[Jul 7 16:23:01] VERBOSE[1994] logger.c: [Jul 7 16:23:01] Asterisk Event Logger restarted [Jul 7 16:23:01] VERBOSE[1994] logger.c: [Jul 7 16:23:01] Asterisk Queue Logger restarted [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] <--- SIP read from 192.168.0.202:5060 ---> INVITE sip:361@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:361@isoemo.com From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd Contact: "Valery Komarov" Content-Type: application/sdp Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899159 INVITE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK37aa964fe22df7185dde30082371da6d Max-Forwards: 70 Content-Length: 251 v=0 o=default 1278505394 1278505394 IN IP4 192.168.0.202 s=- c=IN IP4 192.168.0.202 t=0 0 m=audio 32004 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 16:23:03] DEBUG[2010] chan_sip.c: Header 0: INVITE sip:361@isoemo.com SIP/2.0 (33) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 1: Route: (28) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 2: Supported: 100rel,timer (23) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 4: Session-Expires: 43200 (22) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 5: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 6: To: sip:361@isoemo.com (22) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 7: From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd (83) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 8: Contact: "Valery Komarov" (63) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 10: Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 (55) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 11: CSeq: 2103899159 INVITE (23) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK37aa964fe22df7185dde30082371da6d (83) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 13: Max-Forwards: 70 (16) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 14: Content-Length: 251 (19) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Header 15: (0) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: o=default 1278505394 1278505394 IN IP4 192.168.0.202 (52) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: s=- (3) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: m=audio 32004 RTP/AVP 18 106 4 8 0 (34) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: a=sendrecv (10) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: a=fmtp:4 annexa=no (18) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Line: a=maxptime:90 (13) [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] --- (15 headers 12 lines) --- [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: = No match Their Call ID: NTdmNmE2MTJkNTY4YzJjMWVlMTBiZTM3ODM2MjE2MTI. Their Tag dbe3820e Our tag: as4bbc820a [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: = No match Their Call ID: 144122969413291-60142881020887@192.168.5.13 Their Tag 285257750 Our tag: as69d50b15 [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: = No match Their Call ID: 310581660323243-12532222309725@192.168.5.13 Their Tag 2392520596 Our tag: as309c8f06 [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Setting NAT on RTP to Off [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Allocating new SIP dialog for 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 - INVITE (With RTP) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Begin: parsing SIP "Supported: 100rel,timer" [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Found SIP option: -100rel- [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Matched SIP option: 100rel [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Found SIP option: -timer- [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Matched SIP option: timer [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Sending to 192.168.0.202 : 5060 (no NAT) [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Using INVITE request as basis request - 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found peer 'oxo' [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Setting NAT on RTP to Off [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing session-level SDP o=default 1278505394 1278505394 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found RTP audio format 18 [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found RTP audio format 106 [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found RTP audio format 4 [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found RTP audio format 8 [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found RTP audio format 0 [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Found audio description format telephone-event for ID 106 [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:4 annexa=no... UNSUPPORTED. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=maxptime:90... UNSUPPORTED. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: T38 state changed to 0 on channel [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Peer audio RTP is at port 192.168.0.202:32004 [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Checking SIP call limits for device [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: Updating call counter for incoming call [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] Looking for 361 in oxo (domain isoemo.com) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: This channel will not be able to handle video. [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: build_route: Contact hop: "Valery Komarov" [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] list_route: hop: [Jul 7 16:23:03] DEBUG[2010] chan_sip.c: SIP/oxo-00000000: New call is still down.... Trying... [Jul 7 16:23:03] VERBOSE[2010] logger.c: [Jul 7 16:23:03] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK37aa964fe22df7185dde30082371da6d;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd To: sip:361@isoemo.com Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899159 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 16:23:03] DEBUG[2010] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 16:23:03] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 16:23:03] DEBUG[1996] chan_sip.c: Checking device state for peer oxo [Jul 7 16:23:03] DEBUG[1996] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 16:23:03] DEBUG[2002] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:03] DEBUG[2025] pbx.c: Launching 'Dial' [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] -- Executing [361@oxo:1] Dial("SIP/oxo-00000000", "SIP/361|60|t") in new stack [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Setting NAT on RTP to Off [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: This channel will not be able to handle video. [Jul 7 16:23:03] DEBUG[2025] rtp.c: Seeded SDP of 'SIP/361-00000001' with that of 'SIP/oxo-00000000' [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable DIALEDTIME. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable ANSWEREDTIME. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable DIALEDPEERNAME. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable DIALSTATUS. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable SIPCALLID. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable SIPUSERAGENT. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable SIPDOMAIN. [Jul 7 16:23:03] DEBUG[2025] channel.c: Not copying variable SIPURI. [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Outgoing Call for 361 [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Updating call counter for outgoing call [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] Audio is at 192.168.1.18 port 15808 [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] Adding codec 0x8 (alaw) to SDP [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] Adding codec 0x4 (ulaw) to SDP [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 0: INVITE sip:361@192.168.1.41:58978;rinstance=da0ffef137fc5df6;transport=udp SIP/2.0 (82) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport (63) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as3f4930a0 (60) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 3: To: (73) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 4: Contact: (31) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 5: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 9: Date: Wed, 07 Jul 2010 12:23:03 GMT (35) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 11: Supported: replaces (19) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 13: Content-Length: 235 (19) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Header 14: (0) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: o=root 1994 1994 IN IP4 192.168.1.18 (36) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: s=session (9) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: c=IN IP4 192.168.1.18 (21) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: m=audio 15808 RTP/AVP 8 0 106 (29) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: a=fmtp:106 0-16 (15) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 16:23:03] DEBUG[2025] chan_sip.c: Line: a=sendrecv (10) [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] Reliably Transmitting (no NAT) to 192.168.1.41:58978: INVITE sip:361@192.168.1.41:58978;rinstance=da0ffef137fc5df6;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport From: "Valery Komarov" ;tag=as3f4930a0 To: Contact: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Jul 2010 12:23:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=root 1994 1994 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 15808 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 16:23:03] DEBUG[2025] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:03] VERBOSE[2025] logger.c: [Jul 7 16:23:03] -- Called 361 [Jul 7 16:23:04] VERBOSE[2010] logger.c: [Jul 7 16:23:04] <--- SIP read from 192.168.1.41:58978 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport=5060 To: From: "Valery Komarov" ;tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 102 INVITE Content-Length: 0 <-------------> [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport=5060 (68) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 2: To: (73) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 3: From: "Valery Komarov" ;tag=as3f4930a0 (60) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 4: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 6: Content-Length: 0 (17) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 7: (0) [Jul 7 16:23:04] VERBOSE[2010] logger.c: [Jul 7 16:23:04] --- (7 headers 0 lines) --- [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag Our tag: as3f4930a0 [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: *** SIP TIMER: Cancelling retransmission #31 - INVITE (got response) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' Request 102: Found [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: SIP response 100 to standard invite [Jul 7 16:23:04] VERBOSE[2010] logger.c: [Jul 7 16:23:04] <--- SIP read from 192.168.1.41:58978 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport=5060 Contact: To: ;tag=5982bf54 From: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 102 INVITE User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Allow-Events: hold, talk Content-Length: 0 <-------------> [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport=5060 (68) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 2: Contact: (78) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=5982bf54 (86) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 4: From: "Valery Komarov";tag=as3f4930a0 (59) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 5: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 7: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 8: Allow-Events: hold, talk (24) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: Header 10: (0) [Jul 7 16:23:04] VERBOSE[2010] logger.c: [Jul 7 16:23:04] --- (10 headers 0 lines) --- [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag Our tag: as3f4930a0 [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' Request 102: Found [Jul 7 16:23:04] DEBUG[2010] chan_sip.c: SIP response 180 to standard invite [Jul 7 16:23:04] DEBUG[2010] devicestate.c: Notification of state change to be queued on device/channel SIP/361 [Jul 7 16:23:04] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 7 16:23:04] DEBUG[1996] chan_sip.c: Checking device state for peer 361 [Jul 7 16:23:04] DEBUG[1996] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 7 16:23:04] DEBUG[2002] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:04] VERBOSE[2025] logger.c: [Jul 7 16:23:04] -- SIP/361-00000001 is ringing [Jul 7 16:23:04] DEBUG[2025] rtp.c: Setting early bridge SDP of 'SIP/oxo-00000000' with that of 'SIP/361-00000001' [Jul 7 16:23:04] VERBOSE[2025] logger.c: [Jul 7 16:23:04] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK37aa964fe22df7185dde30082371da6d;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd To: sip:361@isoemo.com;tag=as646f20fc Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899159 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 16:23:05] DEBUG[2025] rtp.c: Got RTCP report of 132 bytes [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] <--- SIP read from 192.168.1.41:58978 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport=5060 Contact: To: ;tag=5982bf54 From: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 327 v=0 o=- 12922978985501815 12922978985501815 IN IP4 192.168.1.41 s=Counterpath Bria 3.0 c=IN IP4 192.168.1.41 t=0 0 m=audio 61130 RTP/AVP 8 0 106 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.41 61130 typ host a=candidate:1 2 UDP 659134 192.168.1.41 61131 typ host <-------------> [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK74cd2542;rport=5060 (68) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 2: Contact: (78) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=5982bf54 (86) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 4: From: "Valery Komarov";tag=as3f4930a0 (59) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 5: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 8: Content-Type: application/sdp (29) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 9: Supported: eventlist (20) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 10: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 11: Content-Length: 327 (19) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 12: (0) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: o=- 12922978985501815 12922978985501815 IN IP4 192.168.1.41 (59) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: s=Counterpath Bria 3.0 (22) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: c=IN IP4 192.168.1.41 (21) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: m=audio 61130 RTP/AVP 8 0 106 (29) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: a=sendrecv (10) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: a=candidate:1 1 UDP 659136 192.168.1.41 61130 typ host (54) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Line: a=candidate:1 2 UDP 659134 192.168.1.41 61131 typ host (54) [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] --- (12 headers 11 lines) --- [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Acked pending invite 102 [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Stopping retransmission on '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' of Request 102: Match Found [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: SIP response 200 to standard invite [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing session-level SDP o=- 12922978985501815 12922978985501815 IN IP4 192.168.1.41... UNSUPPORTED. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing session-level SDP s=Counterpath Bria 3.0... UNSUPPORTED. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.41... OK. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Found RTP audio format 8 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Found RTP audio format 0 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Found RTP audio format 106 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Found audio description format telephone-event for ID 106 [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=candidate:1 1 UDP 659136 192.168.1.41 61130 typ host... UNSUPPORTED. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=candidate:1 2 UDP 659134 192.168.1.41 61131 typ host... UNSUPPORTED. [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: T38 state changed to 0 on channel SIP/361-00000001 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Peer audio RTP is at port 192.168.1.41:61130 [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Updating call counter for outgoing call [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: build_route: Contact hop: [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] list_route: hop: [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Strict routing enforced for session 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] set_destination: Parsing for address/port to send to [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] set_destination: set destination to 192.168.1.41, port 58978 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] Transmitting (no NAT) to 192.168.1.41:58978: ACK sip:361@192.168.1.41:58978;rinstance=da0ffef137fc5df6;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK551efbe7;rport From: "Valery Komarov" ;tag=as3f4930a0 To: ;tag=5982bf54 Contact: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jul 7 16:23:05] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/361 [Jul 7 16:23:05] VERBOSE[2025] logger.c: [Jul 7 16:23:05] -- SIP/361-00000001 answered SIP/oxo-00000000 [Jul 7 16:23:05] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: SIP answering channel: SIP/oxo-00000000 [Jul 7 16:23:05] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: Setting framing from config on incoming call [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 7 16:23:05] VERBOSE[2025] logger.c: [Jul 7 16:23:05] Audio is at 192.168.1.18 port 16984 [Jul 7 16:23:05] VERBOSE[2025] logger.c: [Jul 7 16:23:05] Adding codec 0x8 (alaw) to SDP [Jul 7 16:23:05] VERBOSE[2025] logger.c: [Jul 7 16:23:05] Adding codec 0x4 (ulaw) to SDP [Jul 7 16:23:05] VERBOSE[2025] logger.c: [Jul 7 16:23:05] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 16:23:05] VERBOSE[2025] logger.c: [Jul 7 16:23:05] <--- Reliably Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bK37aa964fe22df7185dde30082371da6d;received=192.168.0.202;rport=5060 From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd To: sip:361@isoemo.com;tag=as646f20fc Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899159 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 1994 1994 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 16984 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 16:23:05] DEBUG[2025] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:05] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:05] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:05] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 7 16:23:05] DEBUG[1996] chan_sip.c: Checking device state for peer 361 [Jul 7 16:23:05] DEBUG[1996] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 7 16:23:05] DEBUG[2002] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:05] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 16:23:05] DEBUG[1996] chan_sip.c: Checking device state for peer oxo [Jul 7 16:23:05] DEBUG[1996] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 16:23:05] DEBUG[2002] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:05] DEBUG[2025] chan_sip.c: Oooh, format changed to 4 [Jul 7 16:23:05] DEBUG[2025] channel.c: Set channel SIP/361-00000001 to read format alaw [Jul 7 16:23:05] DEBUG[2025] channel.c: Set channel SIP/361-00000001 to write format alaw [Jul 7 16:23:05] DEBUG[2025] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 16:23:05] DEBUG[2025] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 16:23:05] DEBUG[2025] rtp.c: Ooh, format changed from unknown to ulaw [Jul 7 16:23:05] DEBUG[2025] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] <--- SIP read from 192.168.0.202:5060 ---> ACK sip:361@192.168.1.18 SIP/2.0 Route: Contact: "Valery Komarov" User-Agent: OxO_GW_700/013.001 To: sip:361@isoemo.com;tag=as646f20fc From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899159 ACK Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK2b9e77e2983bafae57eb1a44ebf20b4f Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 0: ACK sip:361@192.168.1.18 SIP/2.0 (32) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 1: Route: (28) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 2: Contact: "Valery Komarov" (49) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 4: To: sip:361@isoemo.com;tag=as646f20fc (37) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 5: From: "Valery Komarov" ;tag=54fbbc9dc7848b73526ce354fa6a8acd (83) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 6: Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 (55) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 7: CSeq: 2103899159 ACK (20) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bK2b9e77e2983bafae57eb1a44ebf20b4f (83) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 10: Content-Length: 0 (17) [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Header 11: (0) [Jul 7 16:23:05] VERBOSE[2010] logger.c: [Jul 7 16:23:05] --- (11 headers 0 lines) --- [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: = No match Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: = Found Their Call ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 Their Tag 54fbbc9dc7848b73526ce354fa6a8acd Our tag: as646f20fc [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Jul 7 16:23:05] DEBUG[2010] chan_sip.c: Stopping retransmission on '01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202' of Response 2103899159: Match Found [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- SIP read from 192.168.0.202:5060 ---> INVITE sip:361@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:361@isoemo.com;tag=as646f20fc From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899160 INVITE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKa3c40acb19506b4e0a50de6b743b494b Max-Forwards: 70 Content-Length: 215 v=0 o=default 1278505394 1278505395 IN IP4 192.168.0.202 s=- c=IN IP4 192.168.0.202 t=0 0 m=audio 32004 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 16:23:06] DEBUG[2010] chan_sip.c: Header 0: INVITE sip:361@192.168.1.18 SIP/2.0 (35) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 1: Route: (28) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 2: Supported: 100rel,timer (23) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 4: Session-Expires: 43200;refresher=uac (36) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 5: P-Asserted-Identity: "Valery Komarov" (61) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 6: Contact: "Valery Komarov" (63) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 7: Content-Type: application/sdp (29) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 8: To: sip:361@isoemo.com;tag=as646f20fc (37) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 9: From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd (64) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 10: Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 (55) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 11: CSeq: 2103899160 INVITE (23) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 12: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKa3c40acb19506b4e0a50de6b743b494b (83) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 13: Max-Forwards: 70 (16) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 14: Content-Length: 215 (19) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 15: (0) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: o=default 1278505394 1278505395 IN IP4 192.168.0.202 (52) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: s=- (3) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: c=IN IP4 192.168.0.202 (22) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: m=audio 32004 RTP/AVP 8 106 (27) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=sendrecv (10) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=maxptime:90 (13) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] --- (15 headers 11 lines) --- [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: = No match Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: = Found Their Call ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 Their Tag 54fbbc9dc7848b73526ce354fa6a8acd Our tag: as646f20fc [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Sending to 192.168.0.202 : 5060 (NAT) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP o=default 1278505394 1278505395 IN IP4 192.168.0.202... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.202... OK. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found RTP audio format 8 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found RTP audio format 106 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found audio description format telephone-event for ID 106 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=maxptime:90... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: T38 state changed to 0 on channel SIP/oxo-00000000 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Peer audio RTP is at port 192.168.0.202:32004 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Got a SIP re-invite for call 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: SIP/oxo-00000000: This call is UP.... [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKa3c40acb19506b4e0a50de6b743b494b;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd To: sip:361@isoemo.com;tag=as646f20fc Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Setting framing from config on incoming call [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Audio is at 192.168.1.18 port 16984 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Adding codec 0x8 (alaw) to SDP [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- Reliably Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKa3c40acb19506b4e0a50de6b743b494b;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd To: sip:361@isoemo.com;tag=as646f20fc Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 1994 1995 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 16984 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 16:23:06] DEBUG[2010] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:06] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- SIP read from 192.168.0.202:5060 ---> ACK sip:361@192.168.1.18 SIP/2.0 Route: Contact: "Valery Komarov" User-Agent: OxO_GW_700/013.001 To: sip:361@isoemo.com;tag=as646f20fc From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899160 ACK Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKe47c8bf0f60f7f5f6db85066f63659bf Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 0: ACK sip:361@192.168.1.18 SIP/2.0 (32) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 1: Route: (28) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 2: Contact: "Valery Komarov" (49) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 3: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 4: To: sip:361@isoemo.com;tag=as646f20fc (37) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 5: From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd (64) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 6: Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 (55) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 7: CSeq: 2103899160 ACK (20) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 8: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKe47c8bf0f60f7f5f6db85066f63659bf (83) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 10: Content-Length: 0 (17) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 11: (0) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] --- (11 headers 0 lines) --- [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: = No match Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: = Found Their Call ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 Their Tag 54fbbc9dc7848b73526ce354fa6a8acd Our tag: as646f20fc [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #37 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Stopping retransmission on '01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202' of Response 2103899160: Match Found [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- SIP read from 192.168.1.41:58978 ---> INVITE sip:140@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-fa2eeeef1e7cec0c-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Valery Komarov";tag=as3f4930a0 From: ;tag=5982bf54 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 327 v=0 o=- 12922978986576137 12922978986576137 IN IP4 192.168.1.41 s=Counterpath Bria 3.0 c=IN IP4 192.168.1.41 t=0 0 m=audio 61130 RTP/AVP 8 0 106 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-15 a=sendonly a=candidate:1 1 UDP 659136 192.168.1.41 61130 typ host a=candidate:1 2 UDP 659134 192.168.1.41 61131 typ host <-------------> [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 0: INVITE sip:140@192.168.1.18 SIP/2.0 (35) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-fa2eeeef1e7cec0c-1---d8754z-;rport (92) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 3: Contact: (78) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 4: To: "Valery Komarov";tag=as3f4930a0 (57) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 5: From: ;tag=5982bf54 (88) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 6: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 7: CSeq: 2 INVITE (14) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 10: Supported: eventlist (20) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 11: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 12: Content-Length: 327 (19) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 13: (0) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: o=- 12922978986576137 12922978986576137 IN IP4 192.168.1.41 (59) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: s=Counterpath Bria 3.0 (22) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: c=IN IP4 192.168.1.41 (21) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: m=audio 61130 RTP/AVP 8 0 106 (29) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=sendonly (10) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=candidate:1 1 UDP 659136 192.168.1.41 61130 typ host (54) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Line: a=candidate:1 2 UDP 659134 192.168.1.41 61131 typ host (54) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] --- (13 headers 11 lines) --- [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Begin: parsing SIP "Supported: eventlist" [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Found SIP option: -eventlist- [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Matched SIP option: eventlist [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Sending to 192.168.1.41 : 58978 (no NAT) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP o=- 12922978986576137 12922978986576137 IN IP4 192.168.1.41... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP s=Counterpath Bria 3.0... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.41... OK. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found RTP audio format 8 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found RTP audio format 0 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found RTP audio format 106 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Found audio description format telephone-event for ID 106 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=candidate:1 1 UDP 659136 192.168.1.41 61130 typ host... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=candidate:1 2 UDP 659134 192.168.1.41 61131 typ host... UNSUPPORTED. [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: T38 state changed to 0 on channel SIP/361-00000001 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Peer audio RTP is at port 192.168.1.41:61130 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Got a SIP re-invite for call 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: SIP/361-00000001: This call is UP.... [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- Transmitting (no NAT) to 192.168.1.41:58978 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-fa2eeeef1e7cec0c-1---d8754z-;received=192.168.1.41;rport=58978 From: ;tag=5982bf54 To: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Setting framing from config on incoming call [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Audio is at 192.168.1.18 port 15808 [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Adding codec 0x8 (alaw) to SDP [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Adding codec 0x4 (ulaw) to SDP [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- Reliably Transmitting (no NAT) to 192.168.1.41:58978 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-fa2eeeef1e7cec0c-1---d8754z-;received=192.168.1.41;rport=58978 From: ;tag=5982bf54 To: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 1994 1995 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 15808 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=recvonly <------------> [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:06] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:06] VERBOSE[2025] logger.c: [Jul 7 16:23:06] -- Started music on hold, class 'default', on SIP/oxo-00000000 [Jul 7 16:23:06] DEBUG[2025] channel.c: Scheduling timer at 160 sample intervals [Jul 7 16:23:06] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] <--- SIP read from 192.168.1.41:58978 ---> ACK sip:140@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-7fb75a04252fc077-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Valery Komarov";tag=as3f4930a0 From: ;tag=5982bf54 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 2 ACK User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 0 <-------------> [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 0: ACK sip:140@192.168.1.18 SIP/2.0 (32) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-7fb75a04252fc077-1---d8754z-;rport (92) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 3: Contact: (78) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 4: To: "Valery Komarov";tag=as3f4930a0 (57) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 5: From: ;tag=5982bf54 (88) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 6: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 7: CSeq: 2 ACK (11) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 8: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Header 10: (0) [Jul 7 16:23:06] VERBOSE[2010] logger.c: [Jul 7 16:23:06] --- (10 headers 0 lines) --- [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 [Jul 7 16:23:06] DEBUG[2010] chan_sip.c: Stopping retransmission on '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' of Response 2: Match Found [Jul 7 16:23:06] DEBUG[2025] channel.c: Set channel SIP/oxo-00000000 to write format slin [Jul 7 16:23:06] DEBUG[2025] res_musiconhold.c: SIP/oxo-00000000 Opened file 1 '/var/lib/asterisk/moh/03-Chet Atkins-Boo Boo Stick Beat' [Jul 7 16:23:06] WARNING[2025] mp3/interface.c: Junk at the beginning of frame 49443303 [Jul 7 16:23:07] DEBUG[2025] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:23:07] DEBUG[2025] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:23:09] DEBUG[2025] rtp.c: Got RTCP report of 176 bytes [Jul 7 16:23:10] DEBUG[2025] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:23:10] DEBUG[2025] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- SIP read from 192.168.1.41:58978 ---> REFER sip:140@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-ae1e3f47a1bf21ea-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Valery Komarov";tag=as3f4930a0 From: ;tag=5982bf54 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 3 REFER User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Refer-To: Referred-By: Content-Length: 0 <-------------> [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 0: REFER sip:140@192.168.1.18 SIP/2.0 (34) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-ae1e3f47a1bf21ea-1---d8754z-;rport (92) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 3: Contact: (78) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 4: To: "Valery Komarov";tag=as3f4930a0 (57) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 5: From: ;tag=5982bf54 (88) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 6: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 7: CSeq: 3 REFER (13) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 8: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 9: Refer-To: (30) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 10: Referred-By: (82) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 11: Content-Length: 0 (17) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 12: (0) [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] --- (12 headers 0 lines) --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] Call 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 got a SIP call transfer from caller: (REFER)! [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] SIP transfer to extension 360@internal by 361@192.168.1.41:58978 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: This SIP transfer is local : isoemo.com [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: SIP blind transfer: Transferer channel SIP/361-00000001, transferee channel SIP/oxo-00000000 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/oxo-00000000' [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- Transmitting (no NAT) to 192.168.1.41:58978 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-ae1e3f47a1bf21ea-1---d8754z-;received=192.168.1.41;rport=58978 From: ;tag=5982bf54 To: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 3 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: chan1->name: SIP/361-00000001 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Strict routing enforced for session 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] set_destination: Parsing for address/port to send to [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] set_destination: set destination to 192.168.1.41, port 58978 [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] Reliably Transmitting (no NAT) to 192.168.1.41:58978: NOTIFY sip:361@192.168.1.41:58978;rinstance=da0ffef137fc5df6;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK780298b0;rport From: "Valery Komarov";tag=as3f4930a0 To: ;tag=5982bf54 Contact: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=3 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 21 SIP/2.0 183 Ringing --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:10] DEBUG[2010] channel.c: Soft-Hanging up channel 'SIP/oxo-00000000' [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Blind transfer succeeded. Telling transferer. [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Strict routing enforced for session 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] set_destination: Parsing for address/port to send to [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] set_destination: set destination to 192.168.1.41, port 58978 [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] Reliably Transmitting (no NAT) to 192.168.1.41:58978: NOTIFY sip:361@192.168.1.41:58978;rinstance=da0ffef137fc5df6;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK05db8eab;rport From: "Valery Komarov";tag=as3f4930a0 To: ;tag=5982bf54 Contact: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=3 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 16 SIP/2.0 200 Ok --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] -- Stopped music on hold on SIP/oxo-00000000 [Jul 7 16:23:10] DEBUG[2025] channel.c: Set channel SIP/oxo-00000000 to write format alaw [Jul 7 16:23:10] DEBUG[2025] channel.c: Scheduling timer at 0 sample intervals [Jul 7 16:23:10] DEBUG[2025] channel.c: Didn't get a frame from channel: SIP/oxo-00000000 [Jul 7 16:23:10] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:10] DEBUG[2025] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/361-00000001 [Jul 7 16:23:10] DEBUG[2025] channel.c: Hanging up channel 'SIP/361-00000001' [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: update_call_counter(361) - decrement call limit counter on hangup [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Updating call counter for outgoing call [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Call to peer '361' removed from call limit 0 [Jul 7 16:23:10] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/361 [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18. [Jul 7 16:23:10] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 7 16:23:10] DEBUG[1996] chan_sip.c: Checking device state for peer 361 [Jul 7 16:23:10] DEBUG[1996] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 7 16:23:10] DEBUG[2002] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] Scheduling destruction of SIP dialog '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' in 6400 ms (Method: REFER) [Jul 7 16:23:10] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/361 [Jul 7 16:23:10] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 7 16:23:10] DEBUG[1996] chan_sip.c: Checking device state for peer 361 [Jul 7 16:23:10] DEBUG[1996] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 7 16:23:10] DEBUG[2002] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:10] DEBUG[2025] rtp.c: Channel '' has no RTP, not doing anything [Jul 7 16:23:10] DEBUG[2025] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 7 16:23:10] DEBUG[2025] pbx.c: Spawn extension (internal,360,0) exited non-zero on 'SIP/oxo-00000000' [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] == Spawn extension (internal, 360, 0) exited non-zero on 'SIP/oxo-00000000' [Jul 7 16:23:10] DEBUG[2025] pbx.c: Launching 'Dial' [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] -- Executing [360@internal:1] Dial("SIP/oxo-00000000", "SIP/360|60|tT") in new stack [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Setting NAT on RTP to Off [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: This channel will not be able to handle video. [Jul 7 16:23:10] DEBUG[2025] rtp.c: Seeded SDP of 'SIP/360-00000002' with that of 'SIP/oxo-00000000' [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable DIALEDTIME. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable ANSWEREDTIME. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable DIALEDPEERNAME. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable DIALSTATUS. [Jul 7 16:23:10] DEBUG[2025] channel.c: Copying soft-transferable variable SIPTRANSFER_REFERER. [Jul 7 16:23:10] DEBUG[2025] channel.c: Copying soft-transferable variable SIPTRANSFER. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable SIPDOMAIN. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable BLINDTRANSFER. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable BRIDGEPEER. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable SIPCALLID. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable SIPUSERAGENT. [Jul 7 16:23:10] DEBUG[2025] channel.c: Not copying variable SIPURI. [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Call for 360 transfered by 361@192.168.1.41:58978 [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Outgoing Call for 360 [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Updating call counter for outgoing call [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] Audio is at 192.168.1.18 port 13694 [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] Adding codec 0x8 (alaw) to SDP [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] Adding codec 0x4 (ulaw) to SDP [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] Adding non-codec 0x1 (telephone-event) to SDP [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: -- Done with adding codecs to SDP [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 0: INVITE sip:360@192.168.5.13:5060 SIP/2.0 (40) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport (63) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as2daecde9 (60) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 3: To: (31) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 4: Contact: (31) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 5: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 9: Date: Wed, 07 Jul 2010 12:23:10 GMT (35) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 11: Supported: replaces (19) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 13: Content-Length: 235 (19) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Header 14: (0) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: o=root 1994 1994 IN IP4 192.168.1.18 (36) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: s=session (9) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: c=IN IP4 192.168.1.18 (21) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: m=audio 13694 RTP/AVP 8 0 106 (29) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: a=fmtp:106 0-16 (15) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: a=ptime:20 (10) [Jul 7 16:23:10] DEBUG[2025] chan_sip.c: Line: a=sendrecv (10) [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] 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=z9hG4bK61c53869;rport From: "Valery Komarov" ;tag=as2daecde9 To: Contact: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Jul 2010 12:23:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=root 1994 1994 IN IP4 192.168.1.18 s=session c=IN IP4 192.168.1.18 t=0 0 m=audio 13694 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 16:23:10] DEBUG[2025] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] -- Called 360 [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- SIP read from 192.168.1.41:58978 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK780298b0;rport=5060 Contact: To: ;tag=5982bf54 From: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 103 NOTIFY User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 0 <-------------> [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK780298b0;rport=5060 (68) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 2: Contact: (78) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=5982bf54 (86) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 4: From: "Valery Komarov";tag=as3f4930a0 (59) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 5: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 6: CSeq: 103 NOTIFY (16) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 7: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 8: Content-Length: 0 (17) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 9: (0) [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] --- (9 headers 0 lines) --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = No match Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag Our tag: as2daecde9 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Stopping retransmission on '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' of Request 103: Match Found [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] SIP Response message for INCOMING dialog NOTIFY arrived [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport From: "Valery Komarov" ;tag=as2daecde9 To: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport (63) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as2daecde9 (60) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 3: To: (31) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 4: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 6: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 7: Content-Length: 0 (17) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 8: (0) [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] --- (8 headers 0 lines) --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = Found Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag Our tag: as2daecde9 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: *** SIP TIMER: Cancelling retransmission #42 - INVITE (got response) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18' Request 102: Found [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: SIP response 100 to standard invite [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- SIP read from 192.168.1.41:58978 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK05db8eab;rport=5060 Contact: To: ;tag=5982bf54 From: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 104 NOTIFY User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 0 <-------------> [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK05db8eab;rport=5060 (68) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 2: Contact: (78) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=5982bf54 (86) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 4: From: "Valery Komarov";tag=as3f4930a0 (59) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 5: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 6: CSeq: 104 NOTIFY (16) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 7: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 8: Content-Length: 0 (17) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 9: (0) [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] --- (9 headers 0 lines) --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = No match Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag Our tag: as2daecde9 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Stopping retransmission on '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' of Request 104: Match Found [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] SIP Response message for INCOMING dialog NOTIFY arrived [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- SIP read from 192.168.1.41:58978 ---> BYE sip:140@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-2207495947eb7941-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Valery Komarov";tag=as3f4930a0 From: ;tag=5982bf54 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 5 BYE User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 0 <-------------> [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 0: BYE sip:140@192.168.1.18 SIP/2.0 (32) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-2207495947eb7941-1---d8754z-;rport (92) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 3: Contact: (78) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 4: To: "Valery Komarov";tag=as3f4930a0 (57) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 5: From: ;tag=5982bf54 (88) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 6: Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 7: CSeq: 5 BYE (11) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 8: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 (48) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 10: (0) [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] --- (10 headers 0 lines) --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = No match Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag Our tag: as2daecde9 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = Found Their Call ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 Their Tag 5982bf54 Our tag: as3f4930a0 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] Sending to 192.168.1.41 : 58978 (no NAT) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Setting SIP_ALREADYGONE on dialog 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] Scheduling destruction of SIP dialog '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' in 6400 ms (Method: BYE) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Received bye, no owner, selfdestruct soon. [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- Transmitting (no NAT) to 192.168.1.41:58978 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:58978;branch=z9hG4bK-d8754z-2207495947eb7941-1---d8754z-;received=192.168.1.41;rport=58978 From: ;tag=5982bf54 To: "Valery Komarov";tag=as3f4930a0 Call-ID: 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 CSeq: 5 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport From: "Valery Komarov" ;tag=as2daecde9 To: ;tag=308594008 Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@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 16:23:10] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport (63) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as2daecde9 (60) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=308594008 (45) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 4: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 (54) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 6: Contact: (36) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 7: Server: Voip Phone 1.0 (22) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 8: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: Header 10: (0) [Jul 7 16:23:10] VERBOSE[2010] logger.c: [Jul 7 16:23:10] --- (10 headers 0 lines) --- [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: = Found Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag Our tag: as2daecde9 [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18' Request 102: Found [Jul 7 16:23:10] DEBUG[2010] chan_sip.c: SIP response 180 to standard invite [Jul 7 16:23:10] DEBUG[2010] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 16:23:10] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 16:23:10] DEBUG[1996] chan_sip.c: Checking device state for peer 360 [Jul 7 16:23:10] VERBOSE[2025] logger.c: [Jul 7 16:23:10] -- SIP/360-00000002 is ringing [Jul 7 16:23:10] DEBUG[1996] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 16:23:10] DEBUG[2025] rtp.c: Setting early bridge SDP of 'SIP/oxo-00000000' with that of 'SIP/360-00000002' [Jul 7 16:23:10] DEBUG[2025] channel.c: Driver for channel 'SIP/oxo-00000000' does not support indication 3, emulating it [Jul 7 16:23:10] DEBUG[2002] 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 16:23:10] DEBUG[2025] channel.c: Set channel SIP/oxo-00000000 to write format slin [Jul 7 16:23:10] DEBUG[2025] channel.c: Scheduling timer at 160 sample intervals [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport From: "Valery Komarov" ;tag=as2daecde9 To: ;tag=308594008 Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@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 27405232 16716477 IN IP4 192.168.5.13 s=A conversation c=IN IP4 192.168.5.13 t=0 0 m=audio 10002 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 16:23:12] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK61c53869;rport (63) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as2daecde9 (60) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=308594008 (45) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 4: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 (54) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 6: Contact: (36) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 7: Supported: 100rel, replaces, timer (34) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 8: Server: Voip Phone 1.0 (22) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 9: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 11: Content-Length: 237 (19) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Header 12: (0) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: v=0 (3) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: o=360 27405232 16716477 IN IP4 192.168.5.13 (43) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: s=A conversation (16) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: c=IN IP4 192.168.5.13 (21) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: t=0 0 (5) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: m=audio 10002 RTP/AVP 8 0 106 (29) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: a=rtpmap:106 telephone-event/8000 (33) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: a=fmtp:106 0-15 (15) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Line: a=sendrecv (10) [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] --- (12 headers 11 lines) --- [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: = Found Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag 308594008 Our tag: as2daecde9 [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Acked pending invite 102 [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Stopping retransmission on '4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18' of Request 102: Match Found [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: SIP response 200 to standard invite [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing session-level SDP o=360 27405232 16716477 IN IP4 192.168.5.13... UNSUPPORTED. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.13... OK. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Found RTP audio format 8 [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Found RTP audio format 0 [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Found RTP audio format 106 [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Found audio description format PCMA for ID 8 [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Found audio description format PCMU for ID 0 [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Found audio description format telephone-event for ID 106 [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 telephone-event/8000... OK. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:106 0-15... UNSUPPORTED. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: T38 state changed to 0 on channel SIP/360-00000002 [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] Peer audio RTP is at port 192.168.5.13:10002 [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: We have an owner, now see if we need to change this call [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Updating call counter for outgoing call [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: build_route: Contact hop: [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] list_route: hop: [Jul 7 16:23:12] DEBUG[2010] chan_sip.c: Strict routing enforced for session 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] set_destination: Parsing for address/port to send to [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] set_destination: set destination to 192.168.5.13, port 5060 [Jul 7 16:23:12] VERBOSE[2010] logger.c: [Jul 7 16:23:12] 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=z9hG4bK0a6e6f2a;rport From: "Valery Komarov" ;tag=as2daecde9 To: ;tag=308594008 Contact: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jul 7 16:23:12] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 16:23:12] VERBOSE[2025] logger.c: [Jul 7 16:23:12] -- SIP/360-00000002 answered SIP/oxo-00000000 [Jul 7 16:23:12] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 16:23:12] DEBUG[2025] channel.c: Set channel SIP/oxo-00000000 to write format alaw [Jul 7 16:23:12] DEBUG[2025] channel.c: Scheduling timer at 0 sample intervals [Jul 7 16:23:12] DEBUG[1996] chan_sip.c: Checking device state for peer 360 [Jul 7 16:23:12] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:12] DEBUG[1996] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 16:23:12] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:12] DEBUG[2002] 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 16:23:12] DEBUG[2025] rtp.c: Ooh, format changed from unknown to alaw [Jul 7 16:23:12] DEBUG[2025] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 7 16:23:13] DEBUG[2025] rtp.c: Got RTCP report of 60 bytes [Jul 7 16:23:13] DEBUG[2025] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:23:13] DEBUG[2025] rtp.c: Got RTCP report of 64 bytes [Jul 7 16:23:15] DEBUG[2025] rtp.c: Got RTCP report of 36 bytes [Jul 7 16:23:15] DEBUG[2025] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:23:17] DEBUG[2010] chan_sip.c: Auto destroying SIP dialog '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' [Jul 7 16:23:17] DEBUG[2010] chan_sip.c: Destroying SIP dialog 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:17] VERBOSE[2010] logger.c: [Jul 7 16:23:17] Really destroying SIP dialog '69dfde650dab4f780cb8efd308e9a33e@192.168.1.18' Method: BYE [Jul 7 16:23:17] DEBUG[2010] chan_sip.c: Updating call counter for outgoing call [Jul 7 16:23:17] DEBUG[2010] chan_sip.c: Call to peer '361' removed from call limit 0 [Jul 7 16:23:17] DEBUG[2010] devicestate.c: Notification of state change to be queued on device/channel SIP/361 [Jul 7 16:23:17] DEBUG[2010] chan_sip.c: This call did not properly clean up call limits. Call ID 69dfde650dab4f780cb8efd308e9a33e@192.168.1.18 [Jul 7 16:23:17] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 7 16:23:17] DEBUG[1996] chan_sip.c: Checking device state for peer 361 [Jul 7 16:23:17] DEBUG[1996] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 7 16:23:17] DEBUG[2002] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:18] DEBUG[2025] rtp.c: Got RTCP report of 36 bytes [Jul 7 16:23:18] DEBUG[2025] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:23:18] DEBUG[2025] rtp.c: Got RTCP report of 64 bytes [Jul 7 16:23:20] DEBUG[2025] rtp.c: Got RTCP report of 36 bytes [Jul 7 16:23:20] DEBUG[2025] rtp.c: Got RTCP report of 56 bytes [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] <--- SIP read from 192.168.0.202:5060 ---> BYE sip:361@192.168.1.18 SIP/2.0 Route: User-Agent: OxO_GW_700/013.001 To: sip:361@isoemo.com;tag=as646f20fc From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899161 BYE Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKcc38c917fa5535850b43d24c670ecdee Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 0: BYE sip:361@192.168.1.18 SIP/2.0 (32) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 1: Route: (28) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 2: User-Agent: OxO_GW_700/013.001 (30) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 3: To: sip:361@isoemo.com;tag=as646f20fc (37) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 4: From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd (64) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 5: Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 (55) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 6: CSeq: 2103899161 BYE (20) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 7: Via: SIP/2.0/UDP 192.168.0.202;rport;branch=z9hG4bKcc38c917fa5535850b43d24c670ecdee (83) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 10: (0) [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] --- (10 headers 0 lines) --- [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: = No match Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag 308594008 Our tag: as2daecde9 [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: = Found Their Call ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 Their Tag 54fbbc9dc7848b73526ce354fa6a8acd Our tag: as646f20fc [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] Sending to 192.168.0.202 : 5060 (NAT) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Setting SIP_ALREADYGONE on dialog 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Received bye, issuing owner hangup [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] <--- Transmitting (NAT) to 192.168.0.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.202;branch=z9hG4bKcc38c917fa5535850b43d24c670ecdee;received=192.168.0.202;rport=5060 From: sip:140@192.168.0.202;tag=54fbbc9dc7848b73526ce354fa6a8acd To: sip:361@isoemo.com;tag=as646f20fc Call-ID: 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202 CSeq: 2103899161 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Jul 7 16:23:23] DEBUG[2025] channel.c: Didn't get a frame from channel: SIP/oxo-00000000 [Jul 7 16:23:23] DEBUG[2025] rtp.c: Setting the marker bit due to a source update [Jul 7 16:23:23] DEBUG[2025] channel.c: Bridge stops bridging channels SIP/oxo-00000000 and SIP/360-00000002 [Jul 7 16:23:23] DEBUG[2025] channel.c: Hanging up channel 'SIP/360-00000002' [Jul 7 16:23:23] DEBUG[2025] chan_sip.c: Hangup call SIP/360-00000002, SIP callid 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18) [Jul 7 16:23:23] VERBOSE[2025] logger.c: [Jul 7 16:23:23] Scheduling destruction of SIP dialog '4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18' in 6400 ms (Method: INVITE) [Jul 7 16:23:23] DEBUG[2025] chan_sip.c: Strict routing enforced for session 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 [Jul 7 16:23:23] VERBOSE[2025] logger.c: [Jul 7 16:23:23] set_destination: Parsing for address/port to send to [Jul 7 16:23:23] VERBOSE[2025] logger.c: [Jul 7 16:23:23] set_destination: set destination to 192.168.5.13, port 5060 [Jul 7 16:23:23] VERBOSE[2025] logger.c: [Jul 7 16:23:23] Reliably Transmitting (no NAT) to 192.168.5.13:5060: BYE sip:360@192.168.5.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5fee0d72;rport From: "Valery Komarov" ;tag=as2daecde9 To: ;tag=308594008 Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 7 16:23:23] DEBUG[2025] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jul 7 16:23:23] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/360 [Jul 7 16:23:23] DEBUG[2025] rtp.c: Channel '' has no RTP, not doing anything [Jul 7 16:23:23] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 7 16:23:23] DEBUG[2025] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 7 16:23:23] DEBUG[1996] chan_sip.c: Checking device state for peer 360 [Jul 7 16:23:23] DEBUG[2025] pbx.c: Spawn extension (internal,360,1) exited non-zero on 'SIP/oxo-00000000' [Jul 7 16:23:23] DEBUG[1996] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 7 16:23:23] VERBOSE[2025] logger.c: [Jul 7 16:23:23] == Spawn extension (internal, 360, 1) exited non-zero on 'SIP/oxo-00000000' [Jul 7 16:23:23] DEBUG[2002] 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 16:23:23] DEBUG[2025] channel.c: Soft-Hanging up channel 'SIP/oxo-00000000' [Jul 7 16:23:23] DEBUG[2025] channel.c: Hanging up channel 'SIP/oxo-00000000' [Jul 7 16:23:23] DEBUG[2025] chan_sip.c: Hangup call SIP/oxo-00000000, SIP callid 01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202) [Jul 7 16:23:23] DEBUG[2025] devicestate.c: Notification of state change to be queued on device/channel SIP/oxo [Jul 7 16:23:23] DEBUG[1996] devicestate.c: No provider found, checking channel drivers for SIP - oxo [Jul 7 16:23:23] DEBUG[1996] chan_sip.c: Checking device state for peer oxo [Jul 7 16:23:23] DEBUG[1996] devicestate.c: Changing state for SIP/oxo - state 1 (Not in use) [Jul 7 16:23:23] DEBUG[2002] app_queue.c: Device 'SIP/oxo' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] <--- SIP read from 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5fee0d72;rport From: "Valery Komarov" ;tag=as2daecde9 To: ;tag=308594008 Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 CSeq: 103 BYE Server: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK5fee0d72;rport (63) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 2: From: "Valery Komarov" ;tag=as2daecde9 (60) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 3: To: ;tag=308594008 (45) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 4: Call-ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 (54) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 5: CSeq: 103 BYE (13) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 6: Server: Voip Phone 1.0 (22) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 7: Content-Length: 0 (17) [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Header 8: (0) [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] --- (8 headers 0 lines) --- [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: = Found Their Call ID: 4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18 Their Tag 308594008 Our tag: as2daecde9 [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #47 [Jul 7 16:23:23] DEBUG[2010] chan_sip.c: Stopping retransmission on '4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18' of Request 103: Match Found [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] Really destroying SIP dialog '4c582ad11c319dea3079f7520ea6fdb0@192.168.1.18' Method: INVITE [Jul 7 16:23:23] VERBOSE[2010] logger.c: [Jul 7 16:23:23] Really destroying SIP dialog '01856f075bdfdee0e2963e49ebfdfdd7@192.168.0.202' Method: BYE [Jul 7 16:23:28] DEBUG[2010] chan_sip.c: Auto destroying SIP dialog '310581660323243-12532222309725@192.168.5.13' [Jul 7 16:23:28] DEBUG[2010] chan_sip.c: Destroying SIP dialog 310581660323243-12532222309725@192.168.5.13 [Jul 7 16:23:28] VERBOSE[2010] logger.c: [Jul 7 16:23:28] Really destroying SIP dialog '310581660323243-12532222309725@192.168.5.13' Method: REGISTER [Jul 7 16:23:28] VERBOSE[2024] logger.c: [Jul 7 16:23:28] Beginning asterisk shutdown.... [Jul 7 16:23:28] VERBOSE[2024] logger.c: [Jul 7 16:23:28] Executing last minute cleanups [Jul 7 16:23:28] VERBOSE[2024] logger.c: [Jul 7 16:23:28] == Destroying musiconhold processes [Jul 7 16:23:28] DEBUG[2024] res_musiconhold.c: Destroying MOH class 'default' [Jul 7 16:23:28] VERBOSE[2024] logger.c: [Jul 7 16:23:28] Asterisk cleanly ending (0). [Jul 7 16:23:28] DEBUG[2024] asterisk.c: Asterisk ending (0).