[Jul 7 15:59:41] VERBOSE[1731] logger.c: [Jul 7 15:59:41] Asterisk Event Logger restarted [Jul 7 15:59:41] VERBOSE[1731] logger.c: [Jul 7 15:59:41] Asterisk Queue Logger restarted [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] <--- SIP read from 192.168.0.202:5060 ---> INVITE sip:360@isoemo.com SIP/2.0 Route: Supported: 100rel,timer User-Agent: OxO_GW_700/013.001 Session-Expires: 43200 P-Asserted-Identity: "Valery Komarov" To: sip:360@isoemo.com From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d Contact: "Valery Komarov" Content-Type: application/sdp Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670236 INVITE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKf32cb66cfdc9eddabb0d75275a826568 Max-Forwards: 70 Content-Length: 251 v=0 o=default 1278503993 1278503993 IN IP4 192.168.0.202 s=- c=IN IP4 192.168.0.202 t=0 0 m=audio 32000 RTP/AVP 18 106 4 8 0 a=sendrecv a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-15 a=fmtp:4 annexa=no a=maxptime:90 <-------------> [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 0: INVITE sip:360@isoemo.com SIP/2.0 (33) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 1: Route: (28) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 2: Supported: 100rel,timer (23) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 4: Session-Expires: 43200 (22) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 5: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 6: To: sip:360@isoemo.com (22) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 7: From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d (83) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 8: Contact: "Valery Komarov" (63) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 10: Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 (55) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 11: CSeq: 517670236 INVITE (22) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKf32cb66cfdc9eddabb0d75275a826568 (83) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 13: Max-Forwards: 70 (16) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 14: Content-Length: 251 (19) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 15: (0) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: v=0 (3) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: o=default 1278503993 1278503993 IN IP4 192.168.0.202 (52) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: s=- (3) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: m=audio 32000 RTP/AVP 18 106 4 8 0 (34) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: a=fmtp:4 annexa=no (18) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Line: a=maxptime:90 (13) [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] --- (15 headers 12 lines) --- [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: = No match Their Call ID: 218833074417721-87152881325901@192.168.5.13 Their Tag 102725707 Our tag: as1b7aa42e [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: = No match Their Call ID: 23319369128661-235514975462@192.168.5.13 Their Tag 1273021861 Our tag: as7d457728 [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Setting NAT on VRTP to Off [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Allocating new SIP dialog for a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 - INVITE (With RTP) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Begin: parsing SIP "Supported: 100rel,timer" [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Found SIP option: -100rel- [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Matched SIP option: 100rel [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Found SIP option: -timer- [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Matched SIP option: timer [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Sending to 192.168.0.202 : 5060 (no NAT) [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Using INVITE request as basis request - a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found peer 'oxo' [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Setting NAT on VRTP to Off [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing session-level SDP o=default 1278503993 1278503993 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found RTP audio format 18 [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found RTP audio format 106 [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found RTP audio format 4 [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found RTP audio format 8 [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found RTP audio format 0 [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Found audio description format telephone-event for ID 106 [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=fmtp:4 annexa=no... UNSUPPORTED. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=maxptime:90... UNSUPPORTED. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: T38 state changed to 0 on channel [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Peer audio RTP is at port 192.168.0.202:32000 [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Checking SIP call limits for device [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Updating call counter for incoming call [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] Looking for 360 in oxo (domain isoemo.com) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: This channel will not be able to handle video. [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: build_route: Contact hop: "Valery Komarov" [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] list_route: hop: [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: SIP/oxo-00000000: New call is still down.... Trying... [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKf32cb66cfdc9eddabb0d75275a826568;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d To: sip:360@isoemo.com Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670236 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 15:59:43] DEBUG[1750] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:59:43] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:59:43] DEBUG[1733] chan_sip.c: Checking device state for peer oxo [Jul 7 15:59:43] DEBUG[1733] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:59:43] DEBUG[1739] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:43] DEBUG[1765] pbx.c: Launching 'Dial' [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] -- Executing [360@oxo:1] Dial("SIP/oxo-00000000", "SIP/360|60|t") in new stack [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: This channel will not be able to handle video. [Jul 7 15:59:43] DEBUG[1765] rtp.c: Seeded SDP of 'SIP/360-00000001' with that of 'SIP/oxo-00000000' [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable DIALEDTIME. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable ANSWEREDTIME. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable DIALEDPEERNAME. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable DIALSTATUS. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable SIPCALLID. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable SIPUSERAGENT. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable SIPDOMAIN. [Jul 7 15:59:43] DEBUG[1765] channel.c: Not copying variable SIPURI. [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Outgoing Call for 360 [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] Audio is at 192.168.1.18 port 12734 [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] Adding codec 0x8 (alaw) to SDP [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] Adding codec 0x4 (ulaw) to SDP [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 0: INVITE sip:360@192.168.5.13:5060 SIP/2.0 (40) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport (63) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as5e4ab5c0 (60) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 3: To: (31) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 4: Contact: (31) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 5: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 (54) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 7: User-Agent: Asterisk (20) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 9: Remote-Party-ID: "Valery Komarov" ;privacy=off;screen=no (78) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 10: Date: Wed, 07 Jul 2010 11:59:43 GMT (35) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 12: Supported: replaces (19) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 14: Content-Length: 235 (19) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Header 15: (0) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: v=0 (3) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: o=root 1731 1731 IN IP4 192.168.1.18 (36) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: s=session (9) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: c=IN IP4 192.168.1.18 (21) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: m=audio 12734 RTP/AVP 8 0 106 (29) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: a=fmtp:106 0-16 (15) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] Reliably Transmitting (no NAT) to 192.168.5.13:5060: INVITE sip:360@192.168.5.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport From: "Valery Komarov" ;tag=as5e4ab5c0 To: Contact: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "Valery Komarov" ;privacy=off;screen=no Date: Wed, 07 Jul 2010 11:59:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=root 1731 1731 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 12734 RTP/AVP 8 0 106 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=ptime:20 a=sendrecv --- [Jul 7 15:59:43] DEBUG[1765] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] -- Called 360 [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport From: "Valery Komarov" ;tag=as5e4ab5c0 To: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport (63) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as5e4ab5c0 (60) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 3: To: (31) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 4: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 (54) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 6: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 7: Content-Length: 0 (17) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 8: (0) [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] --- (8 headers 0 lines) --- [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: = Found Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag Our tag: as5e4ab5c0 [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: *** SIP TIMER: Cancelling retransmission #22 - INVITE (got response) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18' Request 102: Found [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: SIP response 100 to standard invite [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport From: "Valery Komarov" ;tag=as5e4ab5c0 To: ;tag=14636877 Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 102 INVITE Contact: Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport (63) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as5e4ab5c0 (60) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 3: To: ;tag=14636877 (44) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 4: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 (54) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 6: Contact: (36) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 7: Server: Voip Phone 1.0 (22) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 8: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: Header 10: (0) [Jul 7 15:59:43] VERBOSE[1750] logger.c: [Jul 7 15:59:43] --- (10 headers 0 lines) --- [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: = Found Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag Our tag: as5e4ab5c0 [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18' Request 102: Found [Jul 7 15:59:43] DEBUG[1750] chan_sip.c: SIP response 180 to standard invite [Jul 7 15:59:43] DEBUG[1750] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 15:59:43] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 15:59:43] DEBUG[1733] chan_sip.c: Checking device state for peer 360 [Jul 7 15:59:43] DEBUG[1733] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 15:59:43] DEBUG[1739] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] -- SIP/360-00000001 is ringing [Jul 7 15:59:43] DEBUG[1765] rtp.c: Setting early bridge SDP of 'SIP/oxo-00000000' with that of 'SIP/360-00000001' [Jul 7 15:59:43] VERBOSE[1765] logger.c: [Jul 7 15:59:43] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKf32cb66cfdc9eddabb0d75275a826568;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d To: sip:360@isoemo.com;tag=as078f6861 Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670236 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport From: "Valery Komarov" ;tag=as5e4ab5c0 To: ;tag=14636877 Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 102 INVITE Contact: Supported: 100rel, replaces, timer Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 237 v=0 o=360 27935215 51228425 IN IP4 192.168.5.13 s=A conversation c=IN IP4 192.168.5.13 t=0 0 m=audio 10236 RTP/AVP 8 0 106 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-15 a=sendrecv <-------------> [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5e721fbd;rport (63) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as5e4ab5c0 (60) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 3: To: ;tag=14636877 (44) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 4: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 (54) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 6: Contact: (36) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 7: Supported: 100rel, replaces, timer (34) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 8: Server: Voip Phone 1.0 (22) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 9: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 11: Content-Length: 237 (19) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 12: (0) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: v=0 (3) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: o=360 27935215 51228425 IN IP4 192.168.5.13 (43) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: s=A conversation (16) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: c=IN IP4 192.168.5.13 (21) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: m=audio 10236 RTP/AVP 8 0 106 (29) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] --- (12 headers 11 lines) --- [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = Found Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag 14636877 Our tag: as5e4ab5c0 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Acked pending invite 102 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Stopping retransmission on '7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18' of Request 102: Match Found [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: SIP response 200 to standard invite [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP o=360 27935215 51228425 IN IP4 192.168.5.13... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.13... OK. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found RTP audio format 8 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found RTP audio format 0 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found RTP audio format 106 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found audio description format PCMA for ID 8 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found audio description format PCMU for ID 0 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found audio description format telephone-event for ID 106 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: T38 state changed to 0 on channel SIP/360-00000001 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Peer audio RTP is at port 192.168.5.13:10236 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: build_route: Contact hop: [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] list_route: hop: [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Strict routing enforced for session 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] set_destination: Parsing for address/port to send to [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] set_destination: set destination to 192.168.5.13, port 5060 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Transmitting (no NAT) to 192.168.5.13:5060: ACK sip:360@192.168.5.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK6eb33287;rport From: "Valery Komarov" ;tag=as5e4ab5c0 To: ;tag=14636877 Contact: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 102 ACK User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "Valery Komarov" ;privacy=off;screen=no Content-Length: 0 --- [Jul 7 15:59:44] DEBUG[1765] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 15:59:44] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 15:59:44] VERBOSE[1765] logger.c: [Jul 7 15:59:44] -- SIP/360-00000001 answered SIP/oxo-00000000 [Jul 7 15:59:44] DEBUG[1733] chan_sip.c: Checking device state for peer 360 [Jul 7 15:59:44] DEBUG[1733] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 15:59:44] DEBUG[1739] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:44] DEBUG[1765] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: SIP answering channel: SIP/oxo-00000000 [Jul 7 15:59:44] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:59:44] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:44] DEBUG[1733] chan_sip.c: Checking device state for peer oxo [Jul 7 15:59:44] DEBUG[1733] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: Setting framing from config on incoming call [Jul 7 15:59:44] DEBUG[1739] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 7 15:59:44] VERBOSE[1765] logger.c: [Jul 7 15:59:44] Audio is at 192.168.1.18 port 17098 [Jul 7 15:59:44] VERBOSE[1765] logger.c: [Jul 7 15:59:44] Adding codec 0x8 (alaw) to SDP [Jul 7 15:59:44] VERBOSE[1765] logger.c: [Jul 7 15:59:44] Adding codec 0x4 (ulaw) to SDP [Jul 7 15:59:44] VERBOSE[1765] logger.c: [Jul 7 15:59:44] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 15:59:44] VERBOSE[1765] logger.c: [Jul 7 15:59:44] <--- Reliably Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKf32cb66cfdc9eddabb0d75275a826568;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d To: sip:360@isoemo.com;tag=as078f6861 Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670236 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 1731 1731 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 17098 RTP/AVP 8 0 106 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=ptime:20 a=sendrecv <------------> [Jul 7 15:59:44] DEBUG[1765] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:59:44] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:44] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:44] DEBUG[1765] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 15:59:44] DEBUG[1765] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] <--- SIP read from 192.168.0.202:5060 ---> ACK sip:360@192.168.1.18 SIP/2.0 Route: Contact: "Valery Komarov" User-Agent: OxO_GW_700/013.001 To: sip:360@isoemo.com;tag=as078f6861 From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670236 ACK Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKd28507c3f7aa5239ff58e4167b862071 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 0: ACK sip:360@192.168.1.18 SIP/2.0 (32) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 1: Route: (28) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 2: Contact: "Valery Komarov" (49) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 4: To: sip:360@isoemo.com;tag=as078f6861 (37) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 5: From: "Valery Komarov" ;tag=2eade5bd4fc1015d4d4aee3f8485445d (83) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 6: Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 (55) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 7: CSeq: 517670236 ACK (19) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKd28507c3f7aa5239ff58e4167b862071 (83) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 10: Content-Length: 0 (17) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 11: (0) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] --- (11 headers 0 lines) --- [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = No match Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag 14636877 Our tag: as5e4ab5c0 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = Found Their Call ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 Their Tag 2eade5bd4fc1015d4d4aee3f8485445d Our tag: as078f6861 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Stopping retransmission on 'a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202' of Response 517670236: Match Found [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] <--- SIP read from 192.168.0.202:5060 ---> INVITE sip:360@192.168.1.18 SIP/2.0 Route: Supported: 100rel,timer User-Agent: OxO_GW_700/013.001 Session-Expires: 43200;refresher=uac P-Asserted-Identity: "Valery Komarov" Contact: "Valery Komarov" Content-Type: application/sdp To: sip:360@isoemo.com;tag=as078f6861 From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670237 INVITE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKcb7abbad2f0ab6c58ea74cbeba01735a Max-Forwards: 70 Content-Length: 215 v=0 o=default 1278503993 1278503994 IN IP4 192.168.0.202 s=- c=IN IP4 192.168.0.202 t=0 0 m=audio 32000 RTP/AVP 8 106 a=sendrecv a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-15 a=ptime:20 a=maxptime:90 <-------------> [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 0: INVITE sip:360@192.168.1.18 SIP/2.0 (35) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 1: Route: (28) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 2: Supported: 100rel,timer (23) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 4: Session-Expires: 43200;refresher=uac (36) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 5: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 6: Contact: "Valery Komarov" (63) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 7: Content-Type: application/sdp (29) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 8: To: sip:360@isoemo.com;tag=as078f6861 (37) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 9: From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d (64) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 10: Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 (55) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 11: CSeq: 517670237 INVITE (22) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKcb7abbad2f0ab6c58ea74cbeba01735a (83) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 13: Max-Forwards: 70 (16) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 14: Content-Length: 215 (19) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 15: (0) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: v=0 (3) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: o=default 1278503993 1278503994 IN IP4 192.168.0.202 (52) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: s=- (3) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: m=audio 32000 RTP/AVP 8 106 (27) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Line: a=maxptime:90 (13) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] --- (15 headers 11 lines) --- [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = No match Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag 14636877 Our tag: as5e4ab5c0 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = Found Their Call ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 Their Tag 2eade5bd4fc1015d4d4aee3f8485445d Our tag: as078f6861 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Sending to 192.168.0.202 : 5060 (NAT) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP o=default 1278503993 1278503994 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found RTP audio format 8 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found RTP audio format 106 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Found audio description format telephone-event for ID 106 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=maxptime:90... UNSUPPORTED. [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: T38 state changed to 0 on channel SIP/oxo-00000000 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Peer audio RTP is at port 192.168.0.202:32000 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Got a SIP re-invite for call a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: SIP/oxo-00000000: This call is UP.... [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKcb7abbad2f0ab6c58ea74cbeba01735a;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d To: sip:360@isoemo.com;tag=as078f6861 Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670237 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Setting framing from config on incoming call [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Audio is at 192.168.1.18 port 17098 [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Adding codec 0x8 (alaw) to SDP [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] <--- Reliably Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKcb7abbad2f0ab6c58ea74cbeba01735a;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d To: sip:360@isoemo.com;tag=as078f6861 Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670237 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 1731 1732 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 17098 RTP/AVP 8 106 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=ptime:20 a=sendrecv <------------> [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:59:44] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] <--- SIP read from 192.168.0.202:5060 ---> ACK sip:360@192.168.1.18 SIP/2.0 Route: Contact: "Valery Komarov" User-Agent: OxO_GW_700/013.001 To: sip:360@isoemo.com;tag=as078f6861 From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670237 ACK Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK21f4cde2d9287664ceefbdc46e3ad3ca Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 0: ACK sip:360@192.168.1.18 SIP/2.0 (32) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 1: Route: (28) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 2: Contact: "Valery Komarov" (49) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 4: To: sip:360@isoemo.com;tag=as078f6861 (37) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 5: From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d (64) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 6: Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 (55) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 7: CSeq: 517670237 ACK (19) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK21f4cde2d9287664ceefbdc46e3ad3ca (83) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 10: Content-Length: 0 (17) [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Header 11: (0) [Jul 7 15:59:44] VERBOSE[1750] logger.c: [Jul 7 15:59:44] --- (11 headers 0 lines) --- [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = No match Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag 14636877 Our tag: as5e4ab5c0 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: = Found Their Call ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 Their Tag 2eade5bd4fc1015d4d4aee3f8485445d Our tag: as078f6861 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 [Jul 7 15:59:44] DEBUG[1750] chan_sip.c: Stopping retransmission on 'a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202' of Response 517670237: Match Found [Jul 7 15:59:44] DEBUG[1765] rtp.c: Got RTCP report of 36 bytes [Jul 7 15:59:44] DEBUG[1765] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:44] DEBUG[1765] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 15:59:44] DEBUG[1765] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF begin '*' received on SIP/360-00000001 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF begin passthrough '*' on SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] channel.c: Got DTMF begin on channel (SIP/360-00000001) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF end '*' received on SIP/360-00000001, duration 110 ms [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF end accepted with begin '*' on SIP/360-00000001 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF end passthrough '*' on SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] channel.c: Got DTMF end on channel (SIP/360-00000001) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] res_features.c: Feature interpret: chan=SIP/oxo-00000000, peer=SIP/360-00000001, code=*, sense=2, features=2, dynamic=# [Jul 7 15:59:45] DEBUG[1765] res_features.c: Set time limit to 1000 [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF begin '*' received on SIP/360-00000001 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF begin passthrough '*' on SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] channel.c: Got DTMF begin on channel (SIP/360-00000001) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF end '*' received on SIP/360-00000001, duration 110 ms [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF end accepted with begin '*' on SIP/360-00000001 [Jul 7 15:59:45] DTMF[1765] channel.c: DTMF end passthrough '*' on SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] channel.c: Got DTMF end on channel (SIP/360-00000001) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] DEBUG[1765] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:59:45] DEBUG[1765] res_features.c: Feature interpret: chan=SIP/oxo-00000000, peer=SIP/360-00000001, code=**, sense=2, features=2, dynamic=# [Jul 7 15:59:45] DEBUG[1765] res_features.c: Feature detected: fname=Attended Transfer sname=atxfer exten=** [Jul 7 15:59:45] DEBUG[1765] res_features.c: Executing Attended Transfer SIP/oxo-00000000, SIP/360-00000001 (sense=2) [Jul 7 15:59:45] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:45] VERBOSE[1765] logger.c: [Jul 7 15:59:45] -- Started music on hold, class 'default', on SIP/oxo-00000000 [Jul 7 15:59:45] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:45] DEBUG[1765] channel.c: Set channel SIP/360-00000001 to write format slin [Jul 7 15:59:45] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:45] VERBOSE[1765] logger.c: [Jul 7 15:59:45] -- Playing 'pbx-transfer' (language 'ru') [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:45] DEBUG[1766] channel.c: Set channel SIP/oxo-00000000 to write format slin [Jul 7 15:59:45] DEBUG[1766] res_musiconhold.c: SIP/oxo-00000000 Opened file 9 '/var/lib/asterisk/moh/18-Chet Atkins-On My Way to Cannan's Land' [Jul 7 15:59:45] WARNING[1766] mp3/interface.c: Junk at the beginning of frame 49443303 [Jul 7 15:59:45] DEBUG[1766] rtp.c: Difference is 944, ms is 138 [Jul 7 15:59:45] DEBUG[1765] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF begin '1' received on SIP/360-00000001 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF begin ignored '1' on SIP/360-00000001 [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF end '1' received on SIP/360-00000001, duration 110 ms [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF end passthrough '1' on SIP/360-00000001 [Jul 7 15:59:46] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: Sending dtmf: 52 (4), at 192.168.5.13 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF begin '4' received on SIP/360-00000001 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF begin ignored '4' on SIP/360-00000001 [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: Sending dtmf: 52 (4), at 192.168.5.13 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF end '4' received on SIP/360-00000001, duration 110 ms [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF end passthrough '4' on SIP/360-00000001 [Jul 7 15:59:46] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:46] DEBUG[1765] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF begin '1' received on SIP/360-00000001 [Jul 7 15:59:46] DTMF[1765] channel.c: DTMF begin ignored '1' on SIP/360-00000001 [Jul 7 15:59:46] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:59:47] DTMF[1765] channel.c: DTMF end '1' received on SIP/360-00000001, duration 110 ms [Jul 7 15:59:47] DTMF[1765] channel.c: DTMF end passthrough '1' on SIP/360-00000001 [Jul 7 15:59:47] DEBUG[1765] channel.c: Not copying variable BRIDGEPEER. [Jul 7 15:59:47] DEBUG[1765] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 15:59:47] DEBUG[1765] channel.c: Not copying variable SIPCALLID. [Jul 7 15:59:47] DEBUG[1765] channel.c: Driver for channel 'SIP/360-00000001' does not support indication 3, emulating it [Jul 7 15:59:47] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:47] DEBUG[1767] pbx.c: Launching 'Dial' [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] -- Executing [141@internal:1] Dial("Local/141@internal-bae5,2", "SIP/141@oxo|60|t") in new stack [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: This channel will not be able to handle video. [Jul 7 15:59:47] DEBUG[1767] rtp.c: Channel 'Local/141@internal-bae5,2' has no RTP, not doing anything [Jul 7 15:59:47] DEBUG[1767] channel.c: Not copying variable DIALEDTIME. [Jul 7 15:59:47] DEBUG[1767] channel.c: Not copying variable ANSWEREDTIME. [Jul 7 15:59:47] DEBUG[1767] channel.c: Not copying variable DIALEDPEERNAME. [Jul 7 15:59:47] DEBUG[1767] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 15:59:47] DEBUG[1767] channel.c: Not copying variable DIALSTATUS. [Jul 7 15:59:47] DEBUG[1767] channel.c: Not copying variable TRANSFERERNAME. [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Outgoing Call for 141 [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] Audio is at 192.168.1.18 port 17064 [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] Adding codec 0x8 (alaw) to SDP [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] Adding codec 0x4 (ulaw) to SDP [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 0: INVITE sip:141@192.168.0.202 SIP/2.0 (36) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK3fa17140;rport (63) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 2: From: "360" ;tag=as58f1335b (49) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 3: To: (27) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 4: Contact: (31) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 5: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 (54) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 7: User-Agent: Asterisk (20) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 9: Remote-Party-ID: "360" ;privacy=off;screen=no (67) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 10: Date: Wed, 07 Jul 2010 11:59:47 GMT (35) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 12: Supported: replaces (19) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 14: Content-Length: 235 (19) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Header 15: (0) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: v=0 (3) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: o=root 1731 1731 IN IP4 192.168.1.18 (36) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: s=session (9) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: c=IN IP4 192.168.1.18 (21) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: m=audio 17064 RTP/AVP 8 0 101 (29) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] Reliably Transmitting (no NAT) to 192.168.0.202:5060: INVITE sip:141@192.168.0.202 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK3fa17140;rport From: "360" ;tag=as58f1335b To: Contact: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "360" ;privacy=off;screen=no Date: Wed, 07 Jul 2010 11:59:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=root 1731 1731 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 17064 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=ptime:20 a=sendrecv --- [Jul 7 15:59:47] DEBUG[1767] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] -- Called 141@oxo [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:59:47] DEBUG[1765] rtp.c: Difference is 1616, ms is 222 [Jul 7 15:59:47] VERBOSE[1750] logger.c: [Jul 7 15:59:47] <--- SIP read from 192.168.0.202:5060 ---> SIP/2.0 100 Trying To: From: "360" ;tag=as58f1335b Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK3fa17140;rport=5060 Content-Length: 0 <-------------> [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 1: To: (27) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 2: From: "360" ;tag=as58f1335b (49) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 3: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 (54) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 4: CSeq: 102 INVITE (16) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK3fa17140;rport=5060 (90) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 6: Content-Length: 0 (17) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 7: (0) [Jul 7 15:59:47] VERBOSE[1750] logger.c: [Jul 7 15:59:47] --- (7 headers 0 lines) --- [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: = Found Their Call ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 Their Tag Our tag: as58f1335b [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: *** SIP TIMER: Cancelling retransmission #29 - INVITE (got response) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18' Request 102: Found [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: SIP response 100 to standard invite [Jul 7 15:59:47] VERBOSE[1750] logger.c: [Jul 7 15:59:47] <--- SIP read from 192.168.0.202:5060 ---> SIP/2.0 180 Ringing Contact: "Valery Komarov" User-Agent: OxO_GW_700/013.001 P-Asserted-Identity: "Valery Komarov" To: ;tag=5bf52f217d35f451ebbc098542c50bc4 From: "360" ;tag=as58f1335b Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK3fa17140;rport=5060 Content-Length: 0 <-------------> [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 1: Contact: "Valery Komarov" (49) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 2: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 3: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 4: To: ;tag=5bf52f217d35f451ebbc098542c50bc4 (64) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 5: From: "360" ;tag=as58f1335b (49) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 6: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 (54) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 7: CSeq: 102 INVITE (16) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK3fa17140;rport=5060 (90) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: Header 10: (0) [Jul 7 15:59:47] VERBOSE[1750] logger.c: [Jul 7 15:59:47] --- (10 headers 0 lines) --- [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: = Found Their Call ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 Their Tag Our tag: as58f1335b [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18' Request 102: Found [Jul 7 15:59:47] DEBUG[1750] chan_sip.c: SIP response 180 to standard invite [Jul 7 15:59:47] DEBUG[1750] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:59:47] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:59:47] DEBUG[1733] chan_sip.c: Checking device state for peer oxo [Jul 7 15:59:47] DEBUG[1733] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:59:47] VERBOSE[1767] logger.c: [Jul 7 15:59:47] -- SIP/oxo-00000002 is ringing [Jul 7 15:59:47] DEBUG[1767] rtp.c: Channel 'Local/141@internal-bae5,2' has no RTP, not doing anything [Jul 7 15:59:47] DEBUG[1739] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:47] DEBUG[1767] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:59:47] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:59:47] DEBUG[1733] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:59:47] DEBUG[1733] channel.c: Avoiding initial deadlock for channel '0x3da2080' [Jul 7 15:59:47] DEBUG[1733] devicestate.c: Changing state for Local/141@internal - state 2 (In use) [Jul 7 15:59:47] VERBOSE[1765] logger.c: [Jul 7 15:59:47] -- Local/141@internal-bae5,1 is ringing [Jul 7 15:59:47] DEBUG[1765] channel.c: Driver for channel 'SIP/360-00000001' does not support indication 3, emulating it [Jul 7 15:59:47] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:47] DEBUG[1739] app_queue.c: Device 'Local/141@internal' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 7 15:59:47] DEBUG[1766] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:47] DEBUG[1766] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:48] DEBUG[1765] rtp.c: Got RTCP report of 64 bytes [Jul 7 15:59:50] DEBUG[1766] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:50] DEBUG[1766] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] <--- SIP read from 192.168.0.202:5060 ---> SIP/2.0 200 OK Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY Contact: "Valery Komarov" Supported: 100rel,timer User-Agent: OxO_GW_700/013.001 Session-Expires: 43200;refresher=uas P-Asserted-Identity: "Valery Komarov" To: ;tag=5bf52f217d35f451ebbc098542c50bc4 From: "360" ;tag=as58f1335b Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK3fa17140;rport=5060 Content-Length: 206 v=0 o=default 1278504001 1278504001 IN IP4 192.168.0.202 s=session c=IN IP4 192.168.0.202 t=0 0 m=audio 32004 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 1: Content-Type: application/sdp (29) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 2: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY (62) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 3: Contact: "Valery Komarov" (49) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 4: Supported: 100rel,timer (23) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 5: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 6: Session-Expires: 43200;refresher=uas (36) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 7: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 8: To: ;tag=5bf52f217d35f451ebbc098542c50bc4 (64) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 9: From: "360" ;tag=as58f1335b (49) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 10: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 (54) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 11: CSeq: 102 INVITE (16) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK3fa17140;rport=5060 (90) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 13: Content-Length: 206 (19) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Header 14: (0) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: v=0 (3) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: o=default 1278504001 1278504001 IN IP4 192.168.0.202 (52) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: s=session (9) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: m=audio 32004 RTP/AVP 8 101 (27) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] --- (14 headers 10 lines) --- [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: = Found Their Call ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 Their Tag 5bf52f217d35f451ebbc098542c50bc4 Our tag: as58f1335b [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Acked pending invite 102 [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Stopping retransmission on '55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18' of Request 102: Match Found [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: SIP response 200 to standard invite [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing session-level SDP o=default 1278504001 1278504001 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing session-level SDP s=session... UNSUPPORTED. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Found RTP audio format 8 [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Found RTP audio format 101 [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Found audio description format telephone-event for ID 101 [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: T38 state changed to 0 on channel SIP/oxo-00000002 [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Peer audio RTP is at port 192.168.0.202:32004 [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: build_route: Contact hop: "Valery Komarov" [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] list_route: hop: [Jul 7 15:59:50] DEBUG[1750] chan_sip.c: Strict routing enforced for session 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] set_destination: Parsing for address/port to send to [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] set_destination: set destination to 192.168.0.202, port 5060 [Jul 7 15:59:50] VERBOSE[1750] logger.c: [Jul 7 15:59:50] Transmitting (no NAT) to 192.168.0.202:5060: ACK sip:141@192.168.0.202 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK365422b7;rport From: "360" ;tag=as58f1335b To: ;tag=5bf52f217d35f451ebbc098542c50bc4 Contact: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 102 ACK User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "360" ;privacy=off;screen=no Content-Length: 0 --- [Jul 7 15:59:50] DEBUG[1767] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:59:50] VERBOSE[1767] logger.c: [Jul 7 15:59:50] -- SIP/oxo-00000002 answered Local/141@internal-bae5,2 [Jul 7 15:59:50] DEBUG[1767] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:59:50] DEBUG[1767] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:59:50] DEBUG[1733] chan_sip.c: Checking device state for peer oxo [Jul 7 15:59:50] DEBUG[1733] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:59:50] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:59:50] DEBUG[1733] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:59:50] DEBUG[1733] devicestate.c: Changing state for Local/141@internal - state 2 (In use) [Jul 7 15:59:50] DEBUG[1765] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:59:50] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:50] DEBUG[1765] channel.c: Set channel SIP/360-00000001 to write format alaw [Jul 7 15:59:50] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1765] channel.c: Got a FRAME_CONTROL (-1) frame on channel Local/141@internal-bae5,1 [Jul 7 15:59:50] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1765] channel.c: Bridge stops bridging channels SIP/360-00000001 and Local/141@internal-bae5,1 [Jul 7 15:59:50] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1739] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:50] DEBUG[1739] app_queue.c: Device 'Local/141@internal' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 7 15:59:50] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:59:50] DEBUG[1733] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:59:50] DEBUG[1733] devicestate.c: Changing state for Local/141@internal - state 2 (In use) [Jul 7 15:59:50] DEBUG[1767] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1767] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1767] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:50] DEBUG[1739] app_queue.c: Device 'Local/141@internal' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 7 15:59:50] DEBUG[1767] channel.c: Planning to masquerade channel SIP/oxo-00000002 into the structure of Local/141@internal-bae5,1 [Jul 7 15:59:50] DEBUG[1767] channel.c: Done planning to masquerade channel SIP/oxo-00000002 into the structure of Local/141@internal-bae5,1 [Jul 7 15:59:50] DEBUG[1767] chan_local.c: Not posting to queue since already masked on 'Local/141@internal-bae5,2' [Jul 7 15:59:50] DEBUG[1765] channel.c: Actually Masquerading SIP/oxo-00000002(6) into the structure of Local/141@internal-bae5,1(6) [Jul 7 15:59:50] DEBUG[1765] channel.c: Got clone lock for masquerade on 'SIP/oxo-00000002' at 0x3dae638 [Jul 7 15:59:50] DEBUG[1765] channel.c: Putting channel SIP/oxo-00000002 in 8/8 formats [Jul 7 15:59:50] DEBUG[1765] chan_sip.c: SIP Fixup: New owner for dialogue 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18: SIP/oxo-00000002 (Old parent: Local/141@internal-bae5,1) [Jul 7 15:59:50] DEBUG[1765] channel.c: Released clone lock on 'Local/141@internal-bae5,1' [Jul 7 15:59:50] DEBUG[1765] channel.c: Done Masquerading SIP/oxo-00000002 (6) [Jul 7 15:59:50] DEBUG[1767] channel.c: Didn't get a frame from channel: Local/141@internal-bae5,2 [Jul 7 15:59:50] DEBUG[1767] channel.c: Bridge stops bridging channels Local/141@internal-bae5,2 and Local/141@internal-bae5,1 [Jul 7 15:59:50] DEBUG[1767] channel.c: Hanging up zombie 'Local/141@internal-bae5,1' [Jul 7 15:59:50] DEBUG[1767] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:59:50] DEBUG[1767] rtp.c: Channel 'Local/141@internal-bae5,2' has no RTP, not doing anything [Jul 7 15:59:50] DEBUG[1767] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 7 15:59:50] DEBUG[1767] pbx.c: Spawn extension (internal,141,1) exited non-zero on 'Local/141@internal-bae5,2' [Jul 7 15:59:50] VERBOSE[1767] logger.c: [Jul 7 15:59:50] == Spawn extension (internal, 141, 1) exited non-zero on 'Local/141@internal-bae5,2' [Jul 7 15:59:50] DEBUG[1767] channel.c: Soft-Hanging up channel 'Local/141@internal-bae5,2' [Jul 7 15:59:50] DEBUG[1767] channel.c: Hanging up channel 'Local/141@internal-bae5,2' [Jul 7 15:59:50] DEBUG[1767] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:59:50] DEBUG[1765] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 15:59:50] DEBUG[1765] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 15:59:50] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:59:50] DEBUG[1733] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:59:50] DEBUG[1733] devicestate.c: Changing state for Local/141@internal - state 1 (Not in use) [Jul 7 15:59:50] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:59:50] DEBUG[1733] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:59:50] DEBUG[1733] devicestate.c: Changing state for Local/141@internal - state 1 (Not in use) [Jul 7 15:59:50] DEBUG[1739] app_queue.c: Device 'Local/141@internal' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:50] DEBUG[1739] app_queue.c: Device 'Local/141@internal' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:52] DEBUG[1766] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:52] DEBUG[1766] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:52] DEBUG[1765] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:52] DEBUG[1765] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:53] DEBUG[1765] rtp.c: Got RTCP report of 64 bytes [Jul 7 15:59:54] VERBOSE[1750] logger.c: [Jul 7 15:59:54] <--- SIP read from 192.168.5.13:5060 ---> BYE sip:140@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK10846269491528426545 From: ;tag=14636877 To: "Valery Komarov" ;tag=as5e4ab5c0 Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 1 BYE Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 0: BYE sip:140@192.168.1.18 SIP/2.0 (32) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK10846269491528426545 (69) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 2: From: ;tag=14636877 (46) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 3: To: "Valery Komarov" ;tag=as5e4ab5c0 (58) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 4: Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 (54) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 5: CSeq: 1 BYE (11) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 7: User-Agent: Voip Phone 1.0 (26) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 8: Content-Length: 0 (17) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Header 9: (0) [Jul 7 15:59:54] VERBOSE[1750] logger.c: [Jul 7 15:59:54] --- (9 headers 0 lines) --- [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: = No match Their Call ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 Their Tag 5bf52f217d35f451ebbc098542c50bc4 Our tag: as58f1335b [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: = Found Their Call ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 Their Tag 14636877 Our tag: as5e4ab5c0 [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 7 15:59:54] VERBOSE[1750] logger.c: [Jul 7 15:59:54] Sending to 192.168.5.13 : 5060 (no NAT) [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 [Jul 7 15:59:54] DEBUG[1750] chan_sip.c: Received bye, issuing owner hangup [Jul 7 15:59:54] VERBOSE[1750] logger.c: [Jul 7 15:59:54] <--- Transmitting (no NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK10846269491528426545;received=192.168.5.13 From: ;tag=14636877 To: "Valery Komarov" ;tag=as5e4ab5c0 Call-ID: 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18 CSeq: 1 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jul 7 15:59:54] DEBUG[1765] channel.c: Didn't get a frame from channel: SIP/360-00000001 [Jul 7 15:59:54] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:54] DEBUG[1765] channel.c: Bridge stops bridging channels SIP/360-00000001 and SIP/oxo-00000002 [Jul 7 15:59:54] DEBUG[1765] channel.c: Set channel SIP/oxo-00000000 to write format alaw [Jul 7 15:59:54] DEBUG[1765] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:54] VERBOSE[1765] logger.c: [Jul 7 15:59:54] -- Stopped music on hold on SIP/oxo-00000000 [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Planning to masquerade channel SIP/oxo-00000000 into the structure of Transfered/SIP/oxo-00000000 [Jul 7 15:59:54] DEBUG[1765] channel.c: Done planning to masquerade channel SIP/oxo-00000000 into the structure of Transfered/SIP/oxo-00000000 [Jul 7 15:59:54] DEBUG[1765] channel.c: Actually Masquerading SIP/oxo-00000000(6) into the structure of Transfered/SIP/oxo-00000000(6) [Jul 7 15:59:54] DEBUG[1765] channel.c: Got clone lock for masquerade on 'SIP/oxo-00000000' at 0x3d7d678 [Jul 7 15:59:54] DEBUG[1765] channel.c: Putting channel SIP/oxo-00000000 in 8/8 formats [Jul 7 15:59:54] DEBUG[1765] chan_sip.c: SIP Fixup: New owner for dialogue a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202: SIP/oxo-00000000 (Old parent: Transfered/SIP/oxo-00000000) [Jul 7 15:59:54] DEBUG[1765] channel.c: Driver for channel 'SIP/oxo-00000000' does not support indication 3, emulating it [Jul 7 15:59:54] DEBUG[1765] channel.c: Set channel SIP/oxo-00000000 to write format slin [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Released clone lock on 'Transfered/SIP/oxo-00000000' [Jul 7 15:59:54] DEBUG[1765] channel.c: Done Masquerading SIP/oxo-00000000 (6) [Jul 7 15:59:54] DEBUG[1765] channel.c: Set channel SIP/oxo-00000002 to write format slin [Jul 7 15:59:54] DEBUG[1765] rtp.c: Difference is 1616, ms is 222 [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:59:54] VERBOSE[1765] logger.c: [Jul 7 15:59:54] -- Playing 'beep' (language 'ru') [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 22 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Set channel SIP/oxo-00000002 to write format alaw [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] channel.c: Hanging up channel 'SIP/360-00000001' [Jul 7 15:59:54] DEBUG[1765] chan_sip.c: Hangup call SIP/360-00000001, SIP callid 7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18) [Jul 7 15:59:54] DEBUG[1765] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 15:59:54] DEBUG[1765] rtp.c: Channel 'Transfered/SIP/oxo-00000000' has no RTP, not doing anything [Jul 7 15:59:54] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 15:59:54] DEBUG[1765] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 7 15:59:54] DEBUG[1733] chan_sip.c: Checking device state for peer 360 [Jul 7 15:59:54] DEBUG[1768] channel.c: Set channel SIP/oxo-00000000 to write format alaw [Jul 7 15:59:54] DEBUG[1733] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 15:59:54] DEBUG[1768] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:59:54] DEBUG[1765] pbx.c: Spawn extension (oxo,360,1) exited non-zero on 'Transfered/SIP/oxo-00000000' [Jul 7 15:59:54] DEBUG[1739] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 15:59:54] VERBOSE[1765] logger.c: [Jul 7 15:59:54] == Spawn extension (oxo, 360, 1) exited non-zero on 'Transfered/SIP/oxo-00000000' [Jul 7 15:59:54] DEBUG[1765] channel.c: Soft-Hanging up channel 'Transfered/SIP/oxo-00000000' [Jul 7 15:59:54] DEBUG[1765] channel.c: Hanging up zombie 'Transfered/SIP/oxo-00000000' [Jul 7 15:59:54] DEBUG[1765] devicestate.c: Notification of state change to be queued on device/channel Transfered/SIP/oxo [Jul 7 15:59:54] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for Transfered - SIP/oxo [Jul 7 15:59:54] DEBUG[1733] devicestate.c: Changing state for Transfered/SIP/oxo - state 4 (Invalid) [Jul 7 15:59:54] DEBUG[1739] app_queue.c: Device 'Transfered/SIP/oxo' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jul 7 15:59:54] DEBUG[1768] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:54] DEBUG[1768] rtp.c: Setting the marker bit due to a source update [Jul 7 15:59:55] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:55] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:55] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:55] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:55] VERBOSE[1750] logger.c: [Jul 7 15:59:55] Really destroying SIP dialog '7efb89a20a64d6f111adbbf878a1cf99@192.168.1.18' Method: BYE [Jul 7 15:59:57] DEBUG[1768] rtp.c: Got RTCP report of 36 bytes [Jul 7 15:59:57] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:59:57] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:59:57] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:00] DEBUG[1768] rtp.c: Got RTCP report of 36 bytes [Jul 7 16:00:00] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:00] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:00:00] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:02] DEBUG[1768] rtp.c: Got RTCP report of 36 bytes [Jul 7 16:00:02] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:02] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:00:02] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:05] DEBUG[1768] rtp.c: Got RTCP report of 36 bytes [Jul 7 16:00:05] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:05] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:00:05] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:07] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:00:07] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:08] DEBUG[1768] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:00:08] DEBUG[1768] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] <--- SIP read from 192.168.0.202:5060 ---> BYE sip:360@192.168.1.18 SIP/2.0 Route: User-Agent: OxO_GW_700/013.001 To: sip:360@isoemo.com;tag=as078f6861 From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670238 BYE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK0e2e31f4f0db9c1a367818a4f0287810 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 0: BYE sip:360@192.168.1.18 SIP/2.0 (32) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 1: Route: (28) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 2: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 3: To: sip:360@isoemo.com;tag=as078f6861 (37) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 4: From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d (64) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 5: Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 (55) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 6: CSeq: 517670238 BYE (19) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 7: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK0e2e31f4f0db9c1a367818a4f0287810 (83) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 10: (0) [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] --- (10 headers 0 lines) --- [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: = No match Their Call ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 Their Tag 5bf52f217d35f451ebbc098542c50bc4 Our tag: as58f1335b [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: = Found Their Call ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 Their Tag 2eade5bd4fc1015d4d4aee3f8485445d Our tag: as078f6861 [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] Sending to 192.168.0.202 : 5060 (NAT) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Setting SIP_ALREADYGONE on dialog a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Received bye, issuing owner hangup [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK0e2e31f4f0db9c1a367818a4f0287810;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=2eade5bd4fc1015d4d4aee3f8485445d To: sip:360@isoemo.com;tag=as078f6861 Call-ID: a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202 CSeq: 517670238 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jul 7 16:00:08] DEBUG[1768] channel.c: Didn't get a frame from channel: SIP/oxo-00000000 [Jul 7 16:00:08] DEBUG[1768] rtp.c: Setting the marker bit due to a source update [Jul 7 16:00:08] DEBUG[1768] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/oxo-00000002 [Jul 7 16:00:08] DEBUG[1768] channel.c: Hanging up channel 'SIP/oxo-00000002' [Jul 7 16:00:08] DEBUG[1768] chan_sip.c: Hangup call SIP/oxo-00000002, SIP callid 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18) [Jul 7 16:00:08] VERBOSE[1768] logger.c: [Jul 7 16:00:08] Scheduling destruction of SIP dialog '55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18' in 6400 ms (Method: INVITE) [Jul 7 16:00:08] DEBUG[1768] chan_sip.c: Strict routing enforced for session 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 [Jul 7 16:00:08] VERBOSE[1768] logger.c: [Jul 7 16:00:08] set_destination: Parsing for address/port to send to [Jul 7 16:00:08] VERBOSE[1768] logger.c: [Jul 7 16:00:08] set_destination: set destination to 192.168.0.202, port 5060 [Jul 7 16:00:08] VERBOSE[1768] logger.c: [Jul 7 16:00:08] Reliably Transmitting (no NAT) to 192.168.0.202:5060: BYE sip:141@192.168.0.202 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK22c57a86;rport From: "360" ;tag=as58f1335b To: ;tag=5bf52f217d35f451ebbc098542c50bc4 Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 103 BYE User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "360" ;privacy=off;screen=no X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Jul 7 16:00:08] DEBUG[1768] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:00:08] DEBUG[1768] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 16:00:08] DEBUG[1768] channel.c: Hanging up channel 'SIP/oxo-00000000' [Jul 7 16:00:08] DEBUG[1768] chan_sip.c: Hangup call SIP/oxo-00000000, SIP callid a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202) [Jul 7 16:00:08] DEBUG[1768] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 16:00:08] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 16:00:08] DEBUG[1733] chan_sip.c: Checking device state for peer oxo [Jul 7 16:00:08] DEBUG[1733] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 16:00:08] DEBUG[1733] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 16:00:08] DEBUG[1733] chan_sip.c: Checking device state for peer oxo [Jul 7 16:00:08] DEBUG[1733] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 16:00:08] DEBUG[1739] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:00:08] DEBUG[1739] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] Really destroying SIP dialog 'a1135e063e499a4efdb4d37c8d1d9d05@192.168.0.202' Method: BYE [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] <--- SIP read from 192.168.0.202:5060 ---> SIP/2.0 200 OK Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY,INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY Supported: 100rel,100rel User-Agent: OxO_GW_700/013.001 To: ;tag=5bf52f217d35f451ebbc098542c50bc4 From: "360" ;tag=as58f1335b Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK22c57a86;rport=5060 Content-Length: 0 <-------------> [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 1: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY,INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY (118) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 2: Supported: 100rel,100rel (24) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 4: To: ;tag=5bf52f217d35f451ebbc098542c50bc4 (64) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 5: From: "360" ;tag=as58f1335b (49) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 6: Call-ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 (54) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 7: CSeq: 103 BYE (13) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK22c57a86;rport=5060 (90) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Header 10: (0) [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] --- (10 headers 0 lines) --- [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: = Found Their Call ID: 55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18 Their Tag 5bf52f217d35f451ebbc098542c50bc4 Our tag: as58f1335b [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 [Jul 7 16:00:08] DEBUG[1750] chan_sip.c: Stopping retransmission on '55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18' of Request 103: Match Found [Jul 7 16:00:08] VERBOSE[1750] logger.c: [Jul 7 16:00:08] Really destroying SIP dialog '55c641ce7a3a43361e3d2713322ff4d2@192.168.1.18' Method: INVITE [Jul 7 16:00:10] DEBUG[1750] chan_sip.c: Auto destroying SIP dialog '23319369128661-235514975462@192.168.5.13' [Jul 7 16:00:10] DEBUG[1750] chan_sip.c: Destroying SIP dialog 23319369128661-235514975462@192.168.5.13 [Jul 7 16:00:10] VERBOSE[1750] logger.c: [Jul 7 16:00:10] Really destroying SIP dialog '23319369128661-235514975462@192.168.5.13' Method: REGISTER [Jul 7 16:00:13] VERBOSE[1764] logger.c: [Jul 7 16:00:13] Beginning asterisk shutdown.... [Jul 7 16:00:13] VERBOSE[1764] logger.c: [Jul 7 16:00:13] Executing last minute cleanups [Jul 7 16:00:13] VERBOSE[1764] logger.c: [Jul 7 16:00:13] == Destroying musiconhold processes [Jul 7 16:00:13] DEBUG[1764] res_musiconhold.c: Destroying MOH class 'default' [Jul 7 16:00:13] VERBOSE[1764] logger.c: [Jul 7 16:00:13] Asterisk cleanly ending (0). [Jul 7 16:00:13] DEBUG[1764] asterisk.c: Asterisk ending (0).