[Jul 7 15:56:50] VERBOSE[1671] logger.c: [Jul 7 15:56:50] Asterisk Event Logger restarted [Jul 7 15:56:50] VERBOSE[1671] logger.c: [Jul 7 15:56:50] Asterisk Queue Logger restarted [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] <--- 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=0562b16854491908ed4104b44b7d0670 Contact: "Valery Komarov" Content-Type: application/sdp Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116604 INVITE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK9a95c7415ad00628f5843d999c66a71a Max-Forwards: 70 Content-Length: 251 v=0 o=default 1278503822 1278503822 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 15:56:52] DEBUG[1687] chan_sip.c: Header 0: INVITE sip:360@isoemo.com SIP/2.0 (33) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 1: Route: (28) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 2: Supported: 100rel,timer (23) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 4: Session-Expires: 43200 (22) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 5: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 6: To: sip:360@isoemo.com (22) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 7: From: "Valery Komarov" ;tag=0562b16854491908ed4104b44b7d0670 (83) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 8: Contact: "Valery Komarov" (63) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 10: Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 (55) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 11: CSeq: 1585116604 INVITE (23) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK9a95c7415ad00628f5843d999c66a71a (83) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 13: Max-Forwards: 70 (16) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 14: Content-Length: 251 (19) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 15: (0) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: v=0 (3) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: o=default 1278503822 1278503822 IN IP4 192.168.0.202 (52) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: s=- (3) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: m=audio 32000 RTP/AVP 18 106 4 8 0 (34) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: a=fmtp:4 annexa=no (18) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Line: a=maxptime:90 (13) [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] --- (15 headers 12 lines) --- [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: = No match Their Call ID: 107402794129859-2893711712540@192.168.5.13 Their Tag 1273021861 Our tag: as7960ea74 [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Setting NAT on VRTP to Off [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Allocating new SIP dialog for 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 - INVITE (With RTP) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Begin: parsing SIP "Supported: 100rel,timer" [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Found SIP option: -100rel- [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Matched SIP option: 100rel [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Found SIP option: -timer- [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Matched SIP option: timer [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Sending to 192.168.0.202 : 5060 (no NAT) [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Using INVITE request as basis request - 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found peer 'oxo' [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Setting NAT on VRTP to Off [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing session-level SDP o=default 1278503822 1278503822 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found RTP audio format 18 [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found RTP audio format 106 [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found RTP audio format 4 [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found RTP audio format 8 [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found RTP audio format 0 [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Found audio description format telephone-event for ID 106 [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=fmtp:4 annexa=no... UNSUPPORTED. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=maxptime:90... UNSUPPORTED. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: T38 state changed to 0 on channel [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Peer audio RTP is at port 192.168.0.202:32000 [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Checking SIP call limits for device [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Updating call counter for incoming call [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] Looking for 360 in oxo (domain isoemo.com) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: This channel will not be able to handle video. [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: build_route: Contact hop: "Valery Komarov" [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] list_route: hop: [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: SIP/oxo-00000000: New call is still down.... Trying... [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK9a95c7415ad00628f5843d999c66a71a;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=0562b16854491908ed4104b44b7d0670 To: sip:360@isoemo.com Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116604 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 15:56:52] DEBUG[1687] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:56:52] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:56:52] DEBUG[1673] chan_sip.c: Checking device state for peer oxo [Jul 7 15:56:52] DEBUG[1673] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:56:52] DEBUG[1702] pbx.c: Launching 'Dial' [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] -- Executing [360@oxo:1] Dial("SIP/oxo-00000000", "SIP/360|60|t") in new stack [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jul 7 15:56:52] DEBUG[1679] 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 15:56:52] DEBUG[1702] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: This channel will not be able to handle video. [Jul 7 15:56:52] DEBUG[1702] rtp.c: Seeded SDP of 'SIP/360-00000001' with that of 'SIP/oxo-00000000' [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable DIALEDTIME. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable ANSWEREDTIME. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable DIALEDPEERNAME. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable DIALSTATUS. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable SIPCALLID. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable SIPUSERAGENT. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable SIPDOMAIN. [Jul 7 15:56:52] DEBUG[1702] channel.c: Not copying variable SIPURI. [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Outgoing Call for 360 [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] Audio is at 192.168.1.18 port 11172 [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] Adding codec 0x8 (alaw) to SDP [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] Adding codec 0x4 (ulaw) to SDP [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 0: INVITE sip:360@192.168.5.13:5060 SIP/2.0 (40) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport (63) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as4d922c05 (60) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 3: To: (31) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 4: Contact: (31) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 5: Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 (54) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 7: User-Agent: Asterisk (20) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 9: Remote-Party-ID: "Valery Komarov" ;privacy=off;screen=no (78) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 10: Date: Wed, 07 Jul 2010 11:56:52 GMT (35) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 12: Supported: replaces (19) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 14: Content-Length: 235 (19) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Header 15: (0) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: v=0 (3) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: o=root 1671 1671 IN IP4 192.168.1.18 (36) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: s=session (9) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: c=IN IP4 192.168.1.18 (21) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: m=audio 11172 RTP/AVP 8 0 106 (29) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: a=fmtp:106 0-16 (15) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:56:52] DEBUG[1702] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] 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=z9hG4bK59f840ee;rport From: "Valery Komarov" ;tag=as4d922c05 To: Contact: Call-ID: 1372afc0528c658242ea54576376136a@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 11:56:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=root 1671 1671 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 11172 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 15:56:52] DEBUG[1702] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] -- Called 360 [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport From: "Valery Komarov" ;tag=as4d922c05 To: Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport (63) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as4d922c05 (60) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 3: To: (31) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 4: Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 (54) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 6: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 7: Content-Length: 0 (17) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 8: (0) [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] --- (8 headers 0 lines) --- [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: = Found Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag Our tag: as4d922c05 [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: *** SIP TIMER: Cancelling retransmission #22 - INVITE (got response) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1372afc0528c658242ea54576376136a@192.168.1.18' Request 102: Found [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: SIP response 100 to standard invite [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport From: "Valery Komarov" ;tag=as4d922c05 To: ;tag=2021418042 Call-ID: 1372afc0528c658242ea54576376136a@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 15:56:52] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport (63) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as4d922c05 (60) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 3: To: ;tag=2021418042 (46) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 4: Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 (54) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 6: Contact: (36) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 7: Server: Voip Phone 1.0 (22) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 8: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: Header 10: (0) [Jul 7 15:56:52] VERBOSE[1687] logger.c: [Jul 7 15:56:52] --- (10 headers 0 lines) --- [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: = Found Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag Our tag: as4d922c05 [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1372afc0528c658242ea54576376136a@192.168.1.18' Request 102: Found [Jul 7 15:56:52] DEBUG[1687] chan_sip.c: SIP response 180 to standard invite [Jul 7 15:56:52] DEBUG[1687] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 15:56:52] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 15:56:52] DEBUG[1673] chan_sip.c: Checking device state for peer 360 [Jul 7 15:56:52] DEBUG[1673] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 15:56:52] DEBUG[1679] 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 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] -- SIP/360-00000001 is ringing [Jul 7 15:56:52] DEBUG[1702] rtp.c: Setting early bridge SDP of 'SIP/oxo-00000000' with that of 'SIP/360-00000001' [Jul 7 15:56:52] VERBOSE[1702] logger.c: [Jul 7 15:56:52] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK9a95c7415ad00628f5843d999c66a71a;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=0562b16854491908ed4104b44b7d0670 To: sip:360@isoemo.com;tag=as3bd16478 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116604 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport From: "Valery Komarov" ;tag=as4d922c05 To: ;tag=2021418042 Call-ID: 1372afc0528c658242ea54576376136a@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 13275123 15607144 IN IP4 192.168.5.13 s=A conversation c=IN IP4 192.168.5.13 t=0 0 m=audio 10234 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 15:56:53] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK59f840ee;rport (63) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as4d922c05 (60) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 3: To: ;tag=2021418042 (46) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 4: Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 (54) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 6: Contact: (36) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 7: Supported: 100rel, replaces, timer (34) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 8: Server: Voip Phone 1.0 (22) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 9: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 11: Content-Length: 237 (19) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 12: (0) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: v=0 (3) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: o=360 13275123 15607144 IN IP4 192.168.5.13 (43) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: s=A conversation (16) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: c=IN IP4 192.168.5.13 (21) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: m=audio 10234 RTP/AVP 8 0 106 (29) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] --- (12 headers 11 lines) --- [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = Found Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag 2021418042 Our tag: as4d922c05 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Acked pending invite 102 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Stopping retransmission on '1372afc0528c658242ea54576376136a@192.168.1.18' of Request 102: Match Found [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: SIP response 200 to standard invite [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP o=360 13275123 15607144 IN IP4 192.168.5.13... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.13... OK. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found RTP audio format 8 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found RTP audio format 0 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found RTP audio format 106 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found audio description format PCMA for ID 8 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found audio description format PCMU for ID 0 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found audio description format telephone-event for ID 106 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: T38 state changed to 0 on channel SIP/360-00000001 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Peer audio RTP is at port 192.168.5.13:10234 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: build_route: Contact hop: [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] list_route: hop: [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Strict routing enforced for session 1372afc0528c658242ea54576376136a@192.168.1.18 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] set_destination: Parsing for address/port to send to [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] set_destination: set destination to 192.168.5.13, port 5060 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] 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=z9hG4bK43df5999;rport From: "Valery Komarov" ;tag=as4d922c05 To: ;tag=2021418042 Contact: Call-ID: 1372afc0528c658242ea54576376136a@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 15:56:53] DEBUG[1702] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 15:56:53] VERBOSE[1702] logger.c: [Jul 7 15:56:53] -- SIP/360-00000001 answered SIP/oxo-00000000 [Jul 7 15:56:53] DEBUG[1702] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:56:53] DEBUG[1702] chan_sip.c: SIP answering channel: SIP/oxo-00000000 [Jul 7 15:56:53] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:53] DEBUG[1702] chan_sip.c: Setting framing from config on incoming call [Jul 7 15:56:53] DEBUG[1702] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jul 7 15:56:53] DEBUG[1702] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 7 15:56:53] VERBOSE[1702] logger.c: [Jul 7 15:56:53] Audio is at 192.168.1.18 port 15506 [Jul 7 15:56:53] VERBOSE[1702] logger.c: [Jul 7 15:56:53] Adding codec 0x8 (alaw) to SDP [Jul 7 15:56:53] VERBOSE[1702] logger.c: [Jul 7 15:56:53] Adding codec 0x4 (ulaw) to SDP [Jul 7 15:56:53] VERBOSE[1702] logger.c: [Jul 7 15:56:53] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:56:53] DEBUG[1702] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:56:53] DEBUG[1702] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 15:56:53] VERBOSE[1702] logger.c: [Jul 7 15:56:53] <--- Reliably Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK9a95c7415ad00628f5843d999c66a71a;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=0562b16854491908ed4104b44b7d0670 To: sip:360@isoemo.com;tag=as3bd16478 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116604 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 1671 1671 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 15506 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 15:56:53] DEBUG[1702] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:56:53] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:53] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:53] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 15:56:53] DEBUG[1673] chan_sip.c: Checking device state for peer 360 [Jul 7 15:56:53] DEBUG[1673] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 15:56:53] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:56:53] DEBUG[1673] chan_sip.c: Checking device state for peer oxo [Jul 7 15:56:53] DEBUG[1679] 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 15:56:53] DEBUG[1673] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:56:53] DEBUG[1679] 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 15:56:53] DEBUG[1702] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 15:56:53] DEBUG[1702] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 15:56:53] DEBUG[1702] rtp.c: Got RTCP report of 36 bytes [Jul 7 15:56:53] DEBUG[1702] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] <--- 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=as3bd16478 From: "Valery Komarov" ;tag=0562b16854491908ed4104b44b7d0670 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116604 ACK Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKcb2c592510013f0d2f1defd49420dd67 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 0: ACK sip:360@192.168.1.18 SIP/2.0 (32) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 1: Route: (28) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 2: Contact: "Valery Komarov" (49) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 4: To: sip:360@isoemo.com;tag=as3bd16478 (37) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 5: From: "Valery Komarov" ;tag=0562b16854491908ed4104b44b7d0670 (83) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 6: Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 (55) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 7: CSeq: 1585116604 ACK (20) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKcb2c592510013f0d2f1defd49420dd67 (83) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 10: Content-Length: 0 (17) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 11: (0) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] --- (11 headers 0 lines) --- [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = No match Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag 2021418042 Our tag: as4d922c05 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 Their Tag 0562b16854491908ed4104b44b7d0670 Our tag: as3bd16478 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Stopping retransmission on '7261abed7bd3c20ab4ef0421ab828395@192.168.0.202' of Response 1585116604: Match Found [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] <--- 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=as3bd16478 From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116605 INVITE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKf79ca710514e7c237aa0bfc859f43663 Max-Forwards: 70 Content-Length: 215 v=0 o=default 1278503822 1278503823 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 15:56:53] DEBUG[1687] chan_sip.c: Header 0: INVITE sip:360@192.168.1.18 SIP/2.0 (35) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 1: Route: (28) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 2: Supported: 100rel,timer (23) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 4: Session-Expires: 43200;refresher=uac (36) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 5: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 6: Contact: "Valery Komarov" (63) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 7: Content-Type: application/sdp (29) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 8: To: sip:360@isoemo.com;tag=as3bd16478 (37) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 9: From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 (64) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 10: Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 (55) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 11: CSeq: 1585116605 INVITE (23) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKf79ca710514e7c237aa0bfc859f43663 (83) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 13: Max-Forwards: 70 (16) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 14: Content-Length: 215 (19) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 15: (0) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: v=0 (3) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: o=default 1278503822 1278503823 IN IP4 192.168.0.202 (52) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: s=- (3) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: m=audio 32000 RTP/AVP 8 106 (27) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Line: a=maxptime:90 (13) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] --- (15 headers 11 lines) --- [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = No match Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag 2021418042 Our tag: as4d922c05 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 Their Tag 0562b16854491908ed4104b44b7d0670 Our tag: as3bd16478 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Sending to 192.168.0.202 : 5060 (NAT) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP o=default 1278503822 1278503823 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found RTP audio format 8 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found RTP audio format 106 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Found audio description format telephone-event for ID 106 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=maxptime:90... UNSUPPORTED. [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: T38 state changed to 0 on channel SIP/oxo-00000000 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Peer audio RTP is at port 192.168.0.202:32000 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Got a SIP re-invite for call 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: SIP/oxo-00000000: This call is UP.... [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKf79ca710514e7c237aa0bfc859f43663;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 To: sip:360@isoemo.com;tag=as3bd16478 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116605 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Setting framing from config on incoming call [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Audio is at 192.168.1.18 port 15506 [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Adding codec 0x8 (alaw) to SDP [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] <--- Reliably Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKf79ca710514e7c237aa0bfc859f43663;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 To: sip:360@isoemo.com;tag=as3bd16478 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116605 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 1671 1672 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 15506 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 15:56:53] DEBUG[1687] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:56:53] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] <--- 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=as3bd16478 From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116605 ACK Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKa01d94940506286def33da46f228cd5f Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 0: ACK sip:360@192.168.1.18 SIP/2.0 (32) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 1: Route: (28) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 2: Contact: "Valery Komarov" (49) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 4: To: sip:360@isoemo.com;tag=as3bd16478 (37) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 5: From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 (64) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 6: Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 (55) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 7: CSeq: 1585116605 ACK (20) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKa01d94940506286def33da46f228cd5f (83) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 10: Content-Length: 0 (17) [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Header 11: (0) [Jul 7 15:56:53] VERBOSE[1687] logger.c: [Jul 7 15:56:53] --- (11 headers 0 lines) --- [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = No match Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag 2021418042 Our tag: as4d922c05 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 Their Tag 0562b16854491908ed4104b44b7d0670 Our tag: as3bd16478 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 [Jul 7 15:56:53] DEBUG[1687] chan_sip.c: Stopping retransmission on '7261abed7bd3c20ab4ef0421ab828395@192.168.0.202' of Response 1585116605: Match Found [Jul 7 15:56:53] DEBUG[1702] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 15:56:53] DEBUG[1702] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF begin '*' received on SIP/360-00000001 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF begin passthrough '*' on SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] channel.c: Got DTMF begin on channel (SIP/360-00000001) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF end '*' received on SIP/360-00000001, duration 120 ms [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF end accepted with begin '*' on SIP/360-00000001 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF end passthrough '*' on SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] channel.c: Got DTMF end on channel (SIP/360-00000001) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] res_features.c: Feature interpret: chan=SIP/oxo-00000000, peer=SIP/360-00000001, code=*, sense=2, features=2, dynamic=# [Jul 7 15:56:54] DEBUG[1702] res_features.c: Set time limit to 1000 [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF begin '*' received on SIP/360-00000001 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF begin passthrough '*' on SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] channel.c: Got DTMF begin on channel (SIP/360-00000001) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Sending dtmf: 42 (*), at 192.168.5.13 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF end '*' received on SIP/360-00000001, duration 120 ms [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF end accepted with begin '*' on SIP/360-00000001 [Jul 7 15:56:54] DTMF[1702] channel.c: DTMF end passthrough '*' on SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] channel.c: Got DTMF end on channel (SIP/360-00000001) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] DEBUG[1702] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000001 [Jul 7 15:56:54] DEBUG[1702] res_features.c: Feature interpret: chan=SIP/oxo-00000000, peer=SIP/360-00000001, code=**, sense=2, features=2, dynamic=# [Jul 7 15:56:54] DEBUG[1702] res_features.c: Feature detected: fname=Attended Transfer sname=atxfer exten=** [Jul 7 15:56:54] DEBUG[1702] res_features.c: Executing Attended Transfer SIP/oxo-00000000, SIP/360-00000001 (sense=2) [Jul 7 15:56:54] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:54] VERBOSE[1702] logger.c: [Jul 7 15:56:54] -- Started music on hold, class 'default', on SIP/oxo-00000000 [Jul 7 15:56:54] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:56:54] DEBUG[1702] channel.c: Set channel SIP/360-00000001 to write format slin [Jul 7 15:56:54] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:56:54] VERBOSE[1702] logger.c: [Jul 7 15:56:54] -- Playing 'pbx-transfer' (language 'ru') [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:54] DEBUG[1703] channel.c: Set channel SIP/oxo-00000000 to write format slin [Jul 7 15:56:54] DEBUG[1703] res_musiconhold.c: SIP/oxo-00000000 Opened file 4 '/var/lib/asterisk/moh/22-Chet Atkins-Light My Fire' [Jul 7 15:56:54] WARNING[1703] mp3/interface.c: Junk at the beginning of frame 49443303 [Jul 7 15:56:54] DEBUG[1703] rtp.c: Difference is 1104, ms is 158 [Jul 7 15:56:54] DEBUG[1702] rtp.c: - RTP 2833 Event: 0000000a (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Got RTCP report of 64 bytes [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF begin '1' received on SIP/360-00000001 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF begin ignored '1' on SIP/360-00000001 [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF end '1' received on SIP/360-00000001, duration 120 ms [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF end passthrough '1' on SIP/360-00000001 [Jul 7 15:56:55] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Sending dtmf: 52 (4), at 192.168.5.13 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF begin '4' received on SIP/360-00000001 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF begin ignored '4' on SIP/360-00000001 [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Sending dtmf: 52 (4), at 192.168.5.13 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF end '4' received on SIP/360-00000001, duration 120 ms [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF end passthrough '4' on SIP/360-00000001 [Jul 7 15:56:55] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000004 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF begin '1' received on SIP/360-00000001 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF begin ignored '1' on SIP/360-00000001 [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF end '1' received on SIP/360-00000001, duration 120 ms [Jul 7 15:56:55] DTMF[1702] channel.c: DTMF end passthrough '1' on SIP/360-00000001 [Jul 7 15:56:55] DEBUG[1702] channel.c: Not copying variable BRIDGEPEER. [Jul 7 15:56:55] DEBUG[1702] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 15:56:55] DEBUG[1702] channel.c: Not copying variable SIPCALLID. [Jul 7 15:56:55] DEBUG[1702] channel.c: Driver for channel 'SIP/360-00000001' does not support indication 3, emulating it [Jul 7 15:56:55] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:56:55] DEBUG[1704] pbx.c: Launching 'Dial' [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] -- Executing [141@internal:1] Dial("Local/141@internal-a9e8,2", "SIP/141@oxo|60|t") in new stack [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Setting NAT on RTP to Off [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: This channel will not be able to handle video. [Jul 7 15:56:55] DEBUG[1704] rtp.c: Channel 'Local/141@internal-a9e8,2' has no RTP, not doing anything [Jul 7 15:56:55] DEBUG[1704] channel.c: Not copying variable DIALEDTIME. [Jul 7 15:56:55] DEBUG[1704] channel.c: Not copying variable ANSWEREDTIME. [Jul 7 15:56:55] DEBUG[1704] channel.c: Not copying variable DIALEDPEERNAME. [Jul 7 15:56:55] DEBUG[1704] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 15:56:55] DEBUG[1704] channel.c: Not copying variable DIALSTATUS. [Jul 7 15:56:55] DEBUG[1704] channel.c: Not copying variable TRANSFERERNAME. [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Outgoing Call for 141 [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] Audio is at 192.168.1.18 port 14516 [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] Adding codec 0x8 (alaw) to SDP [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] Adding codec 0x4 (ulaw) to SDP [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 0: INVITE sip:141@192.168.0.202 SIP/2.0 (36) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK2875af93;rport (63) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 2: From: "360" ;tag=as2cab833b (49) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 3: To: (27) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 4: Contact: (31) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 5: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 (54) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 7: User-Agent: Asterisk (20) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 9: Remote-Party-ID: "360" ;privacy=off;screen=no (67) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 10: Date: Wed, 07 Jul 2010 11:56:55 GMT (35) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 12: Supported: replaces (19) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 14: Content-Length: 235 (19) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Header 15: (0) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: v=0 (3) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: o=root 1671 1671 IN IP4 192.168.1.18 (36) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: s=session (9) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: c=IN IP4 192.168.1.18 (21) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: m=audio 14516 RTP/AVP 8 0 101 (29) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:56:55] DEBUG[1704] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] 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=z9hG4bK2875af93;rport From: "360" ;tag=as2cab833b To: Contact: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@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 11:56:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=root 1671 1671 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 14516 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 15:56:55] DEBUG[1704] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] -- Called 141@oxo [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] DEBUG[1702] rtp.c: Difference is 1912, ms is 259 [Jul 7 15:56:55] DEBUG[1702] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 7 15:56:55] VERBOSE[1687] logger.c: [Jul 7 15:56:55] <--- SIP read from 192.168.0.202:5060 ---> SIP/2.0 100 Trying To: From: "360" ;tag=as2cab833b Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK2875af93;rport=5060 Content-Length: 0 <-------------> [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 1: To: (27) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 2: From: "360" ;tag=as2cab833b (49) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 3: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 (54) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 4: CSeq: 102 INVITE (16) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK2875af93;rport=5060 (90) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 6: Content-Length: 0 (17) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 7: (0) [Jul 7 15:56:55] VERBOSE[1687] logger.c: [Jul 7 15:56:55] --- (7 headers 0 lines) --- [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 Their Tag Our tag: as2cab833b [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: *** SIP TIMER: Cancelling retransmission #29 - INVITE (got response) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18' Request 102: Found [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: SIP response 100 to standard invite [Jul 7 15:56:55] DEBUG[1703] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:56:55] DEBUG[1703] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:56:55] VERBOSE[1687] logger.c: [Jul 7 15:56:55] <--- 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=6038a805cf482d5abeed1906e001e4d9 From: "360" ;tag=as2cab833b Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK2875af93;rport=5060 Content-Length: 0 <-------------> [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 1: Contact: "Valery Komarov" (49) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 2: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 3: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 4: To: ;tag=6038a805cf482d5abeed1906e001e4d9 (64) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 5: From: "360" ;tag=as2cab833b (49) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 6: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 (54) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 7: CSeq: 102 INVITE (16) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK2875af93;rport=5060 (90) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: Header 10: (0) [Jul 7 15:56:55] VERBOSE[1687] logger.c: [Jul 7 15:56:55] --- (10 headers 0 lines) --- [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 Their Tag Our tag: as2cab833b [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18' Request 102: Found [Jul 7 15:56:55] DEBUG[1687] chan_sip.c: SIP response 180 to standard invite [Jul 7 15:56:55] DEBUG[1687] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:56:55] VERBOSE[1704] logger.c: [Jul 7 15:56:55] -- SIP/oxo-00000002 is ringing [Jul 7 15:56:55] DEBUG[1704] rtp.c: Channel 'Local/141@internal-a9e8,2' has no RTP, not doing anything [Jul 7 15:56:55] DEBUG[1704] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:56:55] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:56:55] DEBUG[1673] chan_sip.c: Checking device state for peer oxo [Jul 7 15:56:55] DEBUG[1673] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:56:55] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:56:55] DEBUG[1679] 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 15:56:55] VERBOSE[1702] logger.c: [Jul 7 15:56:55] -- Local/141@internal-a9e8,1 is ringing [Jul 7 15:56:55] DEBUG[1702] channel.c: Driver for channel 'SIP/360-00000001' does not support indication 3, emulating it [Jul 7 15:56:55] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:56:55] DEBUG[1673] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:56:55] DEBUG[1673] devicestate.c: Changing state for Local/141@internal - state 2 (In use) [Jul 7 15:56:55] DEBUG[1679] 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 15:56:58] DEBUG[1703] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:56:58] DEBUG[1703] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] <--- 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=6038a805cf482d5abeed1906e001e4d9 From: "360" ;tag=as2cab833b Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK2875af93;rport=5060 Content-Length: 206 v=0 o=default 1278503830 1278503830 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 15:56:59] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 1: Content-Type: application/sdp (29) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 2: Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY (62) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 3: Contact: "Valery Komarov" (49) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 4: Supported: 100rel,timer (23) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 5: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 6: Session-Expires: 43200;refresher=uas (36) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 7: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 8: To: ;tag=6038a805cf482d5abeed1906e001e4d9 (64) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 9: From: "360" ;tag=as2cab833b (49) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 10: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 (54) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 11: CSeq: 102 INVITE (16) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK2875af93;rport=5060 (90) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 13: Content-Length: 206 (19) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Header 14: (0) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: v=0 (3) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: o=default 1278503830 1278503830 IN IP4 192.168.0.202 (52) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: s=session (9) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: t=0 0 (5) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: m=audio 32004 RTP/AVP 8 101 (27) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: a=sendrecv (10) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] --- (14 headers 10 lines) --- [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 Their Tag 6038a805cf482d5abeed1906e001e4d9 Our tag: as2cab833b [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Acked pending invite 102 [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Stopping retransmission on '7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18' of Request 102: Match Found [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: SIP response 200 to standard invite [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing session-level SDP o=default 1278503830 1278503830 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing session-level SDP s=session... UNSUPPORTED. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] Found RTP audio format 8 [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] Found RTP audio format 101 [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] Found audio description format telephone-event for ID 101 [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: T38 state changed to 0 on channel SIP/oxo-00000002 [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] Peer audio RTP is at port 192.168.0.202:32004 [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Updating call counter for outgoing call [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: build_route: Contact hop: "Valery Komarov" [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] list_route: hop: [Jul 7 15:56:59] DEBUG[1687] chan_sip.c: Strict routing enforced for session 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] set_destination: Parsing for address/port to send to [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] set_destination: set destination to 192.168.0.202, port 5060 [Jul 7 15:56:59] VERBOSE[1687] logger.c: [Jul 7 15:56:59] 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=z9hG4bK6733f6a9;rport From: "360" ;tag=as2cab833b To: ;tag=6038a805cf482d5abeed1906e001e4d9 Contact: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@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 15:56:59] DEBUG[1704] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:56:59] VERBOSE[1704] logger.c: [Jul 7 15:56:59] -- SIP/oxo-00000002 answered Local/141@internal-a9e8,2 [Jul 7 15:56:59] DEBUG[1704] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:56:59] DEBUG[1704] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:56:59] DEBUG[1673] chan_sip.c: Checking device state for peer oxo [Jul 7 15:56:59] DEBUG[1673] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:56:59] DEBUG[1679] 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 15:56:59] DEBUG[1702] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:56:59] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:56:59] DEBUG[1702] channel.c: Set channel SIP/360-00000001 to write format alaw [Jul 7 15:56:59] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1702] channel.c: Got a FRAME_CONTROL (-1) frame on channel Local/141@internal-a9e8,1 [Jul 7 15:56:59] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1702] channel.c: Bridge stops bridging channels SIP/360-00000001 and Local/141@internal-a9e8,1 [Jul 7 15:56:59] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1704] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1704] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1704] rtp.c: Setting the marker bit due to a source update [Jul 7 15:56:59] DEBUG[1673] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: Changing state for Local/141@internal - state 2 (In use) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:56:59] DEBUG[1679] 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 15:56:59] DEBUG[1673] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: Changing state for Local/141@internal - state 2 (In use) [Jul 7 15:56:59] DEBUG[1679] 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 15:56:59] DEBUG[1704] channel.c: Planning to masquerade channel SIP/oxo-00000002 into the structure of Local/141@internal-a9e8,1 [Jul 7 15:56:59] DEBUG[1704] channel.c: Done planning to masquerade channel SIP/oxo-00000002 into the structure of Local/141@internal-a9e8,1 [Jul 7 15:56:59] DEBUG[1704] chan_local.c: Not posting to queue since already masked on 'Local/141@internal-a9e8,2' [Jul 7 15:56:59] DEBUG[1702] channel.c: Actually Masquerading SIP/oxo-00000002(6) into the structure of Local/141@internal-a9e8,1(6) [Jul 7 15:56:59] DEBUG[1702] channel.c: Got clone lock for masquerade on 'SIP/oxo-00000002' at 0x1f2ba3b8 [Jul 7 15:56:59] DEBUG[1702] channel.c: Putting channel SIP/oxo-00000002 in 8/8 formats [Jul 7 15:56:59] DEBUG[1702] chan_sip.c: SIP Fixup: New owner for dialogue 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18: SIP/oxo-00000002 (Old parent: Local/141@internal-a9e8,1) [Jul 7 15:56:59] DEBUG[1702] channel.c: Released clone lock on 'Local/141@internal-a9e8,1' [Jul 7 15:56:59] DEBUG[1702] channel.c: Done Masquerading SIP/oxo-00000002 (6) [Jul 7 15:56:59] DEBUG[1704] channel.c: Didn't get a frame from channel: Local/141@internal-a9e8,2 [Jul 7 15:56:59] DEBUG[1704] channel.c: Bridge stops bridging channels Local/141@internal-a9e8,2 and Local/141@internal-a9e8,1 [Jul 7 15:56:59] DEBUG[1704] channel.c: Hanging up zombie 'Local/141@internal-a9e8,1' [Jul 7 15:56:59] DEBUG[1704] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:56:59] DEBUG[1704] rtp.c: Channel 'Local/141@internal-a9e8,2' has no RTP, not doing anything [Jul 7 15:56:59] DEBUG[1704] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 7 15:56:59] DEBUG[1704] pbx.c: Spawn extension (internal,141,1) exited non-zero on 'Local/141@internal-a9e8,2' [Jul 7 15:56:59] VERBOSE[1704] logger.c: [Jul 7 15:56:59] == Spawn extension (internal, 141, 1) exited non-zero on 'Local/141@internal-a9e8,2' [Jul 7 15:56:59] DEBUG[1704] channel.c: Soft-Hanging up channel 'Local/141@internal-a9e8,2' [Jul 7 15:56:59] DEBUG[1704] channel.c: Hanging up channel 'Local/141@internal-a9e8,2' [Jul 7 15:56:59] DEBUG[1704] devicestate.c: Notification of state change to be queued on device/channel Local/141@internal [Jul 7 15:56:59] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:56:59] DEBUG[1673] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: Changing state for Local/141@internal - state 1 (Not in use) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for Local - 141@internal [Jul 7 15:56:59] DEBUG[1679] 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 15:56:59] DEBUG[1673] chan_local.c: Checking if extension 141@internal exists (devicestate) [Jul 7 15:56:59] DEBUG[1673] devicestate.c: Changing state for Local/141@internal - state 1 (Not in use) [Jul 7 15:56:59] DEBUG[1679] 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 15:56:59] DEBUG[1702] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 15:56:59] DEBUG[1702] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 15:57:00] DEBUG[1702] rtp.c: Got RTCP report of 64 bytes [Jul 7 15:57:01] DEBUG[1703] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:01] DEBUG[1703] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:01] DEBUG[1702] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:01] DEBUG[1702] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:01] VERBOSE[1687] logger.c: [Jul 7 15:57:01] <--- 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=z9hG4bK7381159332013523960 From: ;tag=2021418042 To: "Valery Komarov" ;tag=as4d922c05 Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 CSeq: 1 BYE Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 0: BYE sip:140@192.168.1.18 SIP/2.0 (32) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK7381159332013523960 (68) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 2: From: ;tag=2021418042 (48) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 3: To: "Valery Komarov" ;tag=as4d922c05 (58) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 4: Call-ID: 1372afc0528c658242ea54576376136a@192.168.1.18 (54) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 5: CSeq: 1 BYE (11) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 7: User-Agent: Voip Phone 1.0 (26) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 8: Content-Length: 0 (17) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Header 9: (0) [Jul 7 15:57:01] VERBOSE[1687] logger.c: [Jul 7 15:57:01] --- (9 headers 0 lines) --- [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: = No match Their Call ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 Their Tag 6038a805cf482d5abeed1906e001e4d9 Our tag: as2cab833b [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: = Found Their Call ID: 1372afc0528c658242ea54576376136a@192.168.1.18 Their Tag 2021418042 Our tag: as4d922c05 [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 7 15:57:01] VERBOSE[1687] logger.c: [Jul 7 15:57:01] Sending to 192.168.5.13 : 5060 (no NAT) [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1372afc0528c658242ea54576376136a@192.168.1.18 [Jul 7 15:57:01] DEBUG[1687] chan_sip.c: Received bye, issuing owner hangup [Jul 7 15:57:01] VERBOSE[1687] logger.c: [Jul 7 15:57:01] <--- 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=z9hG4bK7381159332013523960;received=192.168.5.13 From: ;tag=2021418042 To: "Valery Komarov" ;tag=as4d922c05 Call-ID: 1372afc0528c658242ea54576376136a@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 15:57:01] DEBUG[1702] channel.c: Didn't get a frame from channel: SIP/360-00000001 [Jul 7 15:57:01] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:57:01] DEBUG[1702] channel.c: Bridge stops bridging channels SIP/360-00000001 and SIP/oxo-00000002 [Jul 7 15:57:01] DEBUG[1702] channel.c: Set channel SIP/oxo-00000000 to write format alaw [Jul 7 15:57:01] DEBUG[1702] rtp.c: Setting the marker bit due to a source update [Jul 7 15:57:01] VERBOSE[1702] logger.c: [Jul 7 15:57:01] -- Stopped music on hold on SIP/oxo-00000000 [Jul 7 15:57:01] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:01] DEBUG[1702] channel.c: Planning to masquerade channel SIP/oxo-00000000 into the structure of Transfered/SIP/oxo-00000000 [Jul 7 15:57:01] DEBUG[1702] channel.c: Done planning to masquerade channel SIP/oxo-00000000 into the structure of Transfered/SIP/oxo-00000000 [Jul 7 15:57:01] DEBUG[1702] channel.c: Actually Masquerading SIP/oxo-00000000(6) into the structure of Transfered/SIP/oxo-00000000(6) [Jul 7 15:57:01] DEBUG[1702] channel.c: Got clone lock for masquerade on 'SIP/oxo-00000000' at 0x1f289518 [Jul 7 15:57:01] DEBUG[1702] channel.c: Putting channel SIP/oxo-00000000 in 8/8 formats [Jul 7 15:57:01] DEBUG[1702] chan_sip.c: SIP Fixup: New owner for dialogue 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202: SIP/oxo-00000000 (Old parent: Transfered/SIP/oxo-00000000) [Jul 7 15:57:01] DEBUG[1702] channel.c: Driver for channel 'SIP/oxo-00000000' does not support indication 3, emulating it [Jul 7 15:57:01] DEBUG[1702] channel.c: Set channel SIP/oxo-00000000 to write format slin [Jul 7 15:57:01] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:57:01] DEBUG[1702] channel.c: Released clone lock on 'Transfered/SIP/oxo-00000000' [Jul 7 15:57:01] DEBUG[1702] channel.c: Done Masquerading SIP/oxo-00000000 (6) [Jul 7 15:57:01] DEBUG[1702] channel.c: Set channel SIP/oxo-00000002 to write format slin [Jul 7 15:57:01] DEBUG[1702] rtp.c: Difference is 1616, ms is 222 [Jul 7 15:57:01] DEBUG[1702] channel.c: Scheduling timer at 160 sample intervals [Jul 7 15:57:01] VERBOSE[1702] logger.c: [Jul 7 15:57:01] -- Playing 'beep' (language 'ru') [Jul 7 15:57:02] DEBUG[1702] channel.c: Scheduling timer at 22 sample intervals [Jul 7 15:57:02] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:02] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:02] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:02] DEBUG[1702] channel.c: Set channel SIP/oxo-00000002 to write format alaw [Jul 7 15:57:02] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:02] DEBUG[1702] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:02] DEBUG[1702] channel.c: Hanging up channel 'SIP/360-00000001' [Jul 7 15:57:02] DEBUG[1702] chan_sip.c: Hangup call SIP/360-00000001, SIP callid 1372afc0528c658242ea54576376136a@192.168.1.18) [Jul 7 15:57:02] DEBUG[1702] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 15:57:02] DEBUG[1702] rtp.c: Channel 'Transfered/SIP/oxo-00000000' has no RTP, not doing anything [Jul 7 15:57:02] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 15:57:02] DEBUG[1702] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 7 15:57:02] DEBUG[1673] chan_sip.c: Checking device state for peer 360 [Jul 7 15:57:02] DEBUG[1702] pbx.c: Spawn extension (oxo,360,1) exited non-zero on 'Transfered/SIP/oxo-00000000' [Jul 7 15:57:02] DEBUG[1673] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 15:57:02] DEBUG[1679] 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 15:57:02] VERBOSE[1702] logger.c: [Jul 7 15:57:02] == Spawn extension (oxo, 360, 1) exited non-zero on 'Transfered/SIP/oxo-00000000' [Jul 7 15:57:02] DEBUG[1702] channel.c: Soft-Hanging up channel 'Transfered/SIP/oxo-00000000' [Jul 7 15:57:02] DEBUG[1702] channel.c: Hanging up zombie 'Transfered/SIP/oxo-00000000' [Jul 7 15:57:02] DEBUG[1702] devicestate.c: Notification of state change to be queued on device/channel Transfered/SIP/oxo [Jul 7 15:57:02] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for Transfered - SIP/oxo [Jul 7 15:57:02] DEBUG[1673] devicestate.c: Changing state for Transfered/SIP/oxo - state 4 (Invalid) [Jul 7 15:57:02] DEBUG[1706] channel.c: Set channel SIP/oxo-00000000 to write format alaw [Jul 7 15:57:02] DEBUG[1679] 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 15:57:02] DEBUG[1706] channel.c: Scheduling timer at 0 sample intervals [Jul 7 15:57:02] DEBUG[1706] rtp.c: Setting the marker bit due to a source update [Jul 7 15:57:02] DEBUG[1706] rtp.c: Setting the marker bit due to a source update [Jul 7 15:57:02] VERBOSE[1687] logger.c: [Jul 7 15:57:02] Really destroying SIP dialog '1372afc0528c658242ea54576376136a@192.168.1.18' Method: BYE [Jul 7 15:57:03] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:03] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:03] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:03] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:06] DEBUG[1706] rtp.c: Got RTCP report of 36 bytes [Jul 7 15:57:06] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:06] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:06] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:08] DEBUG[1706] rtp.c: Got RTCP report of 36 bytes [Jul 7 15:57:08] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:08] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:08] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:11] DEBUG[1706] rtp.c: Got RTCP report of 36 bytes [Jul 7 15:57:11] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:11] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:11] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:13] DEBUG[1687] chan_sip.c: Auto destroying SIP dialog '107402794129859-2893711712540@192.168.5.13' [Jul 7 15:57:13] DEBUG[1687] chan_sip.c: Destroying SIP dialog 107402794129859-2893711712540@192.168.5.13 [Jul 7 15:57:13] VERBOSE[1687] logger.c: [Jul 7 15:57:13] Really destroying SIP dialog '107402794129859-2893711712540@192.168.5.13' Method: REGISTER [Jul 7 15:57:13] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:13] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:14] DEBUG[1706] rtp.c: Got RTCP report of 60 bytes [Jul 7 15:57:14] DEBUG[1706] rtp.c: Got RTCP report of 56 bytes [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] <--- 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=as3bd16478 From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116606 BYE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKd1c12cff6f7c0cc940851fd7c4075cf7 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 0: BYE sip:360@192.168.1.18 SIP/2.0 (32) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 1: Route: (28) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 2: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 3: To: sip:360@isoemo.com;tag=as3bd16478 (37) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 4: From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 (64) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 5: Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 (55) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 6: CSeq: 1585116606 BYE (20) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 7: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKd1c12cff6f7c0cc940851fd7c4075cf7 (83) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 10: (0) [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] --- (10 headers 0 lines) --- [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: = No match Their Call ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 Their Tag 6038a805cf482d5abeed1906e001e4d9 Our tag: as2cab833b [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 Their Tag 0562b16854491908ed4104b44b7d0670 Our tag: as3bd16478 [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] Sending to 192.168.0.202 : 5060 (NAT) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Received bye, issuing owner hangup [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKd1c12cff6f7c0cc940851fd7c4075cf7;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=0562b16854491908ed4104b44b7d0670 To: sip:360@isoemo.com;tag=as3bd16478 Call-ID: 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202 CSeq: 1585116606 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jul 7 15:57:14] DEBUG[1706] channel.c: Didn't get a frame from channel: SIP/oxo-00000000 [Jul 7 15:57:14] DEBUG[1706] rtp.c: Setting the marker bit due to a source update [Jul 7 15:57:14] DEBUG[1706] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/oxo-00000002 [Jul 7 15:57:14] DEBUG[1706] channel.c: Hanging up channel 'SIP/oxo-00000002' [Jul 7 15:57:14] DEBUG[1706] chan_sip.c: Hangup call SIP/oxo-00000002, SIP callid 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18) [Jul 7 15:57:14] VERBOSE[1706] logger.c: [Jul 7 15:57:14] Scheduling destruction of SIP dialog '7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18' in 6400 ms (Method: INVITE) [Jul 7 15:57:14] DEBUG[1706] chan_sip.c: Strict routing enforced for session 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 [Jul 7 15:57:14] VERBOSE[1706] logger.c: [Jul 7 15:57:14] set_destination: Parsing for address/port to send to [Jul 7 15:57:14] VERBOSE[1706] logger.c: [Jul 7 15:57:14] set_destination: set destination to 192.168.0.202, port 5060 [Jul 7 15:57:14] VERBOSE[1706] logger.c: [Jul 7 15:57:14] 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=z9hG4bK1d59e2d7;rport From: "360" ;tag=as2cab833b To: ;tag=6038a805cf482d5abeed1906e001e4d9 Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@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 15:57:14] DEBUG[1706] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 15:57:14] DEBUG[1706] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:57:14] DEBUG[1706] channel.c: Hanging up channel 'SIP/oxo-00000000' [Jul 7 15:57:14] DEBUG[1706] chan_sip.c: Hangup call SIP/oxo-00000000, SIP callid 7261abed7bd3c20ab4ef0421ab828395@192.168.0.202) [Jul 7 15:57:14] DEBUG[1706] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] Really destroying SIP dialog '7261abed7bd3c20ab4ef0421ab828395@192.168.0.202' Method: BYE [Jul 7 15:57:14] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:57:14] DEBUG[1673] chan_sip.c: Checking device state for peer oxo [Jul 7 15:57:14] DEBUG[1673] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:57:14] DEBUG[1673] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 15:57:14] DEBUG[1679] 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 15:57:14] DEBUG[1673] chan_sip.c: Checking device state for peer oxo [Jul 7 15:57:14] DEBUG[1673] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 15:57:14] DEBUG[1679] 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 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] <--- 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=6038a805cf482d5abeed1906e001e4d9 From: "360" ;tag=as2cab833b Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK1d59e2d7;rport=5060 Content-Length: 0 <-------------> [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 15:57:14] DEBUG[1687] 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 15:57:14] DEBUG[1687] chan_sip.c: Header 2: Supported: 100rel,100rel (24) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 4: To: ;tag=6038a805cf482d5abeed1906e001e4d9 (64) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 5: From: "360" ;tag=as2cab833b (49) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 6: Call-ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 (54) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 7: CSeq: 103 BYE (13) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.1.18:5060;received=192.168.1.18;branch=z9hG4bK1d59e2d7;rport=5060 (90) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Header 10: (0) [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] --- (10 headers 0 lines) --- [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: = Found Their Call ID: 7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18 Their Tag 6038a805cf482d5abeed1906e001e4d9 Our tag: as2cab833b [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 [Jul 7 15:57:14] DEBUG[1687] chan_sip.c: Stopping retransmission on '7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18' of Request 103: Match Found [Jul 7 15:57:14] VERBOSE[1687] logger.c: [Jul 7 15:57:14] Really destroying SIP dialog '7f69bed92e60d0de259bdd2954c96b3f@192.168.1.18' Method: INVITE [Jul 7 15:57:15] VERBOSE[1701] logger.c: [Jul 7 15:57:15] Beginning asterisk shutdown.... [Jul 7 15:57:15] VERBOSE[1701] logger.c: [Jul 7 15:57:15] Executing last minute cleanups [Jul 7 15:57:15] VERBOSE[1701] logger.c: [Jul 7 15:57:15] == Destroying musiconhold processes [Jul 7 15:57:15] DEBUG[1701] res_musiconhold.c: Destroying MOH class 'default' [Jul 7 15:57:15] VERBOSE[1701] logger.c: [Jul 7 15:57:15] Asterisk cleanly ending (0). [Jul 7 15:57:15] DEBUG[1701] asterisk.c: Asterisk ending (0).