Feb 2 10:00:11 DEBUG[3799] chan_sip.c: Call '0904701d3d717fb3277e3df22e1ffa2a@193.227.71.32' has no history Feb 2 10:00:11 DEBUG[3799] chan_sip.c: ---------- END SIP HISTORY for '0904701d3d717fb3277e3df22e1ffa2a@193.227.71.32' Feb 2 10:00:11 DEBUG[3799] chan_sip.c: ---------- SIP HISTORY for '45c03a4c503a5f981d31a1892501c46d@193.227.71.32' Feb 2 10:00:11 DEBUG[3799] chan_sip.c: * SIP Call Feb 2 10:00:11 DEBUG[3799] chan_sip.c: Call '45c03a4c503a5f981d31a1892501c46d@193.227.71.32' has no history Feb 2 10:00:11 DEBUG[3799] chan_sip.c: ---------- END SIP HISTORY for '45c03a4c503a5f981d31a1892501c46d@193.227.71.32' Feb 2 10:00:21 VERBOSE[3767] logger.c: -- Remote UNIX connection Feb 2 10:00:39 VERBOSE[3799] logger.c: <-- SIP read from 10.97.1.254:5060: OPTIONS sip:193.227.71.32 SIP/2.0 Via: SIP/2.0/UDP 10.97.1.254:5060;branch=z9hG4bK78e0600f;rport From: "Unknown" ;tag=as390427e4 To: Contact: Call-ID: 49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 09:00:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 0: OPTIONS sip:193.227.71.32 SIP/2.0 (33) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.97.1.254:5060;branch=z9hG4bK78e0600f;rport (62) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 2: From: "Unknown" ;tag=as390427e4 (56) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 3: To: (23) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 4: Contact: (34) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254 (53) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:00:34 GMT (35) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 11: Content-Length: 0 (17) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:00:39 VERBOSE[3799] logger.c: --- (12 headers 0 lines) --- Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Allocating new SIP dialog for 49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254 - OPTIONS (No RTP) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS Feb 2 10:00:39 VERBOSE[3799] logger.c: Looking for s in from-internal (domain 193.227.71.32) Feb 2 10:00:39 VERBOSE[3799] logger.c: Transmitting (no NAT) to 10.97.1.254:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.97.1.254:5060;branch=z9hG4bK78e0600f;received=10.97.1.254;rport=5060 From: "Unknown" ;tag=as390427e4 To: ;tag=as58053344 Call-ID: 49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.97.1.254:5060;branch=z9hG4bK78e0600f;received=10.97.1.254;rport=5060 (88) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 2: From: "Unknown" ;tag=as390427e4 (56) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=as58053344 (38) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 4: Call-ID: 49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254 (53) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 8: Contact: (28) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 9: Accept: application/sdp (23) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:00:39 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:00:39 VERBOSE[3799] logger.c: Destroying call '49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254' Feb 2 10:00:39 DEBUG[3799] chan_sip.c: ---------- SIP HISTORY for '49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254' Feb 2 10:00:39 DEBUG[3799] chan_sip.c: * SIP Call Feb 2 10:00:39 DEBUG[3799] chan_sip.c: 1. Rx OPTIONS / 102 OPTIONS /sip:193.227.71.32 Feb 2 10:00:39 DEBUG[3799] chan_sip.c: 2. TxResp SIP/2.0 / 102 OPTIONS Feb 2 10:00:39 DEBUG[3799] chan_sip.c: ---------- END SIP HISTORY for '49dbd6a27ab19e350ac94f3035df3c1b@10.97.1.254' Feb 2 10:00:44 VERBOSE[3799] logger.c: <-- SIP read from 10.193.71.156:5060: INVITE sip:855@193.227.71.32 SIP/2.0 Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60 From: "Andrea Ballin" ;tag=b91d4746859371f2 To: Contact: Supported: replaces, timer Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49139 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 311 v=0 o=66899 8000 8000 IN IP4 10.193.71.156 s=SIP Call c=IN IP4 10.193.71.156 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 0: INVITE sip:855@193.227.71.32 SIP/2.0 (36) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60 (66) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 3: To: (27) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 4: Contact: (39) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 5: Supported: replaces, timer (26) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 6: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 7: CSeq: 49139 INVITE (18) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 9: Max-Forwards: 70 (16) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 11: Content-Type: application/sdp (29) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 12: Content-Length: 311 (19) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 13: (0) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: v=0 (3) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: o=66899 8000 8000 IN IP4 10.193.71.156 (38) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: s=SIP Call (10) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: c=IN IP4 10.193.71.156 (22) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: m=audio 5004 RTP/AVP 0 8 4 18 3 101 (35) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=sendrecv (10) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=ptime:20 (10) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=fmtp:101 0-11 (15) Feb 2 10:00:44 VERBOSE[3799] logger.c: --- (13 headers 15 lines) --- Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Allocating new SIP dialog for af10d5446545f719@10.193.71.156 - INVITE (With RTP) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Begin: parsing SIP "Supported: replaces, timer" Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Found SIP option: -replaces- Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Matched SIP option: replaces Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Found SIP option: -timer- Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Matched SIP option: timer Feb 2 10:00:44 DEBUG[3799] chan_sip.c: * SIP extension value: 5 for call af10d5446545f719@10.193.71.156 Feb 2 10:00:44 VERBOSE[3799] logger.c: Using INVITE request as basis request - af10d5446545f719@10.193.71.156 Feb 2 10:00:44 VERBOSE[3799] logger.c: Sending to 10.193.71.156 : 5060 (non-NAT) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Setting NAT on RTP to 0 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found user '66899' Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 0 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 8 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 4 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 18 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 3 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 101 Feb 2 10:00:44 VERBOSE[3799] logger.c: Peer audio RTP is at port 10.193.71.156:5004 Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Peer audio RTP is at port 10.193.71.156:5004 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format PCMU Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format PCMA Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format G723 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format G729 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format GSM Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format telephone-event Feb 2 10:00:44 VERBOSE[3799] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Feb 2 10:00:44 VERBOSE[3799] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Checking SIP call limits for device 66899 Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Updating call counter for incoming call Feb 2 10:00:44 VERBOSE[3799] logger.c: Looking for 855 in from-internal (domain 193.227.71.32) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: build_route: Contact hop: Feb 2 10:00:44 VERBOSE[3799] logger.c: list_route: hop: Feb 2 10:00:44 VERBOSE[3799] logger.c: Transmitting (no NAT) to 10.193.71.156:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60;received=10.193.71.156 From: "Andrea Ballin" ;tag=b91d4746859371f2 To: Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49139 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60;received=10.193.71.156 (89) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 3: To: (27) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 4: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 5: CSeq: 49139 INVITE (18) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 8: Contact: (32) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 9: Content-Length: 0 (17) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 10: (0) Feb 2 10:00:44 DEBUG[3769] chan_sip.c: Checking device state for peer 66899 Feb 2 10:00:44 DEBUG[3769] devicestate.c: Changing state for SIP/66899 - state 2 (In use) Feb 2 10:00:44 DEBUG[3769] chan_sip.c: Checking device state for peer 66899 Feb 2 10:00:44 DEBUG[3857] app_queue.c: Device 'SIP/66899' changed to state '2' (In use) but we don't care because they're not a member of any queue. Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'Macro' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing Macro("SIP/66899-086420b0", "dialout-trunk|3|855|") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Expression result is '1' Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'GotoIf' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing GotoIf("SIP/66899-086420b0", "1?3:2)") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Goto (macro-dialout-trunk,s,3) Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'Macro' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing Macro("SIP/66899-086420b0", "user-callerid") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'DBget' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing DBget("SIP/66899-086420b0", "AMPUSER=DEVICE/66899/user") in new stack Feb 2 10:00:44 WARNING[3856] app_db.c: This application has been deprecated, please use the ${DB(family/key)} function instead. Feb 2 10:00:44 VERBOSE[3856] logger.c: -- DBget: varname=AMPUSER, family=DEVICE, key=66899/user Feb 2 10:00:44 VERBOSE[3856] logger.c: -- DBget: set variable AMPUSER to 66899 Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'DBget' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing DBget("SIP/66899-086420b0", "AMPUSERCIDNAME=AMPUSER/66899/cidname") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=66899/cidname Feb 2 10:00:44 VERBOSE[3856] logger.c: -- DBget: set variable AMPUSERCIDNAME to Andrea Ballin Feb 2 10:00:44 DEBUG[3856] pbx.c: Expression result is '0' Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'GotoIf' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing GotoIf("SIP/66899-086420b0", "0?5") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Not taking any branch Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'SetCallerID' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing SetCallerID("SIP/66899-086420b0", ""Andrea Ballin" <66899>") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'NoOp' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing NoOp("SIP/66899-086420b0", "Using CallerID "Andrea Ballin" <66899>") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'Macro' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing Macro("SIP/66899-086420b0", "record-enable|66899|OUT") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Function result is '0' Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'GotoIf' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing GotoIf("SIP/66899-086420b0", "0 > 0?2:4") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Goto (macro-record-enable,s,4) Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'AGI' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing AGI("SIP/66899-086420b0", "recordingcheck|20070202-100044|1170406844.0") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Feb 2 10:00:44 VERBOSE[3856] logger.c: recordingcheck|20070202-100044|1170406844.0: Outbound recording not enabled Feb 2 10:00:44 VERBOSE[3856] logger.c: -- AGI Script recordingcheck completed, returning 0 Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'NoOp' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing NoOp("SIP/66899-086420b0", "No recording needed") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'Macro' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing Macro("SIP/66899-086420b0", "outbound-callerid|3") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'DBget' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing DBget("SIP/66899-086420b0", "USEROUTCID=AMPUSER/66899/outboundcid") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=66899/outboundcid Feb 2 10:00:44 VERBOSE[3856] logger.c: -- DBget: set variable USEROUTCID to Feb 2 10:00:44 DEBUG[3856] pbx.c: Expression result is '1' Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'GotoIf' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing GotoIf("SIP/66899-086420b0", "1?4") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Goto (macro-outbound-callerid,s,4) Feb 2 10:00:44 DEBUG[3856] pbx.c: Expression result is '1' Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'GotoIf' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing GotoIf("SIP/66899-086420b0", "1?6") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Goto (macro-outbound-callerid,s,6) Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'NoOp' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing NoOp("SIP/66899-086420b0", "CallerID set to "Andrea Ballin" <66899>") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'SetGroup' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing SetGroup("SIP/66899-086420b0", "OUT_3") in new stack Feb 2 10:00:44 WARNING[3856] app_groupcount.c: The SetGroup application has been deprecated, please use the GROUP() function. Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'CheckGroup' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing CheckGroup("SIP/66899-086420b0", "") in new stack Feb 2 10:00:44 WARNING[3856] app_groupcount.c: The CheckGroup application has been deprecated, please use a combination of the GotoIf application and the GROUP_COUNT() function. Feb 2 10:00:44 WARNING[3856] app_groupcount.c: CheckGroup requires an argument(max[@category][|options]) Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'SetVar' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing SetVar("SIP/66899-086420b0", "DIAL_NUMBER=855") in new stack Feb 2 10:00:44 WARNING[3856] pbx.c: SetVar is deprecated, please use Set instead. Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'SetVar' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing SetVar("SIP/66899-086420b0", "DIAL_TRUNK=3") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'AGI' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing AGI("SIP/66899-086420b0", "fixlocalprefix") in new stack Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Feb 2 10:00:44 VERBOSE[3856] logger.c: fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf Feb 2 10:00:44 VERBOSE[3856] logger.c: -- AGI Script fixlocalprefix completed, returning 0 Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'SetVar' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing SetVar("SIP/66899-086420b0", "OUTNUM=855") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'Cut' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing Cut("SIP/66899-086420b0", "custom=OUT_3|:|1") in new stack Feb 2 10:00:44 WARNING[3856] app_cut.c: The application Cut is deprecated. Please use the CUT() function instead. Feb 2 10:00:44 WARNING[3856] ast_expr2.y: non-numeric argument Feb 2 10:00:44 WARNING[3856] ast_expr2.y: non-numeric argument Feb 2 10:00:44 DEBUG[3856] pbx.c: Expression result is '0' Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'GotoIf' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing GotoIf("SIP/66899-086420b0", "0?16") in new stack Feb 2 10:00:44 DEBUG[3856] pbx.c: Not taking any branch Feb 2 10:00:44 DEBUG[3856] pbx.c: Launching 'Dial' Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Executing Dial("SIP/66899-086420b0", "SIP/centralinix-in/855|60|tTrWw") in new stack Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Setting NAT on RTP to 0 Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-14. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable MACRO_DEPTH. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-13. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable custom. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-12. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable OUTNUM. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-11. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-10. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable DIAL_TRUNK. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-9. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable DIAL_NUMBER. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-8. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-7. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable GROUP. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-6. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable MACRO_PRIORITY. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable MACRO_CONTEXT. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable MACRO_EXTEN. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable ARG1. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-outbound-callerid-s-6. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-outbound-callerid-s-4. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-outbound-callerid-s-2. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable DBGETSTATUS. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable USEROUTCID. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-outbound-callerid-s-1. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-5. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable ARG2. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-record-enable-s-5. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-record-enable-s-4. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-record-enable-s-1. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-4. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-user-callerid-s-5. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-user-callerid-s-4. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-user-callerid-s-3. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable AMPUSERCIDNAME. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-user-callerid-s-2. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable AMPUSER. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-user-callerid-s-1. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-3. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-macro-dialout-trunk-s-1. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable ARG3. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable STACK-from-internal-855-1. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable SIPCALLID. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable SIPUSERAGENT. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable SIPDOMAIN. Feb 2 10:00:44 DEBUG[3856] channel.c: Not copying variable SIPURI. Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Outgoing Call for 855 Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Updating call counter for outgoing call Feb 2 10:00:44 VERBOSE[3856] logger.c: We're at 193.227.71.32 port 16320 Feb 2 10:00:44 VERBOSE[3856] logger.c: Adding codec 0x8 (alaw) to SDP Feb 2 10:00:44 VERBOSE[3856] logger.c: Adding codec 0x4 (ulaw) to SDP Feb 2 10:00:44 VERBOSE[3856] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 0: INVITE sip:855@193.227.71.21 SIP/2.0 (36) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport (64) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=as7d689b6d (62) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 3: To: (27) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 4: Contact: (34) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 5: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:00:44 GMT (35) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 11: Content-Type: application/sdp (29) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 12: Content-Length: 240 (19) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 13: (0) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: v=0 (3) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: o=root 3725 3725 IN IP4 193.227.71.32 (37) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: s=session (9) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: c=IN IP4 193.227.71.32 (22) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: m=audio 16320 RTP/AVP 8 0 101 (29) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=fmtp:101 0-16 (15) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Feb 2 10:00:44 VERBOSE[3856] logger.c: 13 headers, 11 lines Feb 2 10:00:44 VERBOSE[3856] logger.c: Reliably Transmitting (no NAT) to 193.227.71.21:5060: INVITE sip:855@193.227.71.21 SIP/2.0 Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport From: "Andrea Ballin" ;tag=as7d689b6d To: Contact: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 09:00:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 3725 3725 IN IP4 193.227.71.32 s=session c=IN IP4 193.227.71.32 t=0 0 m=audio 16320 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 0: INVITE sip:855@193.227.71.21 SIP/2.0 (36) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport (64) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=as7d689b6d (62) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 3: To: (27) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 4: Contact: (34) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 5: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:00:44 GMT (35) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 11: Content-Type: application/sdp (29) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 12: Content-Length: 240 (19) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 13: (0) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: v=0 (3) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: o=root 3725 3725 IN IP4 193.227.71.32 (37) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: s=session (9) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: c=IN IP4 193.227.71.32 (22) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: m=audio 16320 RTP/AVP 8 0 101 (29) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=fmtp:101 0-16 (15) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #15 Feb 2 10:00:44 VERBOSE[3856] logger.c: -- Called centralinix-in/855 Feb 2 10:00:44 VERBOSE[3856] logger.c: Transmitting (no NAT) to 10.193.71.156:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60;received=10.193.71.156 From: "Andrea Ballin" ;tag=b91d4746859371f2 To: ;tag=as79d79769 Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49139 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60;received=10.193.71.156 (89) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 3: To: ;tag=as79d79769 (42) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 4: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 5: CSeq: 49139 INVITE (18) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 8: Contact: (32) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 9: Content-Length: 0 (17) Feb 2 10:00:44 DEBUG[3856] chan_sip.c: Header 10: (0) Feb 2 10:00:44 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: SIP/2.0 100 Trying To: From: "Andrea Ballin" ;tag=as7d689b6d Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 102 INVITE Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport Content-Length: 0 Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 1: To: (27) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=as7d689b6d (62) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 3: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 5: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport (64) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 6: Content-Length: 0 (17) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 7: (0) Feb 2 10:00:44 VERBOSE[3799] logger.c: --- (7 headers 0 lines) --- Feb 2 10:00:44 DEBUG[3799] chan_sip.c: *** SIP TIMER: Cancelling retransmission #15 - INVITE (got response) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '552867a734a17a343714285f422ff9fd@193.227.71.32' Request 102: Found Feb 2 10:00:44 DEBUG[3799] chan_sip.c: SIP response 100 to standard invite Feb 2 10:00:44 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: SIP/2.0 180 Ringing Record-Route: Contact: User-Agent: ABS GW v5.1.0 Content-Type: application/sdp To: ;tag=6026870fe754320345f2ea8d6c88b258 From: "Andrea Ballin" ;tag=as7d689b6d Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 102 INVITE Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport Content-Length: 199 v=0 o=default 1170407279 1170407279 IN IP4 193.227.71.21 s=session c=IN IP4 193.227.71.22 t=0 0 m=audio 32680 RTP/AVP 101 8 a=rtpmap:101 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=sendrecv Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 1: Record-Route: (38) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 2: Contact: (47) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 3: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 4: Content-Type: application/sdp (29) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 5: To: ;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 6: From: "Andrea Ballin" ;tag=as7d689b6d (62) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 7: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 8: CSeq: 102 INVITE (16) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 9: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport (64) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 199 (19) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: v=0 (3) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: o=default 1170407279 1170407279 IN IP4 193.227.71.21 (52) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: s=session (9) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: c=IN IP4 193.227.71.22 (22) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: m=audio 32680 RTP/AVP 101 8 (27) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Line: a=sendrecv (10) Feb 2 10:00:44 VERBOSE[3799] logger.c: --- (11 headers 9 lines) --- Feb 2 10:00:44 DEBUG[3799] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '552867a734a17a343714285f422ff9fd@193.227.71.32' Request 102: Found Feb 2 10:00:44 DEBUG[3799] chan_sip.c: SIP response 180 to standard invite Feb 2 10:00:44 DEBUG[3769] chan_sip.c: Checking device state for peer centralinix-in Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 101 Feb 2 10:00:44 DEBUG[3769] channel.c: Avoiding initial deadlock for 'SIP/centralinix-in-b7a2a060' Feb 2 10:00:44 VERBOSE[3799] logger.c: Found RTP audio format 8 Feb 2 10:00:44 VERBOSE[3799] logger.c: Peer audio RTP is at port 193.227.71.22:32680 Feb 2 10:00:44 DEBUG[3799] chan_sip.c: Peer audio RTP is at port 193.227.71.22:32680 Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format telephone-event Feb 2 10:00:44 VERBOSE[3799] logger.c: Found description format PCMA Feb 2 10:00:44 VERBOSE[3799] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Feb 2 10:00:44 VERBOSE[3799] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Feb 2 10:00:44 VERBOSE[3856] logger.c: -- SIP/centralinix-in-b7a2a060 is ringing Feb 2 10:00:44 VERBOSE[3856] logger.c: -- SIP/centralinix-in-b7a2a060 is making progress passing it to SIP/66899-086420b0 Feb 2 10:00:44 DEBUG[3769] devicestate.c: Changing state for SIP/centralinix-in - state 6 (Ringing) Feb 2 10:00:44 DEBUG[3860] app_queue.c: Device 'SIP/centralinix-in' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Feb 2 10:00:46 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: SIP/2.0 200 OK Record-Route: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,SUBSCRIBE,OPTIONS Contact: Supported: replaces,100rel User-Agent: ABS GW v5.1.0 Content-Type: application/sdp To: ;tag=6026870fe754320345f2ea8d6c88b258 From: "Andrea Ballin" ;tag=as7d689b6d Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 102 INVITE Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport Content-Length: 199 v=0 o=default 1170407279 1170407280 IN IP4 193.227.71.21 s=session c=IN IP4 193.227.71.22 t=0 0 m=audio 32680 RTP/AVP 101 8 a=rtpmap:101 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=sendrecv Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 1: Record-Route: (38) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 2: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,SUBSCRIBE,OPTIONS (59) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 3: Contact: (47) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 4: Supported: replaces,100rel (26) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 5: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 6: Content-Type: application/sdp (29) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 7: To: ;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 8: From: "Andrea Ballin" ;tag=as7d689b6d (62) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 9: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 10: CSeq: 102 INVITE (16) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 11: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK6045842c;rport (64) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 12: Content-Length: 199 (19) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 13: (0) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: v=0 (3) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: o=default 1170407279 1170407280 IN IP4 193.227.71.21 (52) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: s=session (9) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: c=IN IP4 193.227.71.22 (22) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: m=audio 32680 RTP/AVP 101 8 (27) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Line: a=sendrecv (10) Feb 2 10:00:46 VERBOSE[3799] logger.c: --- (13 headers 9 lines) --- Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Acked pending invite 102 Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Stopping retransmission on '552867a734a17a343714285f422ff9fd@193.227.71.32' of Request 102: Match Found Feb 2 10:00:46 DEBUG[3799] chan_sip.c: SIP response 200 to standard invite Feb 2 10:00:46 VERBOSE[3799] logger.c: Found RTP audio format 101 Feb 2 10:00:46 VERBOSE[3799] logger.c: Found RTP audio format 8 Feb 2 10:00:46 VERBOSE[3799] logger.c: Peer audio RTP is at port 193.227.71.22:32680 Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Peer audio RTP is at port 193.227.71.22:32680 Feb 2 10:00:46 VERBOSE[3799] logger.c: Found description format telephone-event Feb 2 10:00:46 VERBOSE[3799] logger.c: Found description format PCMA Feb 2 10:00:46 VERBOSE[3799] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Feb 2 10:00:46 VERBOSE[3799] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: build_route: Record-Route hop: Feb 2 10:00:46 VERBOSE[3799] logger.c: list_route: hop: Feb 2 10:00:46 VERBOSE[3799] logger.c: set_destination: Parsing for address/port to send to Feb 2 10:00:46 VERBOSE[3799] logger.c: set_destination: set destination to 193.227.71.21, port 5060 Feb 2 10:00:46 VERBOSE[3799] logger.c: Transmitting (no NAT) to 193.227.71.21:5060: ACK sip:193.227.71.21:6060;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK10c92e00;rport Route: From: "Andrea Ballin" ;tag=as7d689b6d To: ;tag=6026870fe754320345f2ea8d6c88b258 Contact: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 0: ACK sip:193.227.71.21:6060;transport=tcp SIP/2.0 (48) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK10c92e00;rport (64) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 2: Route: (31) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 3: From: "Andrea Ballin" ;tag=as7d689b6d (62) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 4: To: ;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 5: Contact: (34) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 6: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 7: CSeq: 102 ACK (13) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 8: User-Agent: Asterisk PBX (24) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 9: Max-Forwards: 70 (16) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:00:46 DEBUG[3769] chan_sip.c: Checking device state for peer centralinix-in Feb 2 10:00:46 DEBUG[3769] devicestate.c: Changing state for SIP/centralinix-in - state 2 (In use) Feb 2 10:00:46 VERBOSE[3856] logger.c: -- SIP/centralinix-in-b7a2a060 answered SIP/66899-086420b0 Feb 2 10:00:46 DEBUG[3861] app_queue.c: Device 'SIP/centralinix-in' changed to state '2' (In use) but we don't care because they're not a member of any queue. Feb 2 10:00:46 DEBUG[3769] chan_sip.c: Checking device state for peer 66899 Feb 2 10:00:46 DEBUG[3856] chan_sip.c: sip_answer(SIP/66899-086420b0) Feb 2 10:00:46 DEBUG[3769] channel.c: Avoiding initial deadlock for 'SIP/66899-086420b0' Feb 2 10:00:46 VERBOSE[3856] logger.c: We're at 193.227.71.32 port 16014 Feb 2 10:00:46 VERBOSE[3856] logger.c: Adding codec 0x8 (alaw) to SDP Feb 2 10:00:46 VERBOSE[3856] logger.c: Adding codec 0x4 (ulaw) to SDP Feb 2 10:00:46 VERBOSE[3856] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Feb 2 10:00:46 VERBOSE[3856] logger.c: Reliably Transmitting (no NAT) to 10.193.71.156:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60;received=10.193.71.156 From: "Andrea Ballin" ;tag=b91d4746859371f2 To: ;tag=as79d79769 Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49139 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 3725 3725 IN IP4 193.227.71.32 s=session c=IN IP4 193.227.71.32 t=0 0 m=audio 16014 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKe0e8b1cbe6d23b60;received=10.193.71.156 (89) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 3: To: ;tag=as79d79769 (42) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 4: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 5: CSeq: 49139 INVITE (18) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 8: Contact: (32) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 9: Content-Type: application/sdp (29) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 10: Content-Length: 240 (19) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Header 11: (0) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: v=0 (3) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: o=root 3725 3725 IN IP4 193.227.71.32 (37) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: s=session (9) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: c=IN IP4 193.227.71.32 (22) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: m=audio 16014 RTP/AVP 8 0 101 (29) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: a=fmtp:101 0-16 (15) Feb 2 10:00:46 DEBUG[3856] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Feb 2 10:00:46 DEBUG[3769] channel.c: Avoiding initial deadlock for 'SIP/66899-086420b0' Feb 2 10:00:46 DEBUG[3856] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #17 Feb 2 10:00:46 DEBUG[3769] devicestate.c: Changing state for SIP/66899 - state 2 (In use) Feb 2 10:00:46 DEBUG[3769] chan_sip.c: Checking device state for peer 66899 Feb 2 10:00:46 DEBUG[3862] app_queue.c: Device 'SIP/66899' changed to state '2' (In use) but we don't care because they're not a member of any queue. Feb 2 10:00:46 VERBOSE[3799] logger.c: <-- SIP read from 10.193.71.156:5060: ACK sip:855@193.227.71.32 SIP/2.0 Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKf88f2ef8bf09a1c8 From: "Andrea Ballin" ;tag=b91d4746859371f2 To: ;tag=as79d79769 Contact: Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49139 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 0: ACK sip:855@193.227.71.32 SIP/2.0 (33) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKf88f2ef8bf09a1c8 (66) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=as79d79769 (42) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 4: Contact: (39) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 5: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 6: CSeq: 49139 ACK (15) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Grandstream GXP2000 1.1.1.14 (40) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (21) Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:00:46 VERBOSE[3799] logger.c: --- (11 headers 0 lines) --- Feb 2 10:00:46 DEBUG[3799] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Feb 2 10:00:46 DEBUG[3799] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #17 Feb 2 10:00:46 DEBUG[3799] chan_sip.c: Stopping retransmission on 'af10d5446545f719@10.193.71.156' of Response 49139: Match Found Feb 2 10:00:46 DEBUG[3856] rtp.c: Ooh, format changed from unknown to alaw Feb 2 10:00:46 DEBUG[3856] rtp.c: Forcing Marker bit, because SSRC has changed Feb 2 10:00:46 DEBUG[3856] rtp.c: Ooh, format changed from unknown to alaw Feb 2 10:00:46 DEBUG[3856] rtp.c: Forcing Marker bit, because SSRC has changed Feb 2 10:00:50 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: INVITE sip:66899@193.227.71.32 SIP/2.0 Record-Route: Contact: Supported: replaces,100rel User-Agent: ABS GW v5.1.0 Content-Type: application/sdp To: sip:66899@193.227.71.32;tag=as7d689b6d From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863093 INVITE Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bK9a2d146c943a7a1c1f9fd66d35268999809c7723920268611a9786199ab1f14a Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bKab256080413882666acd38dcc9f92d99 Max-Forwards: 69 Content-Length: 195 Session-Expires: 1800 v=0 o=default 1170407279 1170407281 IN IP4 193.227.71.21 s=abs c=IN IP4 193.227.71.22 t=0 0 m=audio 32680 RTP/AVP 101 8 a=sendonly a=rtpmap:101 telephone-event/8000 a=rtpmap:8 PCMA/8000 Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 0: INVITE sip:66899@193.227.71.32 SIP/2.0 (38) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 1: Record-Route: (38) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 2: Contact: (47) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 3: Supported: replaces,100rel (26) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 4: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 5: Content-Type: application/sdp (29) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 6: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 7: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 8: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 9: CSeq: 932863093 INVITE (22) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 10: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bK9a2d146c943a7a1c1f9fd66d35268999809c7723920268611a9786199ab1f14a (109) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 11: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bKab256080413882666acd38dcc9f92d99 (82) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 12: Max-Forwards: 69 (16) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 13: Content-Length: 195 (19) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 14: Session-Expires: 1800 (21) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 15: (0) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: v=0 (3) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: o=default 1170407279 1170407281 IN IP4 193.227.71.21 (52) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: s=abs (5) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: c=IN IP4 193.227.71.22 (22) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: m=audio 32680 RTP/AVP 101 8 (27) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=sendonly (10) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:50 VERBOSE[3799] logger.c: --- (15 headers 9 lines) --- Feb 2 10:00:50 DEBUG[3799] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Begin: parsing SIP "Supported: replaces,100rel" Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Found SIP option: -replaces- Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Matched SIP option: replaces Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Found SIP option: -100rel- Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Matched SIP option: 100rel Feb 2 10:00:50 DEBUG[3799] chan_sip.c: * SIP extension value: 3 for call 552867a734a17a343714285f422ff9fd@193.227.71.32 Feb 2 10:00:50 VERBOSE[3799] logger.c: Using INVITE request as basis request - 552867a734a17a343714285f422ff9fd@193.227.71.32 Feb 2 10:00:50 VERBOSE[3799] logger.c: Sending to 193.227.71.21 : 5060 (non-NAT) Feb 2 10:00:50 VERBOSE[3799] logger.c: Found RTP audio format 101 Feb 2 10:00:50 VERBOSE[3799] logger.c: Found RTP audio format 8 Feb 2 10:00:50 VERBOSE[3799] logger.c: Peer audio RTP is at port 193.227.71.22:32680 Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Peer audio RTP is at port 193.227.71.22:32680 Feb 2 10:00:50 VERBOSE[3799] logger.c: Found description format telephone-event Feb 2 10:00:50 VERBOSE[3799] logger.c: Found description format PCMA Feb 2 10:00:50 VERBOSE[3799] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Feb 2 10:00:50 VERBOSE[3799] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Feb 2 10:00:50 DEBUG[3799] channel.c: Set channel SIP/66899-086420b0 to write format slin Feb 2 10:00:50 VERBOSE[3799] logger.c: -- Started music on hold, class 'default', on channel 'SIP/66899-086420b0' Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Got a SIP re-invite for call 552867a734a17a343714285f422ff9fd@193.227.71.32 Feb 2 10:00:50 VERBOSE[3799] logger.c: We're at 193.227.71.32 port 16320 Feb 2 10:00:50 VERBOSE[3799] logger.c: Adding codec 0x8 (alaw) to SDP Feb 2 10:00:50 VERBOSE[3799] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Feb 2 10:00:50 VERBOSE[3799] logger.c: Reliably Transmitting (no NAT) to 193.227.71.21:5060: SIP/2.0 200 OK Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bK9a2d146c943a7a1c1f9fd66d35268999809c7723920268611a9786199ab1f14a;received=193.227.71.21 Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bKab256080413882666acd38dcc9f92d99 Record-Route: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 To: sip:66899@193.227.71.32;tag=as7d689b6d Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863093 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 3725 3726 IN IP4 193.227.71.32 s=session c=IN IP4 193.227.71.32 t=0 0 m=audio 16320 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bK9a2d146c943a7a1c1f9fd66d35268999809c7723920268611a9786199ab1f14a;received=193.227.71.21 (132) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 2: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bKab256080413882666acd38dcc9f92d99 (82) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 3: Record-Route: (38) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 4: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 5: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 6: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 7: CSeq: 932863093 INVITE (22) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 8: User-Agent: Asterisk PBX (24) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 10: Contact: (34) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 11: Content-Type: application/sdp (29) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 12: Content-Length: 216 (19) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Header 13: (0) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: v=0 (3) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: o=root 3725 3726 IN IP4 193.227.71.32 (37) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: s=session (9) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: c=IN IP4 193.227.71.32 (22) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: t=0 0 (5) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: m=audio 16320 RTP/AVP 8 101 (27) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=fmtp:101 0-16 (15) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Feb 2 10:00:50 DEBUG[3799] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #18 Feb 2 10:00:51 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: ACK sip:66899@193.227.71.32 SIP/2.0 Contact: User-Agent: ABS GW v5.1.0 To: sip:66899@193.227.71.32;tag=as7d689b6d From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863093 ACK Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bK9a2d146c943a7a1c1f9fd66d35268999ec647805151f252c3f0b7712c94ca1db Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bKcc51583df529bf12eff264477138a832 Max-Forwards: 69 Content-Length: 0 Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 0: ACK sip:66899@193.227.71.32 SIP/2.0 (35) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 1: Contact: (47) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 2: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 3: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 4: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 6: CSeq: 932863093 ACK (19) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 7: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bK9a2d146c943a7a1c1f9fd66d35268999ec647805151f252c3f0b7712c94ca1db (109) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 8: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bKcc51583df529bf12eff264477138a832 (82) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 9: Max-Forwards: 69 (16) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:00:51 VERBOSE[3799] logger.c: --- (11 headers 0 lines) --- Feb 2 10:00:51 DEBUG[3799] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Feb 2 10:00:51 DEBUG[3799] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 Feb 2 10:00:51 DEBUG[3799] chan_sip.c: Stopping retransmission on '552867a734a17a343714285f422ff9fd@193.227.71.32' of Response 932863093: Match Found Feb 2 10:00:51 DEBUG[3856] rtp.c: Got RTCP report of 88 bytes Feb 2 10:00:56 DEBUG[3856] rtp.c: Got RTCP report of 88 bytes Feb 2 10:00:57 VERBOSE[3799] logger.c: <-- SIP read from 10.97.1.253:5060: OPTIONS sip:193.227.71.32 SIP/2.0 Via: SIP/2.0/UDP 10.97.1.253:5060;branch=z9hG4bK777f3fd0;rport From: "Unknown" ;tag=as62fb8793 To: Contact: Call-ID: 0cbca9962b7f186b45c5541950625ee6@10.97.1.253 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 09:00:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 0: OPTIONS sip:193.227.71.32 SIP/2.0 (33) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.97.1.253:5060;branch=z9hG4bK777f3fd0;rport (62) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 2: From: "Unknown" ;tag=as62fb8793 (56) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 3: To: (23) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 4: Contact: (34) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 0cbca9962b7f186b45c5541950625ee6@10.97.1.253 (53) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:00:53 GMT (35) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 11: Content-Length: 0 (17) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:00:57 VERBOSE[3799] logger.c: --- (12 headers 0 lines) --- Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Allocating new SIP dialog for 0cbca9962b7f186b45c5541950625ee6@10.97.1.253 - OPTIONS (No RTP) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS Feb 2 10:00:57 VERBOSE[3799] logger.c: Looking for s in from-internal (domain 193.227.71.32) Feb 2 10:00:57 VERBOSE[3799] logger.c: Transmitting (no NAT) to 10.97.1.253:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.97.1.253:5060;branch=z9hG4bK777f3fd0;received=10.97.1.253;rport=5060 From: "Unknown" ;tag=as62fb8793 To: ;tag=as1d777176 Call-ID: 0cbca9962b7f186b45c5541950625ee6@10.97.1.253 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.97.1.253:5060;branch=z9hG4bK777f3fd0;received=10.97.1.253;rport=5060 (88) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 2: From: "Unknown" ;tag=as62fb8793 (56) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=as1d777176 (38) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 4: Call-ID: 0cbca9962b7f186b45c5541950625ee6@10.97.1.253 (53) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 8: Contact: (28) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 9: Accept: application/sdp (23) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:00:57 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:00:57 VERBOSE[3799] logger.c: Destroying call '0cbca9962b7f186b45c5541950625ee6@10.97.1.253' Feb 2 10:00:57 DEBUG[3799] chan_sip.c: ---------- SIP HISTORY for '0cbca9962b7f186b45c5541950625ee6@10.97.1.253' Feb 2 10:00:57 DEBUG[3799] chan_sip.c: * SIP Call Feb 2 10:00:57 DEBUG[3799] chan_sip.c: 1. Rx OPTIONS / 102 OPTIONS /sip:193.227.71.32 Feb 2 10:00:57 DEBUG[3799] chan_sip.c: 2. TxResp SIP/2.0 / 102 OPTIONS Feb 2 10:00:57 DEBUG[3799] chan_sip.c: ---------- END SIP HISTORY for '0cbca9962b7f186b45c5541950625ee6@10.97.1.253' Feb 2 10:00:59 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: REFER sip:66899@193.227.71.32 SIP/2.0 Contact: Supported: replaces,100rel User-Agent: ABS GW v5.1.0 Refer-To: Referred-By: sip:855@193.227.71.21 To: sip:66899@193.227.71.32;tag=as7d689b6d From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863094 REFER Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKd45a39ff30ebfabb5cadae58bbdedc3c656fa70eec1f6c2651814f39e8cc1351 Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK4d732a797b681be3201fc4c1627c09f4 Max-Forwards: 69 Content-Length: 0 Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 0: REFER sip:66899@193.227.71.32 SIP/2.0 (37) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 1: Contact: (47) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 2: Supported: replaces,100rel (26) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 3: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 4: Refer-To: (172) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 5: Referred-By: sip:855@193.227.71.21 (35) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 6: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 7: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 8: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 9: CSeq: 932863094 REFER (21) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 10: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKd45a39ff30ebfabb5cadae58bbdedc3c656fa70eec1f6c2651814f39e8cc1351 (109) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 11: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK4d732a797b681be3201fc4c1627c09f4 (82) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 12: Max-Forwards: 69 (16) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 13: Content-Length: 0 (17) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 14: (0) Feb 2 10:00:59 VERBOSE[3799] logger.c: --- (14 headers 0 lines) --- Feb 2 10:00:59 DEBUG[3799] chan_sip.c: **** Received REFER (9) - Command in SIP REFER Feb 2 10:00:59 DEBUG[3799] chan_sip.c: SIP call transfer received for call 552867a734a17a343714285f422ff9fd@193.227.71.32 (REFER)! Feb 2 10:00:59 VERBOSE[3799] logger.c: Transfer to PAVANELLO%20816 in from-internal Feb 2 10:00:59 VERBOSE[3799] logger.c: Transfer from 855 in from-internal Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Assigning Replace-Call-ID Info 552867a734a17a343714285f422ff9fd@193.227.71.32 to REPLACE_CALL_ID Feb 2 10:00:59 NOTICE[3799] chan_sip.c: Supervised transfer attempted to transfer into same call id (552867a734a17a343714285f422ff9fd@193.227.71.32 == 552867a734a17a343714285f422ff9fd@193.227.71.32)! Feb 2 10:00:59 VERBOSE[3799] logger.c: Transmitting (no NAT) to 193.227.71.21:5060: SIP/2.0 603 Declined Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKd45a39ff30ebfabb5cadae58bbdedc3c656fa70eec1f6c2651814f39e8cc1351;received=193.227.71.21 Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK4d732a797b681be3201fc4c1627c09f4 From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 To: sip:66899@193.227.71.32;tag=as7d689b6d Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863094 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 603 Declined (20) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKd45a39ff30ebfabb5cadae58bbdedc3c656fa70eec1f6c2651814f39e8cc1351;received=193.227.71.21 (132) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 2: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK4d732a797b681be3201fc4c1627c09f4 (82) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 3: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 4: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 6: CSeq: 932863094 REFER (21) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 9: Contact: (34) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 11: X-Asterisk-HangupCause: Normal Clearing (39) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:00:59 DEBUG[3799] chan_sip.c: SIP message could not be handled, bad request: 552867a734a17a343714285f422ff9fd@193.227.71.32 Feb 2 10:00:59 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:00:59 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 0: OPTIONS sip:10.97.1.254 SIP/2.0 (31) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK04d63375;rport (64) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 2: From: "Centralinix" ;tag=as7a5f744c (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 3: To: (21) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 4: Contact: (40) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 42d008f8178eebc23722fc7c7b23ecea@193.227.71.32 (55) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:01:00 GMT (35) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 11: Content-Length: 0 (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:01:00 VERBOSE[3799] logger.c: 12 headers, 0 lines Feb 2 10:01:00 VERBOSE[3799] logger.c: Reliably Transmitting (no NAT) to 10.97.1.254:5060: OPTIONS sip:10.97.1.254 SIP/2.0 Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK04d63375;rport From: "Centralinix" ;tag=as7a5f744c To: Contact: Call-ID: 42d008f8178eebc23722fc7c7b23ecea@193.227.71.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 09:01:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 0: OPTIONS sip:10.97.1.254 SIP/2.0 (31) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK04d63375;rport (64) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 2: From: "Centralinix" ;tag=as7a5f744c (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 3: To: (21) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 4: Contact: (40) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 42d008f8178eebc23722fc7c7b23ecea@193.227.71.32 (55) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:01:00 GMT (35) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 11: Content-Length: 0 (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #19 Feb 2 10:01:00 VERBOSE[3799] logger.c: <-- SIP read from 10.97.1.254:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK04d63375;received=193.227.71.32;rport=5060 From: "Centralinix" ;tag=as7a5f744c To: ;tag=as2e46b940 Call-ID: 42d008f8178eebc23722fc7c7b23ecea@193.227.71.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK04d63375;received=193.227.71.32;rport=5060 (92) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 2: From: "Centralinix" ;tag=as7a5f744c (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=as2e46b940 (36) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 4: Call-ID: 42d008f8178eebc23722fc7c7b23ecea@193.227.71.32 (55) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 8: Contact: (26) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 9: Accept: application/sdp (23) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:01:00 VERBOSE[3799] logger.c: --- (11 headers 0 lines) --- Feb 2 10:01:00 DEBUG[3799] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #19 Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Stopping retransmission on '42d008f8178eebc23722fc7c7b23ecea@193.227.71.32' of Request 102: Match Found Feb 2 10:01:00 VERBOSE[3799] logger.c: Destroying call '42d008f8178eebc23722fc7c7b23ecea@193.227.71.32' Feb 2 10:01:00 DEBUG[3799] chan_sip.c: ---------- SIP HISTORY for '42d008f8178eebc23722fc7c7b23ecea@193.227.71.32' Feb 2 10:01:00 DEBUG[3799] chan_sip.c: * SIP Call Feb 2 10:01:00 DEBUG[3799] chan_sip.c: 1. TxReqRel OPTIONS / 102 OPTIONS Feb 2 10:01:00 DEBUG[3799] chan_sip.c: 2. Rx SIP/2.0 / 102 OPTIONS /200 OK Feb 2 10:01:00 DEBUG[3799] chan_sip.c: ---------- END SIP HISTORY for '42d008f8178eebc23722fc7c7b23ecea@193.227.71.32' Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 0: OPTIONS sip:193.227.71.21 SIP/2.0 (33) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK0ea79734;rport (64) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 2: From: "Centralinix" ;tag=as01f521bf (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 3: To: (23) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 4: Contact: (40) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 0510222077ae1f7453ff473204a49eed@193.227.71.32 (55) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:01:00 GMT (35) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 11: Content-Length: 0 (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:01:00 VERBOSE[3799] logger.c: 12 headers, 0 lines Feb 2 10:01:00 VERBOSE[3799] logger.c: Reliably Transmitting (no NAT) to 193.227.71.21:5060: OPTIONS sip:193.227.71.21 SIP/2.0 Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK0ea79734;rport From: "Centralinix" ;tag=as01f521bf To: Contact: Call-ID: 0510222077ae1f7453ff473204a49eed@193.227.71.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 09:01:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 0: OPTIONS sip:193.227.71.21 SIP/2.0 (33) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK0ea79734;rport (64) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 2: From: "Centralinix" ;tag=as01f521bf (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 3: To: (23) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 4: Contact: (40) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 0510222077ae1f7453ff473204a49eed@193.227.71.32 (55) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 8: Max-Forwards: 70 (16) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 9: Date: Fri, 02 Feb 2007 09:01:00 GMT (35) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 11: Content-Length: 0 (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #22 Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: SIP/2.0 480 Temporarily not available Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,SUBSCRIBE,OPTIONS User-Agent: ABS GW v5.1.0 To: ;tag=90e6548582ca9f89c93de146a6cc6ffa From: "Centralinix" ;tag=as01f521bf Call-ID: 0510222077ae1f7453ff473204a49eed@193.227.71.32 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK0ea79734;rport Content-Length: 0 Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 480 Temporarily not available (37) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 1: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,SUBSCRIBE,OPTIONS (59) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 2: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=90e6548582ca9f89c93de146a6cc6ffa (60) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 4: From: "Centralinix" ;tag=as01f521bf (66) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 0510222077ae1f7453ff473204a49eed@193.227.71.32 (55) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 7: Via: SIP/2.0/UDP 193.227.71.32:5060;branch=z9hG4bK0ea79734;rport (64) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 8: Content-Length: 0 (17) Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Header 9: (0) Feb 2 10:01:00 VERBOSE[3799] logger.c: --- (9 headers 0 lines) --- Feb 2 10:01:00 DEBUG[3799] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22 Feb 2 10:01:00 DEBUG[3799] chan_sip.c: Stopping retransmission on '0510222077ae1f7453ff473204a49eed@193.227.71.32' of Request 102: Match Found Feb 2 10:01:00 VERBOSE[3799] logger.c: Destroying call '0510222077ae1f7453ff473204a49eed@193.227.71.32' Feb 2 10:01:00 DEBUG[3799] chan_sip.c: ---------- SIP HISTORY for '0510222077ae1f7453ff473204a49eed@193.227.71.32' Feb 2 10:01:00 DEBUG[3799] chan_sip.c: * SIP Call Feb 2 10:01:00 DEBUG[3799] chan_sip.c: 1. TxReqRel OPTIONS / 102 OPTIONS Feb 2 10:01:00 DEBUG[3799] chan_sip.c: 2. Rx SIP/2.0 / 102 OPTIONS /480 Temporarily not available Feb 2 10:01:00 DEBUG[3799] chan_sip.c: ---------- END SIP HISTORY for '0510222077ae1f7453ff473204a49eed@193.227.71.32' Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:00 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:01 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:02 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 VERBOSE[3799] logger.c: <-- SIP read from 193.227.71.21:65471: BYE sip:66899@193.227.71.32 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:66899@193.227.71.32;tag=as7d689b6d From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863095 BYE Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKea52ec3ed9f7a17fd282b37ee0a73c9274b2c9c43ee3eee473df413a2ccb56c9 Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK2b770d6bf34f38f9141caf77030adb3c Max-Forwards: 69 Content-Length: 0 Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 0: BYE sip:66899@193.227.71.32 SIP/2.0 (35) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 1: Supported: replaces,100rel (26) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 2: User-Agent: ABS GW v5.1.0 (26) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 3: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 4: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 6: CSeq: 932863095 BYE (19) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 7: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKea52ec3ed9f7a17fd282b37ee0a73c9274b2c9c43ee3eee473df413a2ccb56c9 (109) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 8: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK2b770d6bf34f38f9141caf77030adb3c (82) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 9: Max-Forwards: 69 (16) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:01:03 VERBOSE[3799] logger.c: --- (11 headers 0 lines) --- Feb 2 10:01:03 DEBUG[3799] chan_sip.c: **** Received BYE (8) - Command in SIP BYE Feb 2 10:01:03 VERBOSE[3799] logger.c: Sending to 193.227.71.21 : 5060 (non-NAT) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Received bye, issuing owner hangup .Feb 2 10:01:03 VERBOSE[3799] logger.c: Transmitting (no NAT) to 193.227.71.21:5060: SIP/2.0 200 OK Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKea52ec3ed9f7a17fd282b37ee0a73c9274b2c9c43ee3eee473df413a2ccb56c9;received=193.227.71.21 Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK2b770d6bf34f38f9141caf77030adb3c From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 To: sip:66899@193.227.71.32;tag=as7d689b6d Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 CSeq: 932863095 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/udp 193.227.71.21;branch=z9hG4bKea52ec3ed9f7a17fd282b37ee0a73c9274b2c9c43ee3eee473df413a2ccb56c9;received=193.227.71.21 (132) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 2: Via: SIP/2.0/udp 193.227.71.21:6060;branch=z9hG4bK2b770d6bf34f38f9141caf77030adb3c (82) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 3: From: sip:855@193.227.71.21;tag=6026870fe754320345f2ea8d6c88b258 (64) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 4: To: sip:66899@193.227.71.32;tag=as7d689b6d (42) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 5: Call-ID: 552867a734a17a343714285f422ff9fd@193.227.71.32 (55) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 6: CSeq: 932863095 BYE (19) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 9: Contact: (34) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 10: Content-Length: 0 (17) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 11: X-Asterisk-HangupCause: Normal Clearing (39) Feb 2 10:01:03 DEBUG[3799] chan_sip.c: Header 12: (0) Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:03 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:04 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:05 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:06 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 VERBOSE[3799] logger.c: <-- SIP read from 10.193.71.156:5060: BYE sip:855@193.227.71.32 SIP/2.0 Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKf44b9a6ee5aaa49d From: "Andrea Ballin" ;tag=b91d4746859371f2 To: ;tag=as79d79769 Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49140 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 0: BYE sip:855@193.227.71.32 SIP/2.0 (33) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKf44b9a6ee5aaa49d (66) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=as79d79769 (42) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 4: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 5: CSeq: 49140 BYE (15) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 7: Max-Forwards: 70 (16) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 9: Content-Length: 0 (21) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 10: (0) Feb 2 10:01:07 VERBOSE[3799] logger.c: --- (10 headers 0 lines) --- Feb 2 10:01:07 DEBUG[3799] chan_sip.c: **** Received BYE (8) - Command in SIP BYE Feb 2 10:01:07 VERBOSE[3799] logger.c: Sending to 10.193.71.156 : 5060 (non-NAT) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Received bye, issuing owner hangup .Feb 2 10:01:07 VERBOSE[3799] logger.c: Transmitting (no NAT) to 10.193.71.156:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKf44b9a6ee5aaa49d;received=10.193.71.156 From: "Andrea Ballin" ;tag=b91d4746859371f2 To: ;tag=as79d79769 Call-ID: af10d5446545f719@10.193.71.156 CSeq: 49140 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.193.71.156:5060;branch=z9hG4bKf44b9a6ee5aaa49d;received=10.193.71.156 (89) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 2: From: "Andrea Ballin" ;tag=b91d4746859371f2 (68) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 3: To: ;tag=as79d79769 (42) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 4: Call-ID: af10d5446545f719@10.193.71.156 (39) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 5: CSeq: 49140 BYE (15) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 8: Contact: (32) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 9: Content-Length: 0 (17) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) Feb 2 10:01:07 DEBUG[3799] chan_sip.c: Header 11: (0) Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:07 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:08 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:08 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:08 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe Feb 2 10:01:08 DEBUG[3780] res_musiconhold.c: Only wrote -1 of 1600 bytes to pipe