Jun 29 14:12:14 DEBUG[9679]: rtp.c:193 send_dtmf: Sending dtmf: 42 (*), at 62.3.241.21 Jun 29 14:12:14 DEBUG[9679]: channel.c:3303 ast_generic_bridge: Got DTMF on channel (SIP/212.248.181.59-08636d00) Jun 29 14:12:14 DEBUG[9679]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels SIP/212.248.181.59-08636d00 and SIP/nta-2db6 Jun 29 14:12:14 DEBUG[9679]: res_features.c:996 ast_feature_interpret: Feature interpret: chan=SIP/212.248.181.59-08636d00, peer=SIP/nta-2db6, sense=1, features=1 Jun 29 14:12:14 DEBUG[9679]: res_features.c:1473 ast_bridge_call: Set time limit to 500 Jun 29 14:12:15 DEBUG[9679]: rtp.c:193 send_dtmf: Sending dtmf: 42 (*), at 62.3.241.21 Jun 29 14:12:15 DEBUG[9679]: channel.c:3303 ast_generic_bridge: Got DTMF on channel (SIP/212.248.181.59-08636d00) Jun 29 14:12:15 DEBUG[9679]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels SIP/212.248.181.59-08636d00 and SIP/nta-2db6 Jun 29 14:12:15 DEBUG[9679]: res_features.c:996 ast_feature_interpret: Feature interpret: chan=SIP/212.248.181.59-08636d00, peer=SIP/nta-2db6, sense=1, features=0 -- Feature Found: park_caller exten: park_caller == Parked SIP/212.248.181.59-08636d00 on 701. Will timeout back to extension [nta] s, 1 in 45 seconds Jun 29 14:12:16 DEBUG[9679]: pbx.c:4876 ast_add_extension2: Added extension '701' priority 1 to parkedcalls Jun 29 14:12:16 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format gsm Jun 29 14:12:16 DEBUG[9679]: rtp.c:1260 ast_rtp_raw_write: Difference is 8120, ms is 1035 Jun 29 14:12:16 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'digits/7' (language 'en') Jun 29 14:12:16 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 14:12:16 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 14:12:16 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format slin Jun 29 14:12:16 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format gsm Jun 29 14:12:16 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'digits/0' (language 'en') Jun 29 14:12:17 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 14:12:17 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 14:12:17 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format slin Jun 29 14:12:17 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format gsm Jun 29 14:12:17 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'digits/1' (language 'en') Jun 29 14:12:18 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 14:12:18 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 14:12:18 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format slin Jun 29 14:12:18 DEBUG[9679]: channel.c:2350 set_format: Set channel SIP/212.248.181.59-08636d00 to write format slin -- Started music on hold, class 'default', on channel 'SIP/212.248.181.59-08636d00' Jun 29 14:12:18 DEBUG[9679]: channel.c:1711 ast_settimeout: Scheduling timer at 160 sample intervals Jun 29 14:12:18 DEBUG[9679]: channel.c:1323 ast_hangup: Hanging up channel 'SIP/nta-2db6' Jun 29 14:12:18 DEBUG[9679]: chan_sip.c:2418 sip_hangup: Hangup call SIP/nta-2db6, SIP callid 76a4133442939f334f79c5414bf1c81e@212.187.189.12) Jun 29 14:12:18 DEBUG[9679]: chan_sip.c:2426 sip_hangup: update_call_counter(07870699479) - decrement call limit counter Jun 29 14:12:18 DEBUG[9679]: chan_sip.c:2209 update_call_counter: Updating call counter for outgoing call set_destination: Parsing for address/port to send to set_destination: set destination to 212.187.189.12, port 5060 Reliably Transmitting (no NAT) to 212.187.189.12:5060: BYE sip:07870699479@212.187.189.14:5060 SIP/2.0 Via: SIP/2.0/UDP 212.248.181.59:5070;branch=z9hG4bK35e8575d;rport Route: From: "Unknown" ;tag=as38070a22 To: ;tag=as6f901c08 Contact: Call-ID: 76a4133442939f334f79c5414bf1c81e@212.187.189.12 CSeq: 104 BYE User-Agent: Asterisk Max-Forwards: 70 Proxy-Authorization: Content-Length: 0 --- Jun 29 14:12:18 DEBUG[9679]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #119241 Jun 29 14:12:18 DEBUG[9679]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=ANSWER. Jun 29 14:12:18 DEBUG[9679]: pbx.c:2309 __ast_pbx_run: Spawn extension (nta,s,1) exited KEEPALIVE on 'SIP/212.248.181.59-08636d00' Jun 29 14:12:18 DEBUG[2835]: chan_sip.c:11668 sip_devicestate: Checking device state for peer nta Jun 29 14:12:18 DEBUG[2835]: devicestate.c:187 do_state_change: Changing state for SIP/nta - state 1 (Not in use) Jun 29 14:12:18 DEBUG[9685]: app_queue.c:523 changethread: Device 'SIP/nta' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 29 14:12:18 DEBUG[2840]: channel.c:1975 ast_read: Generator got voice, switching to phase locked mode Jun 29 14:12:18 DEBUG[2840]: channel.c:1711 ast_settimeout: Scheduling timer at 0 sample intervals itsp3*CLI> <-- SIP read from 212.187.189.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.248.181.59:5070;branch=z9hG4bK35e8575d;rport=5070 Record-Route: From: "Unknown" ;tag=as38070a22 To: ;tag=as6f901c08 Call-ID: 76a4133442939f334f79c5414bf1c81e@212.187.189.12 CSeq: 104 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 200 OK (14) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 212.248.181.59:5070;branch=z9hG4bK35e8575d;rport=5070 (70) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 2: Record-Route: (56) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 3: From: "Unknown" ;tag=as38070a22 (60) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 4: To: ;tag=as6f901c08 (51) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 5: Call-ID: 76a4133442939f334f79c5414bf1c81e@212.187.189.12 (56) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 6: CSeq: 104 BYE (13) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 7: User-Agent: Asterisk (21) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 9: Contact: (46) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 10: Content-Length: 0 (17) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 11: X-Asterisk-HangupCause: Normal Clearing (39) Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3363 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:3195 find_call: = Found Their Call ID: 76a4133442939f334f79c5414bf1c81e@212.187.189.12 Their Tag as6f901c08 Our tag: as38070a22 Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #119241 Jun 29 14:12:18 DEBUG[2849]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '76a4133442939f334f79c5414bf1c81e@212.187.189.12' of Request 104: Match Found Destroying call '76a4133442939f334f79c5414bf1c81e@212.187.189.12' Jun 29 14:12:19 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:20 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:20 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:21 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:21 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:21 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:22 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:22 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:22 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:22 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:23 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:24 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe Jun 29 14:12:24 DEBUG[2836]: res_musiconhold.c:545 monmp3thread: Only wrote -1 of 640 bytes to pipe