bjmg*CLI> <-- SIP read from 201.32.22.1:5060: INVITE sip:013606118682@bjmg.pbx.hahacom.com:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK7057.d532e314.0 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f180d0000690800000973 Content-Length: 337 Contact: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 Content-Type: application/sdp CSeq: 2 INVITE From: "ppyy";tag=1349071764480 Max-Forwards: 69 To: User-Agent: SJphone/1.60.289a (SJ Labs) Babale-hint: NAThelper Babale-hint: route3 Babale-hint: route5 Babale-hint: route5-donat v=0 o=- 3359217293 3359217293 IN IP4 192.168.0.88 s=SJphone c=IN IP4 192.168.0.88 t=0 0 a=direction:active m=audio 49264 RTP/AVP 97 98 0 8 3 101 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:013606118682@bjmg.pbx.hahacom.com:5060 SIP/2.0 (57) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 1: Record-Route: (58) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 2: Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK7057.d532e314.0 (60) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f180d0000690800000973 (119) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 4: Content-Length: 337 (19) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 5: Contact: (39) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 6: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 (58) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 7: Content-Type: application/sdp (29) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 8: CSeq: 2 INVITE (14) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 9: From: "ppyy";tag=1349071764480 (54) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 10: Max-Forwards: 69 (16) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 11: To: (36) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 12: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 13: Babale-hint: NAThelper (22) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 14: Babale-hint: route3 (19) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 15: Babale-hint: route5 (19) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 16: Babale-hint: route5-donat (25) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 17: (0) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: o=- 3359217293 3359217293 IN IP4 192.168.0.88 (45) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: s=SJphone (9) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.88 (21) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=direction:active (18) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: m=audio 49264 RTP/AVP 97 98 0 8 3 101 (37) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=rtpmap:98 iLBC/8000 (21) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=fmtp:98 mode=20 (17) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-11,16 (18) --- (17 headers 15 lines)--- Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 - INVITE (With RTP) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 Sending to 201.32.22.1 : 5060 (NAT) Found peer 'ser' Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:7242 check_user_full: Setting NAT on RTP to 524288 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.88:49264 Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 192.168.0.88:49264 Found description format iLBC Found description format iLBC Found description format PCMU Found description format PCMA Found description format GSM Found description format telephone-event Capabilities: us - 0x406 (gsm|ulaw|ilbc), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x406 (gsm|ulaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:2209 update_call_counter: Updating call counter for incoming call Looking for 013606118682 in dialout (domain bjmg.pbx.hahacom.com) Jun 14 03:55:41 DEBUG[23362]: chan_sip.c:6109 build_route: build_route: Record-Route hop: list_route: hop: Transmitting (NAT) to 201.32.22.1:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK7057.d532e314.0;received=201.32.22.1 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f180d0000690800000973 From: "ppyy";tag=1349071764480 To: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 2 INVITE User-Agent: Babale PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 14 03:55:41 DEBUG[23349]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 201.32.22.1 Jun 14 03:55:41 DEBUG[23389]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '8002' Jun 14 03:55:41 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for SIP/201.32.22.1 - state 2 (In use) Jun 14 03:55:41 DEBUG[23389]: pbx.c:1677 pbx_extension_helper: Launching 'DeadAGI' -- Executing DeadAGI("SIP/201.32.22.1-09f232a8", "dialout.agi| 8002|013606118682") in new stack Jun 14 03:55:41 DEBUG[23390]: app_queue.c:523 changethread: Device 'SIP/201.32.22.1' changed to state '2' (In use) but we don't care because they're not a member of any queue. -- Launched AGI Script /var/lib/asterisk/agi-bin/dialout.agi -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/17909013606118682|120) Jun 14 03:55:41 DEBUG[23389]: chan_zap.c:7579 zt_request: Using channel 1 Jun 14 03:55:41 DEBUG[23389]: channel.c:2836 ast_channel_inherit_variables: Not copying variable STACK-dialout-013606118682-1. Jun 14 03:55:41 DEBUG[23389]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jun 14 03:55:41 DEBUG[23389]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jun 14 03:55:41 DEBUG[23389]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jun 14 03:55:41 DEBUG[23389]: channel.c:2836 ast_channel_inherit_variables: Not copying variable SIPURI. Jun 14 03:55:41 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 2 (In use) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/17909013606118682 Jun 14 03:55:41 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 2 (In use) Jun 14 03:55:41 DEBUG[23389]: channel.c:2363 set_format: Set channel Zap/1-1 to read format slin Jun 14 03:55:41 DEBUG[23389]: channel.c:2363 set_format: Set channel SIP/201.32.22.1-09f232a8 to write format slin Jun 14 03:55:41 DEBUG[23389]: channel.c:2363 set_format: Set channel SIP/201.32.22.1-09f232a8 to read format slin Jun 14 03:55:41 DEBUG[23389]: channel.c:2363 set_format: Set channel Zap/1-1 to write format slin Jun 14 03:55:41 DEBUG[23392]: app_queue.c:523 changethread: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 14 03:55:41 DEBUG[23393]: app_queue.c:523 changethread: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. We're at 201.32.22.2 port 16294 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 201.32.22.1:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK7057.d532e314.0;received=201.32.22.1 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f180d0000690800000973 From: "ppyy";tag=1349071764480 To: ;tag=as6bb02b65 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 2 INVITE User-Agent: Babale PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 23344 23344 IN IP4 201.32.22.2 s=session c=IN IP4 201.32.22.2 t=0 0 m=audio 16294 RTP/AVP 0 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 14 03:55:41 DEBUG[23389]: rtp.c:1352 ast_rtp_write: Ooh, format changed from unknown to ulaw Jun 14 03:55:42 DEBUG[23389]: rtp.c:479 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 218.93.158.125:49264 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 2 Jun 14 03:55:46 DEBUG[23359]: chan_zap.c:8709 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1 -- Zap/1-1 is proceeding passing it to SIP/201.32.22.1-09f232a8 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 2 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 2 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:57 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 2 Jun 14 03:55:58 DEBUG[23359]: chan_zap.c:1405 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 is ringing Transmitting (NAT) to 201.32.22.1:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK7057.d532e314.0;received=201.32.22.1 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f180d0000690800000973 From: "ppyy";tag=1349071764480 To: ;tag=as6bb02b65 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 2 INVITE User-Agent: Babale PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 14 03:55:58 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 6 (Ringing) Jun 14 03:55:58 DEBUG[23408]: app_queue.c:523 changethread: Device 'Zap/1' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Jun 14 03:55:58 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Jun 14 03:55:58 DEBUG[23389]: chan_zap.c:4711 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 2 -- Channel 0/1, span 1 got hangup request -- Zap/1-1 is busy Jun 14 03:56:03 DEBUG[23389]: channel.c:1336 ast_hangup: Hanging up channel 'Zap/1-1' Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:2311 zt_hangup: zt_hangup(Zap/1-1) Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:2863 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:2344 zt_hangup: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:2493 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:2784 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:1374 update_conf: Updated conferencing on 1, with 0 conference users Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:2859 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Jun 14 03:56:03 DEBUG[23389]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:1/0/0) Jun 14 03:56:03 DEBUG[23389]: app_dial.c:1628 dial_exec_full: Exiting with DIALSTATUS=BUSY. Jun 14 03:56:03 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 0 (Unknown) Jun 14 03:56:03 DEBUG[23413]: app_queue.c:523 changethread: Device 'Zap/1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Jun 14 03:56:03 DEBUG[23389]: chan_sip.c:2540 sip_answer: sip_answer(SIP/201.32.22.1-09f232a8) Jun 14 03:56:03 DEBUG[23349]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 201.32.22.1 We're at 201.32.22.2 port 16294 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 201.32.22.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK7057.d532e314.0;received=201.32.22.1 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f180d0000690800000973 Record-Route: From: "ppyy";tag=1349071764480 To: ;tag=as6bb02b65 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 2 INVITE User-Agent: Babale PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 23344 23345 IN IP4 201.32.22.2 s=session c=IN IP4 201.32.22.2 t=0 0 m=audio 16294 RTP/AVP 0 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 14 03:56:03 DEBUG[23349]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/201.32.22.1-09f232a8' Jun 14 03:56:03 DEBUG[23389]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1 Jun 14 03:56:03 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for SIP/201.32.22.1 - state 2 (In use) Jun 14 03:56:03 DEBUG[23414]: app_queue.c:523 changethread: Device 'SIP/201.32.22.1' changed to state '2' (In use) but we don't care because they're not a member of any queue. -- AGI Script dialout.agi completed, returning 0 Jun 14 03:56:03 DEBUG[23389]: pbx.c:1677 pbx_extension_helper: Launching 'Hangup' -- Executing Hangup("SIP/201.32.22.1-09f232a8", "") in new stack Jun 14 03:56:03 DEBUG[23389]: pbx.c:2316 __ast_pbx_run: Spawn extension (dialout,013606118682,2) exited non-zero on 'SIP/201.32.22.1-09f232a8' Jun 14 03:56:03 DEBUG[23389]: channel.c:1336 ast_hangup: Hanging up channel 'SIP/201.32.22.1-09f232a8' Jun 14 03:56:03 DEBUG[23389]: chan_sip.c:2418 sip_hangup: Hangup call SIP/201.32.22.1-09f232a8, SIP callid A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88) Jun 14 03:56:03 DEBUG[23389]: chan_sip.c:2426 sip_hangup: update_call_counter() - decrement call limit counter Jun 14 03:56:03 DEBUG[23389]: chan_sip.c:2209 update_call_counter: Updating call counter for incoming call Jun 14 03:56:03 DEBUG[23349]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 201.32.22.1 Jun 14 03:56:03 DEBUG[23349]: devicestate.c:187 do_state_change: Changing state for SIP/201.32.22.1 - state 1 (Not in use) Jun 14 03:56:03 DEBUG[23415]: app_queue.c:523 changethread: Device 'SIP/201.32.22.1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. bjmg*CLI> <-- SIP read from 201.32.22.1:5060: ACK sip:013606118682@201.32.22.2 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 201.32.22.1;branch=0 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f1823000032e300000978 Content-Length: 0 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 2 ACK From: "ppyy";tag=1349071764480 Max-Forwards: 69 To: ;tag=as6bb02b65 User-Agent: SJphone/1.60.289a (SJ Labs) Babale-hint: NAThelper Babale-hint: loose_route Babale-hint: route4 Babale-hint: route4-donat Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 0: ACK sip:013606118682@201.32.22.2 SIP/2.0 (42) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 1: Record-Route: (58) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 2: Via: SIP/2.0/UDP 201.32.22.1;branch=0 (39) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f1823000032e300000978 (119) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 4: Content-Length: 0 (17) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 5: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 (58) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 6: CSeq: 2 ACK (11) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 7: From: "ppyy";tag=1349071764480 (54) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 69 (16) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 9: To: ;tag=as6bb02b65 (51) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 11: Babale-hint: NAThelper (22) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 12: Babale-hint: loose_route (24) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 13: Babale-hint: route4 (19) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 14: Babale-hint: route4-donat (25) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 15: (0) --- (15 headers 0 lines)--- Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1 Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88' of Response 2: Match Found set_destination: Parsing for address/port to send to set_destination: set destination to 201.32.22.1, port 5060 Reliably Transmitting (NAT) to 201.32.22.1:5060: CANCEL :013606118682@201.32.22.2 SIP/2.0 Via: SIP/2.0/UDP 201.32.22.2:5060;branch=z9hG4bK132b6977;rport Route: From: To: "ppyy";tag=1349071764480 Contact: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 101 CANCEL User-Agent: Babale PBX Max-Forwards: 70 Content-Length: 0 --- Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #2 Scheduling destruction of call 'A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88' in 32000 ms bjmg*CLI> <-- SIP read from 201.32.22.1:5060: SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL) Via: SIP/2.0/UDP 201.32.22.2:5060;branch=z9hG4bK132b6977;rport=5060 From: To: "ppyy";tag=1349071764480 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 101 CANCEL Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 201.32.22.1:5060 "Noisy feedback tells: pid=15800 req_src_ip=201.32.22.2 req_src_port=5060 in_uri=:013606118682@201.32.22.2 out_uri=:013606118682@201.32.22.2 via_cnt==1" Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL) (69) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 201.32.22.2:5060;branch=z9hG4bK132b6977;rport=5060 (69) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 2: From: (38) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 3: To: "ppyy";tag=1349071764480 (52) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 4: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 (58) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 5: CSeq: 101 CANCEL (16) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 6: Server: OpenSer (1.0.1 (i386/linux)) (36) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 7: Content-Length: 0 (17) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 8: Warning: 392 201.32.22.1:5060 "Noisy feedback tells: pid=15800 req_src_ip=201.32.22.2 req_src_port=5060 in_uri=:013606118682@201.32.22.2 out_uri=:013606118682@201.32.22.2 via_cnt==1" (191) Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 Jun 14 03:56:03 DEBUG[23362]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88' of Request 101: Match Found -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from 201.32.22.1 Destroying call 'A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88' bjmg*CLI> <-- SIP read from 201.32.22.1:5060: BYE sip:013606118682@201.32.22.2 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK8057.49d325d1.0 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f182700001c5a00000979 Content-Length: 0 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 3 BYE From: "ppyy";tag=1349071764480 Max-Forwards: 69 To: ;tag=as6bb02b65 User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="8002",realm="bj.pbx.hahacom.com",nonce="448f19699128c00b82244a6d01585e9ad7b01924",uri="sip:013606118682@201.32.22.1",response="d821064c01019b273d7cfc64ac8c52ba",cnonce="13490736624084",qop="auth",nc="00000001" Babale-hint: NAThelper Babale-hint: loose_route Babale-hint: route4 Babale-hint: route4-donat Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 0: BYE sip:013606118682@201.32.22.2 SIP/2.0 (42) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 1: Record-Route: (58) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 2: Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK8057.49d325d1.0 (60) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 3: Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f182700001c5a00000979 (119) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 4: Content-Length: 0 (17) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 5: Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 (58) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 6: CSeq: 3 BYE (11) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 7: From: "ppyy";tag=1349071764480 (54) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 69 (16) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 9: To: ;tag=as6bb02b65 (51) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 10: User-Agent: SJphone/1.60.289a (SJ Labs) (39) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 11: Proxy-Authorization: Digest username="8002",realm="bj.pbx.hahacom.com",nonce="448f19699128c00b82244a6d01585e9ad7b01924",uri="sip:013606118682@201.32.22.1",response="d821064c01019b273d7cfc64ac8c52ba",cnonce="13490736624084",qop="auth",nc="00000001" (249) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 12: Babale-hint: NAThelper (22) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 13: Babale-hint: loose_route (24) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 14: Babale-hint: route4 (19) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 15: Babale-hint: route4-donat (25) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3363 parse_request: Header 16: (0) --- (16 headers 0 lines)--- Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 - BYE (No RTP) Jun 14 03:56:07 DEBUG[23362]: chan_sip.c:11137 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 201.32.22.1 : 5060 (NAT) Transmitting (NAT) to 201.32.22.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.32.22.1;branch=z9hG4bK8057.49d325d1.0;received=201.32.22.1 Via: SIP/2.0/UDP 192.168.0.88;received=218.93.158.125;rport=5060;branch=z9hG4bKc0a80058000001ee448f182700001c5a00000979 Record-Route: From: "ppyy";tag=1349071764480 To: ;tag=as6bb02b65 Call-ID: A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88 CSeq: 3 BYE User-Agent: Babale PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call 'A0A90047-50F1-4659-A463-246DB9F73325@192.168.0.88'