30478 [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] Adding codec 0x8 (alaw) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] Adding non-codec 0x1 (telephone-event) to SDP [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] Reliably Transmitting (no NAT) to 172.25.18.67:5060: INVITE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK32d215dd;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 3444 3446 IN IP4 172.25.18.66 s=session c=IN IP4 172.25.18.66 t=0 0 m=audio 30478 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [s@macro-customdial:33] Goto("SIP/aviano-b7302078", "s-ANSWER|1") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Goto (macro-customdial,s-ANSWER,1) [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [s-ANSWER@macro-customdial:1] NoOp("SIP/aviano-b7302078", "") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [s-ANSWER@macro-customdial:2] Set("SIP/aviano-b7302078", "CDR(userfield)=HANGUP CALLED") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [s-ANSWER@macro-customdial:3] Hangup("SIP/aviano-b7302078", "16") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] == Spawn extension (macro-customdial, s-ANSWER, 3) exited non-zero on 'SIP/aviano-b7302078' in macro 'customdial' [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] == Spawn extension (macro-customdial, s-ANSWER, 3) exited non-zero on 'SIP/aviano-b7302078' [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:1] NoOp("SIP/aviano-b7302078", "CALLBACK_HANGUP") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:2] NoOp("SIP/aviano-b7302078", "CALLERID(num) - CHANNEL - CalledNum - DIALEDPEERNUMBER - BLINDTRANSFER - OrigCaller") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:3] NoOp("SIP/aviano-b7302078", "aviano - SIP/aviano-b7302078 - 2559 - tnn2559 - - aviano") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:4] GotoIf("SIP/aviano-b7302078", "0?fine") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:5] ExecIf("SIP/aviano-b7302078", "0|Set|CalledNum=tnn2") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:6] ExecIf("SIP/aviano-b7302078", "0|Set|BlindNum=") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:7] GotoIf("SIP/aviano-b7302078", "0?fine") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:8] GotoIf("SIP/aviano-b7302078", "0?caller") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:9] GotoIf("SIP/aviano-b7302078", "0?called") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:10] Set("SIP/aviano-b7302078", "CalledNum=tnn2") in new stack [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] -- Executing [h@macro-customdial:11] GotoIf("SIP/aviano-b7302078", "0?caller") in new stack [2008-10-08 08:50:47] -- Executing [h@macro-customdial:12] GotoIf("SIP/aviano-b7302078", "1?pren2") in new stack [2008-10-08 08:50:47] -- Goto (macro-customdial,h,19) [2008-10-08 08:50:47] -- Executing [h@macro-customdial:19] GotoIf("SIP/aviano-b7302078", "1?caller") in new stack [2008-10-08 08:50:47] -- Goto (macro-customdial,h,23) [2008-10-08 08:50:47] -- Executing [h@macro-customdial:23] GotoIf("SIP/aviano-b7302078", "0?blind") in new stack [2008-10-08 08:50:47] -- Executing [h@macro-customdial:24] Set("SIP/aviano-b7302078", "CallerNum=avia") in new stack [2008-10-08 08:50:47] -- Executing [h@macro-customdial:25] GotoIf("SIP/aviano-b7302078", "1?pren4") in new stack [2008-10-08 08:50:47] -- Goto (macro-customdial,h,32) [2008-10-08 08:50:47] -- Executing [h@macro-customdial:32] GotoIf("SIP/aviano-b7302078", "1?blind") in new stack [2008-10-08 08:50:47] -- Goto (macro-customdial,h,36) [2008-10-08 08:50:47] -- Executing [h@macro-customdial:36] GotoIf("SIP/aviano-b7302078", "1?fine") in new stack [2008-10-08 08:50:47] -- Goto (macro-customdial,h,48) [2008-10-08 08:50:47] -- Executing [h@macro-customdial:48] NoOp("SIP/aviano-b7302078", "Fine Verifica CallBack") in new stack [2008-10-08 08:50:47] Scheduling destruction of SIP dialog '63623e8230ea1c267015915a7b77284b@172.25.18.67' in 32000 ms (Method: ACK) [2008-10-08 08:50:47] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK32d215dd;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [2008-10-08 08:50:47] --- (11 headers 0 lines) --- [2008-10-08 08:50:47] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK32d215dd;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3388 IN IP4 172.25.18.60 s=session c=IN IP4 172.25.18.60 t=0 0 m=audio 16430 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:47] --- (12 headers 12 lines) --- [2008-10-08 08:50:47] Found RTP audio format 8 [2008-10-08 08:50:47] Found RTP audio format 101 [2008-10-08 08:50:47] Peer audio RTP is at port 172.25.18.60:16430 [2008-10-08 08:50:47] Found audio description format PCMA for ID 8 [2008-10-08 08:50:47] Found audio description format telephone-event for ID 101 [2008-10-08 08:50:47] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [2008-10-08 08:50:47] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2008-10-08 08:50:47] Peer audio RTP is at port 172.25.18.60:16430 [2008-10-08 08:50:47] set_destination: Parsing for address/port to send to [2008-10-08 08:50:47] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:47] Transmitting (no NAT) to 172.25.18.67:5060: ACK sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK04835ea6;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 103 ACK User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:47] set_destination: Parsing for address/port to send to [2008-10-08 08:50:47] set_destination: set destination to 172.25.18.67, port 5060 [2008-10-08 08:50:47] Reliably Transmitting (no NAT) to 172.25.18.67:5060: BYE sip:aviano@172.25.18.67 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK347afc7a;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 BYE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Content-Length: 0 --- [2008-10-08 08:50:47] Scheduling destruction of SIP dialog '63623e8230ea1c267015915a7b77284b@172.25.18.67' in 32000 ms (Method: ACK) [Kaviano*CLI> sip set debug [2008-10-08 08:50:47] <--- SIP read from 172.25.18.67:5060 ---> INVITE sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK37f43359;rport From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Contact: Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 3385 3389 IN IP4 172.25.18.60 s=session c=IN IP4 172.25.18.60 t=0 0 m=audio 16430 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2008-10-08 08:50:47] --- (13 headers 12 lines) --- [2008-10-08 08:50:47] Sending to 172.25.18.67 : 5060 (NAT) [2008-10-08 08:50:47] Using INVITE request as basis request - 63623e8230ea1c267015915a7b77284b@172.25.18.67 [2008-10-08 08:50:47] NOTICE[3456]: chan_sip.c:14615 handle_request_invite: Unable to create/find SIP channel for this INVITE [2008-10-08 08:50:47] <--- Reliably Transmitting (NAT) to 172.25.18.67:5060 ---> SIP/2.0 503 Unavailable Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK37f43359;received=172.25.18.67;rport=5060 From: "Lorenzo Grosselli" ;tag=as5a33c665 To: ;tag=as5aa1ffc2 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [2008-10-08 08:50:47] Scheduling destruction of SIP dialog '63623e8230ea1c267015915a7b77284b@172.25.18.67' in 32000 ms (Method: INVITE) [2008-10-08 08:50:47] <--- SIP read from 172.25.18.67:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.66:5060;branch=z9hG4bK347afc7a;received=172.25.18.66;rport=5060 From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Call-ID: 63623e8230ea1c267015915a7b77284b@172.25.18.67 CSeq: 104 BYE User-Agent: Asterisk PBX 1.4 Test Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [2008-10-08 08:50:47] --- (11 headers 0 lines) --- [2008-10-08 08:50:47] <--- SIP read from 172.25.18.67:5060 ---> ACK sip:0461842559@172.25.18.66 SIP/2.0 Via: SIP/2.0/UDP 172.25.18.67:5060;branch=z9hG4bK37f43359;rport From: ;tag=as5aa1ffc2 To: "Lorenzo Grosselli" ;tag=as5a33c665 Contact: Call-ID: 63623