[Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 0: INVITE sip:31880016499@devel.alt001.com SIP/2.0 (47) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 1: Record-Route: (55) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 2: Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK48ae.993b081.0 (59) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 3: Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK1935ef52;rport=8066 (70) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 4: From: " " ;tag=as16153a28 (62) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 5: To: (44) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 6: Contact: (46) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 7: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 8: CSeq: 102 INVITE (16) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 9: User-Agent: SpeakUp PSTN (24) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 10: Max-Forwards: 69 (16) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 11: Date: Thu, 17 Apr 2008 07:42:41 GMT (35) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 14: Content-Length: 267 (19) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 15: (0) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: o=root 23936 23936 IN IP4 217.114.103.72 (40) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Setting NAT on RTP to Off [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Allocating new SIP dialog for 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 - INVITE (With RTP) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Setting NAT on RTP to Off [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Checking SIP call limits for device alt001-test [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Updating call counter for incoming call [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: *** Our native formats are 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: This channel will not be able to handle video. [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: build_route: Record-Route hop: [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: SIP/alt001-test-08199278: New call is still down.... Trying... [Apr 17 09:42:41] DEBUG[32503] devicestate.c: Notification of state change to be queued on device/channel SIP/alt001-test-08199278 [Apr 17 09:42:41] DEBUG[32503] devicestate.c: Notification of state change to be queued on device/channel SIP/alt001-test [Apr 17 09:42:41] DEBUG[32571] pbx.c: Launching 'Set' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Launching 'Goto' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Function result is '31152563045' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Launching 'Log' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Function result is '09:42:41' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Function result is '31152563045' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Function result is ' ' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Launching 'Verbose' [Apr 17 09:42:41] DEBUG[32571] pbx.c: Launching 'Dial' [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Setting NAT on RTP to Off [Apr 17 09:42:41] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - alt001-test-08199278 [Apr 17 09:42:41] DEBUG[32501] chan_sip.c: Checking device state for peer alt001-test-08199278 [Apr 17 09:42:41] DEBUG[32501] devicestate.c: Changing state for SIP/alt001-test-08199278 - state 4 (Invalid) [Apr 17 09:42:41] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - alt001-test [Apr 17 09:42:41] DEBUG[32501] chan_sip.c: Checking device state for peer alt001-test [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: *** Our native formats are 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: *** Our capabilities are 0x40a (gsm|alaw|ilbc) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: This channel will not be able to handle video. [Apr 17 09:42:41] DEBUG[32571] rtp.c: Seeded SDP of 'SIP/test2-0819d1f0' with that of 'SIP/alt001-test-08199278' [Apr 17 09:42:41] DEBUG[32571] channel.c: Not copying variable CUSTOMER_EXT. [Apr 17 09:42:41] DEBUG[32571] channel.c: Not copying variable SIPCALLID. [Apr 17 09:42:41] DEBUG[32571] channel.c: Not copying variable SIPUSERAGENT. [Apr 17 09:42:41] DEBUG[32571] channel.c: Not copying variable SIPDOMAIN. [Apr 17 09:42:41] DEBUG[32571] channel.c: Not copying variable SIPURI. [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Outgoing Call for 31880016499 [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Updating call counter for outgoing call [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: ** Our capability: 0xa (gsm|alaw) Video flag: False [Apr 17 09:42:41] DEBUG[32501] devicestate.c: Changing state for SIP/alt001-test - state 4 (Invalid) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: -- Done with adding codecs to SDP [Apr 17 09:42:41] DEBUG[32571] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Done building SDP. Settling with this capability: 0xa (gsm|alaw) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 0: INVITE sip:31880016499@92.65.24.219 SIP/2.0 (43) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK0ed54b95;rport (64) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 2: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 3: To: (34) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 4: Contact: (40) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 5: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 7: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 9: Date: Thu, 17 Apr 2008 07:42:41 GMT (35) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 13: Content-Length: 265 (19) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Header 14: (0) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: o=root 31837 31837 IN IP4 87.253.148.89 (39) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: s=session (9) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: c=IN IP4 87.253.148.89 (22) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: m=audio 12990 RTP/AVP 8 3 101 (29) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=ptime:20 (10) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: Line: a=sendrecv (10) [Apr 17 09:42:41] DEBUG[32571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK0ed54b95;received=87.253.148.89;rport=5060 (92) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 2: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 3: To: (34) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 8: Supported: replaces (19) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 9: Contact: (39) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 10: Content-Length: 0 (17) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 11: (0) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: = Found Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag Our tag: as0053a0c1 [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: *** SIP TIMER: Cancelling retransmission #27 - INVITE (got response) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1eead9035e96d8f1476c36891de254d5@87.253.148.89' Request 102: Found [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: SIP response 100 to standard invite [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK0ed54b95;received=87.253.148.89;rport=5060 (92) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 2: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=as05f16626 (49) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 8: Supported: replaces (19) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 9: Contact: (39) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 10: Content-Length: 0 (17) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: Header 11: (0) [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: = Found Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag Our tag: as0053a0c1 [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1eead9035e96d8f1476c36891de254d5@87.253.148.89' Request 102: Found [Apr 17 09:42:41] DEBUG[32503] chan_sip.c: SIP response 180 to standard invite [Apr 17 09:42:41] DEBUG[32503] devicestate.c: Notification of state change to be queued on device/channel SIP/test2-0819d1f0 [Apr 17 09:42:41] DEBUG[32503] devicestate.c: Notification of state change to be queued on device/channel SIP/test2 [Apr 17 09:42:41] DEBUG[32571] rtp.c: Setting early bridge SDP of 'SIP/alt001-test-08199278' with that of 'SIP/test2-0819d1f0' [Apr 17 09:42:41] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - test2-0819d1f0 [Apr 17 09:42:41] DEBUG[32501] chan_sip.c: Checking device state for peer test2-0819d1f0 [Apr 17 09:42:41] DEBUG[32501] devicestate.c: Changing state for SIP/test2-0819d1f0 - state 4 (Invalid) [Apr 17 09:42:41] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - test2 [Apr 17 09:42:41] DEBUG[32501] chan_sip.c: Checking device state for peer test2 [Apr 17 09:42:41] DEBUG[32501] devicestate.c: Changing state for SIP/test2 - state 1 (Not in use) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 0: OPTIONS sip:switch2.sip.speakup.nl SIP/2.0 (42) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK127ee098;rport (64) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 2: From: "asterisk" ;tag=as6f99ef45 (60) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 3: To: (32) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 4: Contact: (37) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 5: Call-ID: 6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89 (55) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 7: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 9: Date: Thu, 17 Apr 2008 07:42:47 GMT (35) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 12: Content-Length: 0 (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK127ee098;rport=5060 (69) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 2: From: "asterisk" ;tag=as6f99ef45 (60) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=42d463797deba61b4d623719e04bf1cd.5d5c (74) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89 (55) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 6: Accept: */* (11) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 7: Accept-Encoding: (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 8: Accept-Language: en (19) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 9: Supported: (11) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 10: Server: SpeakUp SIP Proxy (25) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 0 (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89 Their Tag Our tag: as6f99ef45 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Stopping retransmission on '6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89' of Request 102: Match Found [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ---------- SIP HISTORY for '6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89' [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: * SIP Call [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Call '6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89' has no history [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ---------- END SIP HISTORY for '6d51220e791a83ff72cf5ac72490b8c5@87.253.148.89' [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 0: OPTIONS sip:switch1.sip.speakup.nl SIP/2.0 (42) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK09423364;rport (64) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 2: From: "asterisk" ;tag=as19129c8d (60) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 3: To: (32) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 4: Contact: (37) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 5: Call-ID: 3fa400be466175b1172686a418264271@87.253.148.89 (55) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 7: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 9: Date: Thu, 17 Apr 2008 07:42:47 GMT (35) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 12: Content-Length: 0 (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK09423364;rport=5060 (69) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 2: From: "asterisk" ;tag=as19129c8d (60) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=2cf07e53f64bf7f426c2eaafa79bee36.7634 (74) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 3fa400be466175b1172686a418264271@87.253.148.89 (55) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 6: Accept: */* (11) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 7: Accept-Encoding: (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 8: Accept-Language: en (19) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 9: Supported: (11) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 10: Server: SpeakUp SIP Proxy (25) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 0 (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: = Found Their Call ID: 3fa400be466175b1172686a418264271@87.253.148.89 Their Tag Our tag: as19129c8d [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #32 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Stopping retransmission on '3fa400be466175b1172686a418264271@87.253.148.89' of Request 102: Match Found [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ---------- SIP HISTORY for '3fa400be466175b1172686a418264271@87.253.148.89' [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: * SIP Call [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Call '3fa400be466175b1172686a418264271@87.253.148.89' has no history [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ---------- END SIP HISTORY for '3fa400be466175b1172686a418264271@87.253.148.89' [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 0: OPTIONS sip:s@92.65.24.219 SIP/2.0 (34) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK1a983920;rport (64) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 2: From: "asterisk" ;tag=as0498d324 (60) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 3: To: (24) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 4: Contact: (37) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 5: Call-ID: 630f809f4145b35d72a542370d330980@87.253.148.89 (55) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 7: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 9: Date: Thu, 17 Apr 2008 07:42:47 GMT (35) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 12: Content-Length: 0 (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 404 Not Found (21) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK1a983920;received=87.253.148.89;rport=5060 (92) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 2: From: "asterisk" ;tag=as0498d324 (60) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=as2e341e71 (39) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 630f809f4145b35d72a542370d330980@87.253.148.89 (55) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 8: Supported: replaces (19) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 9: Accept: application/sdp (23) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 10: Content-Length: 0 (17) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Header 11: (0) [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: = Found Their Call ID: 630f809f4145b35d72a542370d330980@87.253.148.89 Their Tag Our tag: as0498d324 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Stopping retransmission on '630f809f4145b35d72a542370d330980@87.253.148.89' of Request 102: Match Found [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ---------- SIP HISTORY for '630f809f4145b35d72a542370d330980@87.253.148.89' [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: * SIP Call [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: Call '630f809f4145b35d72a542370d330980@87.253.148.89' has no history [Apr 17 09:42:47] DEBUG[32503] chan_sip.c: ---------- END SIP HISTORY for '630f809f4145b35d72a542370d330980@87.253.148.89' [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK0ed54b95;received=87.253.148.89;rport=5060 (92) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 2: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=as05f16626 (49) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 8: Supported: replaces (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 9: Contact: (39) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 263 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: o=root 15794 15794 IN IP4 92.65.24.219 (38) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: c=IN IP4 92.65.24.219 (21) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: m=audio 12652 RTP/AVP 8 3 101 (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=ptime:20 (10) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=sendrecv (10) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = Found Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Acked pending invite 102 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Stopping retransmission on '1eead9035e96d8f1476c36891de254d5@87.253.148.89' of Request 102: Match Found [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: SIP response 200 to standard invite [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP/test2-0819d1f0 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: We're settling with these formats: 0xa (gsm|alaw) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: We have an owner, now see if we need to change this call [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Updating call counter for outgoing call [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: build_route: Contact hop: [Apr 17 09:42:48] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/test2-0819d1f0 [Apr 17 09:42:48] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/test2 [Apr 17 09:42:48] DEBUG[32571] rtp.c: Setting early bridge SDP of 'SIP/alt001-test-08199278' with that of 'SIP/test2-0819d1f0' [Apr 17 09:42:48] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/alt001-test-08199278 [Apr 17 09:42:48] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/alt001-test [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: SIP answering channel: SIP/alt001-test-08199278 [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Setting framing from config on incoming call [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: -- Done with adding codecs to SDP [Apr 17 09:42:48] DEBUG[32571] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Deferring reinvite on SIP '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' - It's audio will be redirected to IP 92.65.24.219 [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Sending reinvite on SIP '1eead9035e96d8f1476c36891de254d5@87.253.148.89' - It's audio soon redirected to IP 217.114.103.72 [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: ** Our capability: 0xa (gsm|alaw) Video flag: True [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: -- Done with adding codecs to SDP [Apr 17 09:42:48] DEBUG[32571] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Done building SDP. Settling with this capability: 0xa (gsm|alaw) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Initializing already initialized SIP dialog 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (presumably reinvite) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 0: INVITE sip:31880016499@92.65.24.219 SIP/2.0 (43) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK3f7b182f;rport (64) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 2: Route: (37) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 3: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 4: To: ;tag=as05f16626 (49) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 5: Contact: (40) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 6: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 7: CSeq: 103 INVITE (16) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 8: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 14: Content-Length: 267 (19) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Header 15: (0) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: o=root 31837 31838 IN IP4 217.114.103.72 (40) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: s=session (9) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 3 101 (29) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=ptime:20 (10) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: Line: a=sendrecv (10) [Apr 17 09:42:48] DEBUG[32571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:48] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - test2-0819d1f0 [Apr 17 09:42:48] DEBUG[32501] chan_sip.c: Checking device state for peer test2-0819d1f0 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 0: ACK sip:31880016499@87.253.148.89 SIP/2.0 (41) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.249.114.95;branch=z9hG4bK48ae.993b081.2 (59) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 2: Via: SIP/2.0/UDP 217.114.103.72:8066;branch=z9hG4bK60571e12;rport=8066 (70) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 3: From: " " ;tag=as16153a28 (62) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 4: To: ;tag=as4ba8aefb (59) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 5: Contact: (46) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 6: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 7: CSeq: 102 ACK (13) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 8: User-Agent: SpeakUp PSTN (24) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 9: Max-Forwards: 69 (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 10: Content-Length: 0 (17) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 11: P-hint: RR-enforced (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = No match Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Stopping retransmission on '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' of Response 102: Match Found [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Sending pending reinvite on '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: ** Our capability: 0xa (gsm|alaw) Video flag: True [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: -- Done with adding codecs to SDP [Apr 17 09:42:48] DEBUG[32503] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Done building SDP. Settling with this capability: 0xa (gsm|alaw) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Initializing already initialized SIP dialog 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (presumably reinvite) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 0: INVITE sip:31152563045@217.114.103.72:8066 SIP/2.0 (50) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK052968c4;rport (64) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 2: Route: (48) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 3: From: ;tag=as4ba8aefb (61) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 4: To: " " ;tag=as16153a28 (60) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 5: Contact: (40) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 6: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 7: CSeq: 102 INVITE (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 8: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 14: Content-Length: 263 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 15: (0) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: o=root 31837 31838 IN IP4 92.65.24.219 (38) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: c=IN IP4 92.65.24.219 (21) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: m=audio 12652 RTP/AVP 8 3 101 (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=ptime:20 (10) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=sendrecv (10) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK052968c4;rport=5060 (69) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: o=root 23936 23937 IN IP4 217.114.103.72 (40) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = No match Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Acked pending invite 102 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Stopping retransmission on '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' of Request 102: Match Found [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: SIP response 200 to RE-invite on outgoing call 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP/alt001-test-08199278 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: We're settling with these formats: 0xa (gsm|alaw) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: We have an owner, now see if we need to change this call [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Updating call counter for incoming call [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Strange... The other side of the bridge does not have a udptl struct [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP/alt001-test-08199278 [Apr 17 09:42:48] DEBUG[32571] rtp.c: Ooh, format changed from unknown to alaw [Apr 17 09:42:48] DEBUG[32571] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK3f7b182f;received=87.253.148.89;rport=5060 (92) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 2: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=as05f16626 (49) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 8: Supported: replaces (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 9: Contact: (39) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 10: Content-Length: 0 (17) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 11: (0) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = Found Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: *** SIP TIMER: Cancelling retransmission #39 - INVITE (got response) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1eead9035e96d8f1476c36891de254d5@87.253.148.89' Request 103: Found [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: SIP response 100 to RE-invite on outgoing call 1eead9035e96d8f1476c36891de254d5@87.253.148.89 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK3f7b182f;received=87.253.148.89;rport=5060 (92) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 2: From: " " ;tag=as0053a0c1 (56) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 3: To: ;tag=as05f16626 (49) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 8: Supported: replaces (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 9: Contact: (39) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 263 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: o=root 15794 15795 IN IP4 92.65.24.219 (38) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: c=IN IP4 92.65.24.219 (21) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: m=audio 12652 RTP/AVP 8 3 101 (29) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=ptime:20 (10) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Line: a=sendrecv (10) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: = Found Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Acked pending invite 103 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Stopping retransmission on '1eead9035e96d8f1476c36891de254d5@87.253.148.89' of Request 103: Match Found [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: SIP response 200 to RE-invite on outgoing call 1eead9035e96d8f1476c36891de254d5@87.253.148.89 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP/test2-0819d1f0 [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: We're settling with these formats: 0xa (gsm|alaw) [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: We have an owner, now see if we need to change this call [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Updating call counter for outgoing call [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: Strange... The other side of the bridge does not have a udptl struct [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP [Apr 17 09:42:48] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel SIP/test2-0819d1f0 [Apr 17 09:42:48] DEBUG[32571] rtp.c: Ooh, format changed from unknown to alaw [Apr 17 09:42:48] DEBUG[32571] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Apr 17 09:42:48] DEBUG[32501] devicestate.c: Changing state for SIP/test2-0819d1f0 - state 4 (Invalid) [Apr 17 09:42:48] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - test2 [Apr 17 09:42:48] DEBUG[32501] chan_sip.c: Checking device state for peer test2 [Apr 17 09:42:48] DEBUG[32501] devicestate.c: Changing state for SIP/test2 - state 1 (Not in use) [Apr 17 09:42:48] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - alt001-test-08199278 [Apr 17 09:42:48] DEBUG[32501] chan_sip.c: Checking device state for peer alt001-test-08199278 [Apr 17 09:42:48] DEBUG[32501] devicestate.c: Changing state for SIP/alt001-test-08199278 - state 4 (Invalid) [Apr 17 09:42:48] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - alt001-test [Apr 17 09:42:48] DEBUG[32501] chan_sip.c: Checking device state for peer alt001-test [Apr 17 09:42:48] DEBUG[32501] devicestate.c: Changing state for SIP/alt001-test - state 4 (Invalid) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 0: BYE sip:31152563045@87.253.148.89 SIP/2.0 (41) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 92.65.24.219:5060;branch=z9hG4bK7d071405;rport (63) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 2: Route: (38) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 3: From: ;tag=as05f16626 (51) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 4: To: " " ;tag=as0053a0c1 (54) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 5: Call-ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 (55) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 6: CSeq: 102 BYE (13) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 9: Content-Length: 0 (17) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 10: (0) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: = Found Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1eead9035e96d8f1476c36891de254d5@87.253.148.89 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Received bye, issuing owner hangup [Apr 17 09:42:55] DEBUG[32571] rtp.c: Oooh, got a hangup [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Sending reinvite on SIP '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' - It's audio soon redirected to IP 87.253.148.89 [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: ** Our capability: 0xa (gsm|alaw) Video flag: True [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: -- Done with adding codecs to SDP [Apr 17 09:42:55] DEBUG[32571] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Done building SDP. Settling with this capability: 0xa (gsm|alaw) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Initializing already initialized SIP dialog 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (presumably reinvite) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 0: INVITE sip:31152563045@217.114.103.72:8066 SIP/2.0 (50) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport (64) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 2: Route: (48) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 3: From: ;tag=as4ba8aefb (61) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 4: To: " " ;tag=as16153a28 (60) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 5: Contact: (40) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 6: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 7: CSeq: 103 INVITE (16) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 8: User-Agent: Asterisk 1.4 on identificeer.bokxing.nl (51) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 11: Supported: replaces (19) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 12: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 14: Content-Length: 265 (19) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Header 15: (0) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: o=root 31837 31839 IN IP4 87.253.148.89 (39) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: s=session (9) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: c=IN IP4 87.253.148.89 (22) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: m=audio 13366 RTP/AVP 8 3 101 (29) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=ptime:20 (10) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Line: a=sendrecv (10) [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:55] DEBUG[32571] channel.c: Returning from native bridge, channels: SIP/alt001-test-08199278, SIP/test2-0819d1f0 [Apr 17 09:42:55] DEBUG[32571] channel.c: Hanging up channel 'SIP/test2-0819d1f0' [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Hangup call SIP/test2-0819d1f0, SIP callid 1eead9035e96d8f1476c36891de254d5@87.253.148.89) [Apr 17 09:42:55] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/test2-0819d1f0 [Apr 17 09:42:55] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/test2 [Apr 17 09:42:55] DEBUG[32571] rtp.c: Channel '' has no RTP, not doing anything [Apr 17 09:42:55] DEBUG[32571] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Apr 17 09:42:55] DEBUG[32571] pbx.c: Spawn extension (dial-customer,31880016499,3) exited non-zero on 'SIP/alt001-test-08199278' [Apr 17 09:42:55] DEBUG[32571] channel.c: Soft-Hanging up channel 'SIP/alt001-test-08199278' [Apr 17 09:42:55] DEBUG[32571] pbx.c: Function result is '09:42:55' [Apr 17 09:42:55] DEBUG[32571] pbx.c: Function result is '31152563045' [Apr 17 09:42:55] DEBUG[32571] pbx.c: Launching 'Verbose' [Apr 17 09:42:55] DEBUG[32571] pbx.c: Launching 'Log' [Apr 17 09:42:55] DEBUG[32571] pbx.c: Launching 'Hangup' [Apr 17 09:42:55] DEBUG[32571] pbx.c: Spawn extension (dial-customer,h,3) exited non-zero on 'SIP/alt001-test-08199278' [Apr 17 09:42:55] DEBUG[32571] channel.c: Hanging up channel 'SIP/alt001-test-08199278' [Apr 17 09:42:55] DEBUG[32571] chan_sip.c: Hangup call SIP/alt001-test-08199278, SIP callid 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72) [Apr 17 09:42:55] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/alt001-test-08199278 [Apr 17 09:42:55] DEBUG[32571] devicestate.c: Notification of state change to be queued on device/channel SIP/alt001-test [Apr 17 09:42:55] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - test2-0819d1f0 [Apr 17 09:42:55] DEBUG[32501] chan_sip.c: Checking device state for peer test2-0819d1f0 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: = No match Their Call ID: 1eead9035e96d8f1476c36891de254d5@87.253.148.89 Their Tag as05f16626 Our tag: as0053a0c1 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Acked pending invite 103 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Stopping retransmission on '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' of Request 103: Match Found [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: SIP response 200 to standard invite [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: T38 state changed to 0 on channel [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: We're settling with these formats: 0xa (gsm|alaw) [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: Updating call counter for incoming call [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: build_route: Contact hop: [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: ---------- SIP HISTORY for '1eead9035e96d8f1476c36891de254d5@87.253.148.89' [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: * SIP Call [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: 001. Hangup Cause Normal Clearing [Apr 17 09:42:55] DEBUG[32503] chan_sip.c: ---------- END SIP HISTORY for '1eead9035e96d8f1476c36891de254d5@87.253.148.89' [Apr 17 09:42:56] DEBUG[32501] devicestate.c: Changing state for SIP/test2-0819d1f0 - state 4 (Invalid) [Apr 17 09:42:56] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - test2 [Apr 17 09:42:56] DEBUG[32501] chan_sip.c: Checking device state for peer test2 [Apr 17 09:42:56] DEBUG[32501] devicestate.c: Changing state for SIP/test2 - state 1 (Not in use) [Apr 17 09:42:56] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - alt001-test-08199278 [Apr 17 09:42:56] DEBUG[32501] chan_sip.c: Checking device state for peer alt001-test-08199278 [Apr 17 09:42:56] DEBUG[32501] devicestate.c: Changing state for SIP/alt001-test-08199278 - state 4 (Invalid) [Apr 17 09:42:56] DEBUG[32501] devicestate.c: No provider found, checking channel drivers for SIP - alt001-test [Apr 17 09:42:56] DEBUG[32501] chan_sip.c: Checking device state for peer alt001-test [Apr 17 09:42:56] DEBUG[32501] devicestate.c: Changing state for SIP/alt001-test - state 4 (Invalid) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (1) BYE - 8 [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #45)) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (2) BYE - 8 [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #45)) [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (3) BYE - 8 [Apr 17 09:42:56] DEBUG[32503] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #45)) [Apr 17 09:42:57] DEBUG[32503] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (4) BYE - 8 [Apr 17 09:42:57] DEBUG[32503] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #45)) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:42:58] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:42:59] DEBUG[32503] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (5) BYE - 8 [Apr 17 09:42:59] DEBUG[32503] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #45)) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: SIP TIMER: Rescheduling retransmission #45 (6) BYE - 8 [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #45)) [Apr 17 09:43:02] DEBUG[32503] chan_sip.c: Re-scheduled destruction of SIP call 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: = Found Their Call ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 Their Tag as16153a28 Our tag: as4ba8aefb [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: ---------- SIP HISTORY for '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: * SIP Call [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 001. Hangup Cause Normal Clearing [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 002. CancelDestroy [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 003. Ignore Ignoring this retransmit [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 004. ReTx 200 BYE sip:31152563045@217.114.103.72:8066 SIP/2.0 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 005. ReTx 400 BYE sip:31152563045@217.114.103.72:8066 SIP/2.0 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 006. ReTx 800 BYE sip:31152563045@217.114.103.72:8066 SIP/2.0 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 007. ReTx 1600 BYE sip:31152563045@217.114.103.72:8066 SIP/2.0 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 008. Ignore Ignoring this retransmit [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 009. ReTx 3200 BYE sip:31152563045@217.114.103.72:8066 SIP/2.0 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 010. Ignore Ignoring this retransmit [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 011. ReTx 4000 BYE sip:31152563045@217.114.103.72:8066 SIP/2.0 [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 012. ReliableXmit timeout [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 013. Ignore Ignoring this retransmit [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 014. MaxRetries (Non-critical) [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: 015. ByeFailure Remote peer doesn't respond to bye. Destroying call anyway. [Apr 17 09:43:06] DEBUG[32503] chan_sip.c: ---------- END SIP HISTORY for '6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72' [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:43:10] DEBUG[32503] chan_sip.c: Invalid SIP message - rejected , no callid, len 769 [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 1: Via: SIP/2.0/UDP 87.253.148.89:5060;branch=z9hG4bK77f08770;rport=5060 (69) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 2: From: ;tag=as4ba8aefb (61) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 3: To: " " ;tag=as16153a28 (60) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 4: Call-ID: 6834ff3761d241da3d0d3a5b0b30dea7@217.114.103.72 (56) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 5: CSeq: 103 INVITE (16) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 6: User-Agent: SpeakUp PSTN (24) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 9: Contact: (46) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 11: Content-Length: 267 (19) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Header 12: (0) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: v=0 (3) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: o=root 23936 23938 IN IP4 217.114.103.72 (40) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: s=session (9) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: c=IN IP4 217.114.103.72 (23) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: t=0 0 (5) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: m=audio 18680 RTP/AVP 8 0 3 101 (31) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 17 09:43:14] DEBUG[32503] chan_sip.c: Invalid SIP message - rejected , no callid, len 769