[Jun 16 10:28:26] VERBOSE[28962]: Asterisk Event Logger restarted [Jun 16 10:28:26] VERBOSE[28962]: Asterisk Queue Logger restarted [Jun 16 10:28:29] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:28:29] DEBUG[23786]: Header 0: (0) [Jun 16 10:28:29] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:28:29] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:28:29] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:28:32] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: INVITE sip:999@196.40.106.33 SIP/2.0 To: From: Steve 200;tag=02deceb6 Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-1053333316-1--d87543-;rport Call-ID: 1272c55300bbf46c CSeq: 1 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17552 Content-Length: 309 v=0 o=- 256229550 256229624 IN IP4 192.168.1.249 s=eyeBeam c=IN IP4 192.168.1.249 t=0 0 m=audio 10070 RTP/AVP 98 8 0 101 a=alt:1 1 : 1933C098 9A000000 192.168.1.249 10070 a=fmtp:101 0-15 a=rtpmap:98 ilbc/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv [Jun 16 10:28:32] DEBUG[23786]: Header 0: INVITE sip:999@196.40.106.33 SIP/2.0 (36) [Jun 16 10:28:32] DEBUG[23786]: Header 1: To: (27) [Jun 16 10:28:32] DEBUG[23786]: Header 2: From: Steve 200;tag=02deceb6 (51) [Jun 16 10:28:32] DEBUG[23786]: Header 3: Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-1053333316-1--d87543-;rport (85) [Jun 16 10:28:32] DEBUG[23786]: Header 4: Call-ID: 1272c55300bbf46c (25) [Jun 16 10:28:32] DEBUG[23786]: Header 5: CSeq: 1 INVITE (14) [Jun 16 10:28:32] DEBUG[23786]: Header 6: Contact: (37) [Jun 16 10:28:32] DEBUG[23786]: Header 7: Max-Forwards: 70 (16) [Jun 16 10:28:32] DEBUG[23786]: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jun 16 10:28:32] DEBUG[23786]: Header 9: Content-Type: application/sdp (29) [Jun 16 10:28:32] DEBUG[23786]: Header 10: User-Agent: eyeBeam release 3006o stamp 17552 (45) [Jun 16 10:28:32] DEBUG[23786]: Header 11: Content-Length: 309 (19) [Jun 16 10:28:32] DEBUG[23786]: Header 12: (0) [Jun 16 10:28:32] DEBUG[23786]: Line: v=0 (3) [Jun 16 10:28:32] DEBUG[23786]: Line: o=- 256229550 256229624 IN IP4 192.168.1.249 (44) [Jun 16 10:28:32] DEBUG[23786]: Line: s=eyeBeam (9) [Jun 16 10:28:32] DEBUG[23786]: Line: c=IN IP4 192.168.1.249 (22) [Jun 16 10:28:32] DEBUG[23786]: Line: t=0 0 (5) [Jun 16 10:28:32] DEBUG[23786]: Line: m=audio 10070 RTP/AVP 98 8 0 101 (32) [Jun 16 10:28:32] DEBUG[23786]: Line: a=alt:1 1 : 1933C098 9A000000 192.168.1.249 10070 (49) [Jun 16 10:28:32] DEBUG[23786]: Line: a=fmtp:101 0-15 (15) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:98 ilbc/8000 (21) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:8 pcma/8000 (20) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:0 pcmu/8000 (20) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 16 10:28:32] DEBUG[23786]: Line: a=sendrecv (10) [Jun 16 10:28:32] VERBOSE[23786]: --- (12 headers 13 lines)[Jun 16 10:28:32] VERBOSE[23786]: --- (12 headers 13 lines)--- [Jun 16 10:28:32] DEBUG[23786]: Allocating new SIP dialog for 1272c55300bbf46c - INVITE (With RTP) [Jun 16 10:28:32] DEBUG[23786]: **** Received INVITE (5) - Command in SIP INVITE [Jun 16 10:28:32] VERBOSE[23786]: Sending to 165.165.21.61 : 8259 (NAT) [Jun 16 10:28:32] VERBOSE[23786]: Using INVITE request as basis request - 1272c55300bbf46c [Jun 16 10:28:32] DEBUG[23786]: Setting NAT on RTP to On [Jun 16 10:28:32] VERBOSE[23786]: Reliably Transmitting (NAT) to 165.165.21.61:8259: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-1053333316-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as32237cdb Call-ID: 1272c55300bbf46c CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="005f8c1b" Content-Length: 0 --- [Jun 16 10:28:32] DEBUG[23786]: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10 [Jun 16 10:28:32] VERBOSE[23786]: Scheduling destruction of SIP dialog '1272c55300bbf46c' in 32000 ms (Method: INVITE) [Jun 16 10:28:32] VERBOSE[23786]: Found user '200' [Jun 16 10:28:32] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: ACK sip:999@196.40.106.33 SIP/2.0 To: ;tag=as32237cdb From: Steve 200;tag=02deceb6 Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-1053333316-1--d87543-;rport Call-ID: 1272c55300bbf46c CSeq: 1 ACK Content-Length: 0 [Jun 16 10:28:32] DEBUG[23786]: Header 0: ACK sip:999@196.40.106.33 SIP/2.0 (33) [Jun 16 10:28:32] DEBUG[23786]: Header 1: To: ;tag=as32237cdb (42) [Jun 16 10:28:32] DEBUG[23786]: Header 2: From: Steve 200;tag=02deceb6 (51) [Jun 16 10:28:32] DEBUG[23786]: Header 3: Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-1053333316-1--d87543-;rport (85) [Jun 16 10:28:32] DEBUG[23786]: Header 4: Call-ID: 1272c55300bbf46c (25) [Jun 16 10:28:32] DEBUG[23786]: Header 5: CSeq: 1 ACK (11) [Jun 16 10:28:32] DEBUG[23786]: Header 6: Content-Length: 0 (17) [Jun 16 10:28:32] DEBUG[23786]: Header 7: (0) [Jun 16 10:28:32] VERBOSE[23786]: --- (7 headers 0 lines)[Jun 16 10:28:32] VERBOSE[23786]: --- (7 headers 0 lines)--- [Jun 16 10:28:32] DEBUG[23786]: = Found Their Call ID: 1272c55300bbf46c Their Tag 02deceb6 Our tag: as32237cdb [Jun 16 10:28:32] DEBUG[23786]: **** Received ACK (6) - Command in SIP ACK [Jun 16 10:28:32] DEBUG[23786]: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Jun 16 10:28:32] DEBUG[23786]: Stopping retransmission on '1272c55300bbf46c' of Response 1: Match Not Found [Jun 16 10:28:32] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: INVITE sip:999@196.40.106.33 SIP/2.0 To: From: Steve 200;tag=02deceb6 Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;rport Call-ID: 1272c55300bbf46c CSeq: 2 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="200",realm="asterisk",nonce="005f8c1b",uri="sip:999@196.40.106.33",response="6bae3f985f3dbcca69a749354e68bab2",algorithm=MD5 User-Agent: eyeBeam release 3006o stamp 17552 Content-Length: 309 v=0 o=- 256229550 256229624 IN IP4 192.168.1.249 s=eyeBeam c=IN IP4 192.168.1.249 t=0 0 m=audio 10070 RTP/AVP 98 8 0 101 a=alt:1 1 : 1933C098 9A000000 192.168.1.249 10070 a=fmtp:101 0-15 a=rtpmap:98 ilbc/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv [Jun 16 10:28:32] DEBUG[23786]: Header 0: INVITE sip:999@196.40.106.33 SIP/2.0 (36) [Jun 16 10:28:32] DEBUG[23786]: Header 1: To: (27) [Jun 16 10:28:32] DEBUG[23786]: Header 2: From: Steve 200;tag=02deceb6 (51) [Jun 16 10:28:32] DEBUG[23786]: Header 3: Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;rport (83) [Jun 16 10:28:32] DEBUG[23786]: Header 4: Call-ID: 1272c55300bbf46c (25) [Jun 16 10:28:32] DEBUG[23786]: Header 5: CSeq: 2 INVITE (14) [Jun 16 10:28:32] DEBUG[23786]: Header 6: Contact: (37) [Jun 16 10:28:32] DEBUG[23786]: Header 7: Max-Forwards: 70 (16) [Jun 16 10:28:32] DEBUG[23786]: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jun 16 10:28:32] DEBUG[23786]: Header 9: Content-Type: application/sdp (29) [Jun 16 10:28:32] DEBUG[23786]: Header 10: Proxy-Authorization: Digest username="200",realm="asterisk",nonce="005f8c1b",uri="sip:999@196.40.106.33",response="6bae3f985f3dbcca69a749354e68bab2",algorithm=MD5 (162) [Jun 16 10:28:32] DEBUG[23786]: Header 11: User-Agent: eyeBeam release 3006o stamp 17552 (45) [Jun 16 10:28:32] DEBUG[23786]: Header 12: Content-Length: 309 (19) [Jun 16 10:28:32] DEBUG[23786]: Header 13: (0) [Jun 16 10:28:32] DEBUG[23786]: Line: v=0 (3) [Jun 16 10:28:32] DEBUG[23786]: Line: o=- 256229550 256229624 IN IP4 192.168.1.249 (44) [Jun 16 10:28:32] DEBUG[23786]: Line: s=eyeBeam (9) [Jun 16 10:28:32] DEBUG[23786]: Line: c=IN IP4 192.168.1.249 (22) [Jun 16 10:28:32] DEBUG[23786]: Line: t=0 0 (5) [Jun 16 10:28:32] DEBUG[23786]: Line: m=audio 10070 RTP/AVP 98 8 0 101 (32) [Jun 16 10:28:32] DEBUG[23786]: Line: a=alt:1 1 : 1933C098 9A000000 192.168.1.249 10070 (49) [Jun 16 10:28:32] DEBUG[23786]: Line: a=fmtp:101 0-15 (15) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:98 ilbc/8000 (21) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:8 pcma/8000 (20) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:0 pcmu/8000 (20) [Jun 16 10:28:32] DEBUG[23786]: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 16 10:28:32] DEBUG[23786]: Line: a=sendrecv (10) [Jun 16 10:28:32] VERBOSE[23786]: --- (13 headers 13 lines)[Jun 16 10:28:32] VERBOSE[23786]: --- (13 headers 13 lines)--- [Jun 16 10:28:32] DEBUG[23786]: = Found Their Call ID: 1272c55300bbf46c Their Tag 02deceb6 Our tag: as32237cdb [Jun 16 10:28:32] DEBUG[23786]: **** Received INVITE (5) - Command in SIP INVITE [Jun 16 10:28:32] VERBOSE[23786]: Sending to 165.165.21.61 : 8259 (NAT) [Jun 16 10:28:32] VERBOSE[23786]: Using INVITE request as basis request - 1272c55300bbf46c [Jun 16 10:28:32] DEBUG[23786]: Setting NAT on RTP to On [Jun 16 10:28:32] VERBOSE[23786]: Found user '200' [Jun 16 10:28:32] VERBOSE[23786]: Found RTP audio format 98 [Jun 16 10:28:32] VERBOSE[23786]: Found RTP audio format 8 [Jun 16 10:28:32] VERBOSE[23786]: Found RTP audio format 0 [Jun 16 10:28:32] VERBOSE[23786]: Found RTP audio format 101 [Jun 16 10:28:32] VERBOSE[23786]: Peer audio RTP is at port 192.168.1.249:10070 [Jun 16 10:28:32] VERBOSE[23786]: Found description format alt:1 1 : 1933C098 9A000000 192.168.1.249 10070 for ID 101 [Jun 16 10:28:32] VERBOSE[23786]: Got unsupported a:fmtp in SDP offer [Jun 16 10:28:32] VERBOSE[23786]: Found description format rtpmap:98 ilbc/8000 for ID 101 [Jun 16 10:28:32] VERBOSE[23786]: Found description format rtpmap:8 pcma/8000 for ID 101 [Jun 16 10:28:32] VERBOSE[23786]: Found description format rtpmap:0 pcmu/8000 for ID 101 [Jun 16 10:28:32] VERBOSE[23786]: Found description format rtpmap:101 telephone-event/8000 for ID 101 [Jun 16 10:28:32] DEBUG[23786]: T38 state changed to 0 on channel [Jun 16 10:28:32] VERBOSE[23786]: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0xc (ulaw|alaw) [Jun 16 10:28:32] VERBOSE[23786]: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 16 10:28:32] VERBOSE[23786]: Peer audio RTP is at port 192.168.1.249:10070 [Jun 16 10:28:32] DEBUG[23786]: We're settling with these formats: 0xc (ulaw|alaw) [Jun 16 10:28:32] DEBUG[23786]: Checking SIP call limits for device 200 [Jun 16 10:28:32] DEBUG[23786]: Updating call counter for incoming call [Jun 16 10:28:32] VERBOSE[23786]: Looking for 999 in default (domain 196.40.106.33) [Jun 16 10:28:32] DEBUG[23786]: *** Our native formats are 0x8 (alaw) [Jun 16 10:28:32] DEBUG[23786]: *** Joint capabilities are 0xc (ulaw|alaw) [Jun 16 10:28:32] DEBUG[23786]: *** Our capabilities are 0xc (ulaw|alaw) [Jun 16 10:28:32] DEBUG[23786]: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 16 10:28:32] DEBUG[23786]: This channel will not be able to handle video. [Jun 16 10:28:32] DEBUG[23786]: build_route: Contact hop: [Jun 16 10:28:32] VERBOSE[23786]: list_route: hop: [Jun 16 10:28:32] DEBUG[23786]: SIP/200-3ab0: New call is still down.... Trying... [Jun 16 10:28:32] VERBOSE[23786]: Transmitting (NAT) to 165.165.21.61:8259: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:32] DEBUG[21319]: Checking device state for peer 200 [Jun 16 10:28:32] DEBUG[21319]: Changing state for SIP/200 - state 1 (Not in use) [Jun 16 10:28:32] DEBUG[20743]: Launching 'Wait' [Jun 16 10:28:32] VERBOSE[20743]: -- Executing [999@default:1] Wait("SIP/200-3ab0", "1") in new stack [Jun 16 10:28:32] DEBUG[31990]: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 16 10:28:33] DEBUG[20743]: Launching 'Answer' [Jun 16 10:28:33] VERBOSE[20743]: -- Executing [999@default:2] Answer("SIP/200-3ab0", "") in new stack [Jun 16 10:28:33] DEBUG[20743]: SIP answering channel: SIP/200-3ab0 [Jun 16 10:28:33] DEBUG[20743]: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jun 16 10:28:33] DEBUG[20743]: ** Our prefcodec: 0x0 (nothing) [Jun 16 10:28:33] VERBOSE[20743]: Audio is at 196.40.106.33 port 17018 [Jun 16 10:28:33] VERBOSE[20743]: Adding codec 0x8 (alaw) to SDP [Jun 16 10:28:33] VERBOSE[20743]: Adding codec 0x4 (ulaw) to SDP [Jun 16 10:28:33] VERBOSE[20743]: Adding non-codec 0x1 (telephone-event) to SDP [Jun 16 10:28:33] DEBUG[20743]: -- Done with adding codecs to SDP [Jun 16 10:28:33] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:33] DEBUG[20743]: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jun 16 10:28:33] VERBOSE[20743]: Reliably Transmitting (NAT) to 165.165.21.61:8259: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 254 v=0 o=root 20743 20743 IN IP4 196.40.106.33 s=session c=IN IP4 196.40.106.33 t=0 0 m=audio 17018 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 - - - - a=sendrecv --- [Jun 16 10:28:33] DEBUG[20743]: *** SIP TIMER: Initalizing retransmit timer on packet: Id #12 [Jun 16 10:28:33] DEBUG[20743]: Launching 'Wait' [Jun 16 10:28:33] VERBOSE[20743]: -- Executing [999@default:3] Wait("SIP/200-3ab0", "2") in new stack [Jun 16 10:28:33] DEBUG[21319]: Checking device state for peer 200 [Jun 16 10:28:33] DEBUG[21319]: Changing state for SIP/200 - state 1 (Not in use) [Jun 16 10:28:33] DEBUG[31581]: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 16 10:28:33] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: ACK sip:999@196.40.106.33 SIP/2.0 To: ;tag=as183cfcfe From: Steve 200;tag=02deceb6 Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-341652481-1--d87543-;rport Call-ID: 1272c55300bbf46c CSeq: 2 ACK Contact: Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="005f8c1b",uri="sip:999@196.40.106.33",response="6bae3f985f3dbcca69a749354e68bab2",algorithm=MD5 User-Agent: eyeBeam release 3006o stamp 17552 Content-Length: 0 [Jun 16 10:28:33] DEBUG[23786]: Header 0: ACK sip:999@196.40.106.33 SIP/2.0 (33) [Jun 16 10:28:33] DEBUG[23786]: Header 1: To: ;tag=as183cfcfe (42) [Jun 16 10:28:33] DEBUG[23786]: Header 2: From: Steve 200;tag=02deceb6 (51) [Jun 16 10:28:33] DEBUG[23786]: Header 3: Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-341652481-1--d87543-;rport (84) [Jun 16 10:28:33] DEBUG[23786]: Header 4: Call-ID: 1272c55300bbf46c (25) [Jun 16 10:28:33] DEBUG[23786]: Header 5: CSeq: 2 ACK (11) [Jun 16 10:28:33] DEBUG[23786]: Header 6: Contact: (37) [Jun 16 10:28:33] DEBUG[23786]: Header 7: Max-Forwards: 70 (16) [Jun 16 10:28:33] DEBUG[23786]: Header 8: Proxy-Authorization: Digest username="200",realm="asterisk",nonce="005f8c1b",uri="sip:999@196.40.106.33",response="6bae3f985f3dbcca69a749354e68bab2",algorithm=MD5 (162) [Jun 16 10:28:33] DEBUG[23786]: Header 9: User-Agent: eyeBeam release 3006o stamp 17552 (45) [Jun 16 10:28:33] DEBUG[23786]: Header 10: Content-Length: 0 (17) [Jun 16 10:28:33] DEBUG[23786]: Header 11: (0) [Jun 16 10:28:33] VERBOSE[23786]: --- (11 headers 0 lines)[Jun 16 10:28:33] VERBOSE[23786]: --- (11 headers 0 lines)--- [Jun 16 10:28:33] DEBUG[23786]: = Found Their Call ID: 1272c55300bbf46c Their Tag 02deceb6 Our tag: as183cfcfe [Jun 16 10:28:33] DEBUG[23786]: **** Received ACK (6) - Command in SIP ACK [Jun 16 10:28:33] DEBUG[23786]: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #12 [Jun 16 10:28:33] DEBUG[23786]: Stopping retransmission on '1272c55300bbf46c' of Response 2: Match Not Found [Jun 16 10:28:33] DEBUG[20743]: RTCP NAT: Got RTCP from other end. Now sending to address 165.165.21.61:10071 [Jun 16 10:28:33] DEBUG[20743]: Got RTCP report of 72 bytes [Jun 16 10:28:33] DEBUG[20743]: RTP NAT: Got audio from other end. Now sending to address 165.165.21.61:10070 [Jun 16 10:28:35] DEBUG[20743]: Launching 'Busy' [Jun 16 10:28:35] VERBOSE[20743]: -- Executing [999@default:4] Busy("SIP/200-3ab0", "5") in new stack [Jun 16 10:28:35] DEBUG[20743]: Driver for channel 'SIP/200-3ab0' does not support indication 5, emulating it [Jun 16 10:28:35] DEBUG[20743]: Set channel SIP/200-3ab0 to write format slin [Jun 16 10:28:35] DEBUG[20743]: Scheduling timer at 160 sample intervals [Jun 16 10:28:35] DEBUG[21319]: Checking device state for peer 200 [Jun 16 10:28:35] DEBUG[21319]: Changing state for SIP/200 - state 1 (Not in use) [Jun 16 10:28:35] DEBUG[7251]: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 16 10:28:35] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:35] DEBUG[20743]: Generator got voice, switching to phase locked mode [Jun 16 10:28:35] DEBUG[20743]: Scheduling timer at 0 sample intervals [Jun 16 10:28:35] DEBUG[20743]: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jun 16 10:28:35] DEBUG[20743]: ** Our prefcodec: 0x0 (nothing) [Jun 16 10:28:35] VERBOSE[20743]: Audio is at 196.40.106.33 port 17018 [Jun 16 10:28:35] VERBOSE[20743]: Adding codec 0x8 (alaw) to SDP [Jun 16 10:28:35] VERBOSE[20743]: Adding codec 0x4 (ulaw) to SDP [Jun 16 10:28:35] VERBOSE[20743]: Adding non-codec 0x1 (telephone-event) to SDP [Jun 16 10:28:35] DEBUG[20743]: -- Done with adding codecs to SDP [Jun 16 10:28:35] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:35] DEBUG[20743]: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jun 16 10:28:35] VERBOSE[20743]: Transmitting (NAT) to 165.165.21.61:8259: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 254 v=0 o=root 20743 20744 IN IP4 196.40.106.33 s=session c=IN IP4 196.40.106.33 t=0 0 m=audio 17018 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 - - - - a=sendrecv --- [Jun 16 10:28:35] DEBUG[20743]: Ooh, format changed from unknown to alaw [Jun 16 10:28:35] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) ... lots of those ... [Jun 16 10:28:37] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Got RTCP report of 52 bytes [Jun 16 10:28:38] DEBUG[20743]: Got RTCP report of 72 bytes [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:38] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:28:38] DEBUG[23786]: Header 0: (0) [Jun 16 10:28:38] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:28:38] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:28:38] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:28:38] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) ... lots of those again ... [Jun 16 10:28:39] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:39] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:39] DEBUG[20743]: Got RTCP report of 52 bytes [Jun 16 10:28:40] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) ... lots of those again ... [Jun 16 10:28:41] DEBUG[20743]: Internal timing is disabled (option_internal_timing=0 chan->timingfd=23) [Jun 16 10:28:41] DEBUG[20743]: Spawn extension (default,999,4) exited non-zero on 'SIP/200-3ab0' [Jun 16 10:28:41] DEBUG[20743]: Set channel SIP/200-3ab0 to write format alaw [Jun 16 10:28:41] DEBUG[20743]: Function result is '"Steve 200" <200>' [Jun 16 10:28:41] DEBUG[20743]: Function result is '200' [Jun 16 10:28:41] DEBUG[20743]: Function result is '999' [Jun 16 10:28:41] DEBUG[20743]: Function result is 'default' [Jun 16 10:28:41] DEBUG[20743]: Function result is 'SIP/200-3ab0' [Jun 16 10:28:41] DEBUG[20743]: Function result is '' [Jun 16 10:28:41] DEBUG[20743]: Function result is 'Busy' [Jun 16 10:28:41] DEBUG[20743]: Function result is '5' [Jun 16 10:28:41] DEBUG[20743]: Function result is '2006-06-16 10:28:32' [Jun 16 10:28:41] DEBUG[20743]: Function result is '2006-06-16 10:28:33' [Jun 16 10:28:41] DEBUG[20743]: Function result is '2006-06-16 10:28:41' [Jun 16 10:28:41] DEBUG[20743]: Function result is '9' [Jun 16 10:28:41] DEBUG[20743]: Function result is '8' [Jun 16 10:28:41] DEBUG[20743]: Function result is 'ANSWERED' [Jun 16 10:28:41] DEBUG[20743]: Function result is 'DOCUMENTATION' [Jun 16 10:28:41] DEBUG[20743]: Function result is '' [Jun 16 10:28:41] DEBUG[20743]: Function result is '1150446512.1' [Jun 16 10:28:41] DEBUG[20743]: Function result is '' [Jun 16 10:28:41] DEBUG[20743]: Hanging up channel 'SIP/200-3ab0' [Jun 16 10:28:41] DEBUG[20743]: Hangup call SIP/200-3ab0, SIP callid 1272c55300bbf46c) [Jun 16 10:28:41] DEBUG[20743]: update_call_counter(200) - decrement call limit counter on hangup [Jun 16 10:28:41] DEBUG[20743]: Updating call counter for incoming call [Jun 16 10:28:41] DEBUG[20743]: Hanging up channel in state Busy (not UP) [Jun 16 10:28:41] VERBOSE[20743]: Reliably Transmitting (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:41] DEBUG[20743]: *** SIP TIMER: Initalizing retransmit timer on packet: Id #14 [Jun 16 10:28:41] DEBUG[21319]: Checking device state for peer 200 [Jun 16 10:28:41] DEBUG[21319]: Changing state for SIP/200 - state 1 (Not in use) [Jun 16 10:28:41] DEBUG[12002]: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 16 10:28:42] DEBUG[23786]: SIP TIMER: Rescheduling retransmission #14 (1) SIP/2.0 - 1 [Jun 16 10:28:42] DEBUG[23786]: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #14)) [Jun 16 10:28:42] VERBOSE[23786]: Retransmitting #1 (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:43] DEBUG[23786]: SIP TIMER: Rescheduling retransmission #14 (2) SIP/2.0 - 1 [Jun 16 10:28:43] DEBUG[23786]: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #14)) [Jun 16 10:28:43] VERBOSE[23786]: Retransmitting #2 (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:45] DEBUG[23786]: SIP TIMER: Rescheduling retransmission #14 (3) SIP/2.0 - 1 [Jun 16 10:28:45] DEBUG[23786]: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #14)) [Jun 16 10:28:45] VERBOSE[23786]: Retransmitting #3 (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:47] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:28:47] DEBUG[23786]: Header 0: (0) [Jun 16 10:28:47] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:28:47] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:28:47] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:28:49] DEBUG[23786]: SIP TIMER: Rescheduling retransmission #14 (4) SIP/2.0 - 1 [Jun 16 10:28:49] DEBUG[23786]: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #14)) [Jun 16 10:28:49] VERBOSE[23786]: Retransmitting #4 (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:53] DEBUG[23786]: SIP TIMER: Rescheduling retransmission #14 (5) SIP/2.0 - 1 [Jun 16 10:28:53] DEBUG[23786]: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #14)) [Jun 16 10:28:53] VERBOSE[23786]: Retransmitting #5 (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:28:56] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:28:56] DEBUG[23786]: Header 0: (0) [Jun 16 10:28:56] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:28:56] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:28:56] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:28:57] DEBUG[23786]: SIP TIMER: Rescheduling retransmission #14 (6) SIP/2.0 - 1 [Jun 16 10:28:57] DEBUG[23786]: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #14)) [Jun 16 10:28:57] VERBOSE[23786]: Retransmitting #6 (NAT) to 165.165.21.61:8259: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-38794511-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Jun 16 10:29:01] WARNING[23786]: Maximum retries exceeded on transmission 1272c55300bbf46c for seqno 2 (Critical Response) [Jun 16 10:29:01] VERBOSE[23786]: Really destroying SIP dialog '1272c55300bbf46c' Method: ACK [Jun 16 10:29:05] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:29:05] DEBUG[23786]: Header 0: (0) [Jun 16 10:29:05] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:29:05] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:29:05] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:29:14] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:29:14] DEBUG[23786]: Header 0: (0) [Jun 16 10:29:14] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:29:14] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:29:14] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:29:23] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:29:23] DEBUG[23786]: Header 0: (0) [Jun 16 10:29:23] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:29:23] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:29:23] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:29:27] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: BYE sip:999@196.40.106.33 SIP/2.0 To: ;tag=as183cfcfe From: Steve 200;tag=02deceb6 Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-722767870-1--d87543-;rport Call-ID: 1272c55300bbf46c CSeq: 3 BYE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="200",realm="asterisk",nonce="005f8c1b",uri="sip:999@196.40.106.33",response="a75aeda39d8ad2a2ebee020a50f6273f",algorithm=MD5 User-Agent: eyeBeam release 3006o stamp 17552 Content-Length: 0 [Jun 16 10:29:27] DEBUG[23786]: Header 0: BYE sip:999@196.40.106.33 SIP/2.0 (33) [Jun 16 10:29:27] DEBUG[23786]: Header 1: To: ;tag=as183cfcfe (42) [Jun 16 10:29:27] DEBUG[23786]: Header 2: From: Steve 200;tag=02deceb6 (51) [Jun 16 10:29:27] DEBUG[23786]: Header 3: Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-722767870-1--d87543-;rport (84) [Jun 16 10:29:27] DEBUG[23786]: Header 4: Call-ID: 1272c55300bbf46c (25) [Jun 16 10:29:27] DEBUG[23786]: Header 5: CSeq: 3 BYE (11) [Jun 16 10:29:27] DEBUG[23786]: Header 6: Contact: (37) [Jun 16 10:29:27] DEBUG[23786]: Header 7: Max-Forwards: 70 (16) [Jun 16 10:29:27] DEBUG[23786]: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jun 16 10:29:27] DEBUG[23786]: Header 9: Proxy-Authorization: Digest username="200",realm="asterisk",nonce="005f8c1b",uri="sip:999@196.40.106.33",response="a75aeda39d8ad2a2ebee020a50f6273f",algorithm=MD5 (162) [Jun 16 10:29:27] DEBUG[23786]: Header 10: User-Agent: eyeBeam release 3006o stamp 17552 (45) [Jun 16 10:29:27] DEBUG[23786]: Header 11: Content-Length: 0 (17) [Jun 16 10:29:27] DEBUG[23786]: Header 12: (0) [Jun 16 10:29:27] VERBOSE[23786]: --- (12 headers 0 lines)[Jun 16 10:29:27] VERBOSE[23786]: --- (12 headers 0 lines)--- [Jun 16 10:29:27] DEBUG[23786]: Allocating new SIP dialog for 1272c55300bbf46c - BYE (No RTP) [Jun 16 10:29:27] DEBUG[23786]: **** Received BYE (8) - Command in SIP BYE [Jun 16 10:29:27] VERBOSE[23786]: Sending to 165.165.21.61 : 8259 (NAT) [Jun 16 10:29:27] VERBOSE[23786]: Transmitting (NAT) to 165.165.21.61:8259: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.249:8259;branch=z9hG4bK-d87543-722767870-1--d87543-;received=165.165.21.61;rport=8259 From: Steve 200;tag=02deceb6 To: ;tag=as183cfcfe Call-ID: 1272c55300bbf46c CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 16 10:29:27] VERBOSE[23786]: Really destroying SIP dialog '1272c55300bbf46c' Method: BYE [Jun 16 10:29:31] VERBOSE[28962]: -- Remote UNIX connection disconnected [Jun 16 10:29:32] VERBOSE[23786]: <-- SIP read from 165.165.21.61:8259: [Jun 16 10:29:32] DEBUG[23786]: Header 0: (0) [Jun 16 10:29:32] VERBOSE[23786]: --- (0 headers 0 lines)[Jun 16 10:29:32] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive [Jun 16 10:29:32] VERBOSE[23786]: --- (0 headers 0 lines) Nat keepalive --- [Jun 16 10:29:34] VERBOSE[13826]: Executing last minute cleanups [Jun 16 10:29:34] VERBOSE[13826]: == Destroying musiconhold processes [Jun 16 10:29:34] DEBUG[13826]: Asterisk ending (15).