(none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> INVITE sip:12347@192.168.222.160 SIP/2.0 Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0 Max-Forwards: 70 From: "x" ;tag=fded0efa50 To: "12347" Call-ID: 2a8af3779cf82b2c CSeq: 29923 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "x" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 53i/2.5.2.1010 Content-Type: application/sdp Content-Length: 263 v=0 o=MxSIP 0 0 IN IP4 172.16.102.104 s=SIP Call c=IN IP4 172.16.102.104 t=0 0 m=audio 3000 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:12347@192.168.222.160 SIP/2.0 (40) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0 (86) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Max-Forwards: 70 (16) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: To: "12347" (39) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 29923 INVITE (18) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (87) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Allow-Events: talk, hold, conference, LocalModeStatus (53) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Contact: "x" ;+sip.instance="" (122) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Supported: gruu, path, timer, 100rel, replaces (46) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: User-Agent: Aastra 53i/2.5.2.1010 (33) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 263 (19) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=MxSIP 0 0 IN IP4 172.16.102.104 (33) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=SIP Call (10) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 172.16.102.104 (23) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 3000 RTP/AVP 18 8 101 (29) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-15 (15) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) --- (14 headers 13 lines) --- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:2894 do_setnat: Setting NAT on RTP to Off [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:4786 sip_alloc: Allocating new SIP dialog for 2a8af3779cf82b2c - INVITE (With RTP) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:16443 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1766 parse_sip_options: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1774 parse_sip_options: Found SIP option: -gruu- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1780 parse_sip_options: Matched SIP option: gruu [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1774 parse_sip_options: Found SIP option: -path- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1780 parse_sip_options: Matched SIP option: path [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1774 parse_sip_options: Found SIP option: -timer- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1780 parse_sip_options: Matched SIP option: timer [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1774 parse_sip_options: Found SIP option: -100rel- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1780 parse_sip_options: Matched SIP option: 100rel [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1774 parse_sip_options: Found SIP option: -replaces- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:1780 parse_sip_options: Matched SIP option: replaces Sending to 172.16.102.104 : 5060 (no NAT) Using INVITE request as basis request - 2a8af3779cf82b2c Found peer '14237' [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:2894 do_setnat: Setting NAT on RTP to Off [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP o=MxSIP 0 0 IN IP4 172.16.102.104... UNSUPPORTED. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP c=IN IP4 172.16.102.104... OK. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. Found audio description format PCMA for ID 8 [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. Found audio description format telephone-event for ID 101 [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5582 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.102.104:3000 [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5710 process_sdp: We're settling with these formats: 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:15083 handle_request_invite: Checking SIP call limits for device 14237 [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:3389 update_call_counter: Updating call counter for incoming call Looking for 12347 in testtransfer (domain 192.168.222.160) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:4238 sip_new: *** Our native formats are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:4239 sip_new: *** Joint capabilities are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:4240 sip_new: *** Our capabilities are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:4241 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:4264 sip_new: This channel will not be able to handle video. [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:8906 build_route: build_route: Contact hop: "x" ;+sip.instance="" list_route: hop: [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:15182 handle_request_invite: SIP/14237-00000007: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.102.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0;received=172.16.102.104 From: "x" ;tag=fded0efa50 To: "12347" Call-ID: 2a8af3779cf82b2c CSeq: 29923 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 100 Trying (18) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0;received=172.16.102.104 (110) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "12347" (39) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 29923 INVITE (18) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Supported: replaces (19) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Contact: (36) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 0 (17) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: (0) [Nov 18 10:29:07] DEBUG[19213]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/14237 [Nov 18 10:29:07] DEBUG[19261]: pbx.c:1861 pbx_extension_helper: Launching 'Set' -- Executing [12347@testtransfer:1] Set("SIP/14237-00000007", "TRANSFER_CONTEXT=transfers") in new stack [Nov 18 10:29:07] DEBUG[19188]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 14237 [Nov 18 10:29:07] DEBUG[19188]: chan_sip.c:17217 sip_devicestate: Checking device state for peer 14237 [Nov 18 10:29:07] DEBUG[19188]: devicestate.c:287 do_state_change: Changing state for SIP/14237 - state 1 (Not in use) [Nov 18 10:29:07] DEBUG[19209]: app_queue.c:676 handle_statechange: Device 'SIP/14237' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 18 10:29:07] DEBUG[19261]: pbx.c:1861 pbx_extension_helper: Launching 'DumpChan' -- Executing [12347@testtransfer:2] DumpChan("SIP/14237-00000007", "") in new stack Dumping Info For Channel: SIP/14237-00000007: ================================================================================ Info: Name= SIP/14237-00000007 Type= SIP UniqueID= 1258536547.7 CallerID= 14237 CallerIDName= x DNIDDigits= 12347 RDNIS= (N/A) State= Ring (4) Rings= 0 NativeFormat= 0x8 (alaw) WriteFormat= 0x8 (alaw) ReadFormat= 0x8 (alaw) 1stFileDescriptor= 23 Framesin= 0 Framesout= 0 TimetoHangup= 0 ElapsedTime= 0h0m0s Context= testtransfer Extension= 12347 Priority= 2 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: TRANSFER_CONTEXT=transfers SIPCALLID=2a8af3779cf82b2c SIPUSERAGENT=Aastra 53i/2.5.2.1010 SIPDOMAIN=192.168.222.160 SIPURI=sip:14237@172.16.102.104:5060 ================================================================================ [Nov 18 10:29:07] DEBUG[19261]: pbx.c:1861 pbx_extension_helper: Launching 'Dial' -- Executing [12347@testtransfer:3] Dial("SIP/14237-00000007", "SIP/12347||M(dumpchan)") in new stack [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:17293 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4786 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:2894 do_setnat: Setting NAT on RTP to Off [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4238 sip_new: *** Our native formats are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4239 sip_new: *** Joint capabilities are 0x0 (nothing) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4240 sip_new: *** Our capabilities are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4241 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4243 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:4264 sip_new: This channel will not be able to handle video. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable TRANSFER_CONTEXT. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Nov 18 10:29:07] DEBUG[19261]: channel.c:3743 ast_channel_inherit_variables: Not copying variable SIPURI. [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:3182 sip_call: Outgoing Call for 12347 [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:3389 update_call_counter: Updating call counter for outgoing call [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:3197 sip_call: Our T38 capability (0), joint T38 capability (0) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:6943 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:6944 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.222.160 port 19664 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:7061 add_sdp: -- Done with adding codecs to SDP [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:7170 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 (72) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport (66) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=as3fc6596f (52) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: (63) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 INVITE (16) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Date: Wed, 18 Nov 2009 09:29:07 GMT (35) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: Supported: replaces (19) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 246 (19) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19181 IN IP4 192.168.222.160 (41) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.160 (24) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 19664 RTP/AVP 8 101 (27) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.222.79:42400: INVITE sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport From: "x" ;tag=as3fc6596f To: Contact: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Nov 2009 09:29:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 246 v=0 o=root 19181 19181 IN IP4 192.168.222.160 s=session c=IN IP4 192.168.222.160 t=0 0 m=audio 19664 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 (72) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport (66) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=as3fc6596f (52) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: (63) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 INVITE (16) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Date: Wed, 18 Nov 2009 09:29:07 GMT (35) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: Supported: replaces (19) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 246 (19) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19181 IN IP4 192.168.222.160 (41) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.160 (24) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 19664 RTP/AVP 8 101 (27) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 -- Called 12347 (none)*CLI> <--- SIP read from 192.168.222.79:42400 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport=5060 Contact: To: ;tag=ea5fed10 From: "x";tag=as3fc6596f Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 102 INVITE User-Agent: eyeBeam release 1003l stamp 30936 Content-Length: 0 <-------------> [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport=5060 (71) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Contact: (68) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: From: "x";tag=as3fc6596f (51) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 INVITE (16) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: eyeBeam release 1003l stamp 30936 (45) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Content-Length: 0 (17) --- (9 headers 0 lines) --- [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:2305 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #64 - INVITE (got response) [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:2313 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' Request 102: Found [Nov 18 10:29:07] DEBUG[19213]: chan_sip.c:12840 handle_response_invite: SIP response 180 to standard invite [Nov 18 10:29:07] DEBUG[19213]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/12347 [Nov 18 10:29:07] DEBUG[19188]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 12347 [Nov 18 10:29:07] DEBUG[19188]: chan_sip.c:17217 sip_devicestate: Checking device state for peer 12347 [Nov 18 10:29:07] DEBUG[19188]: devicestate.c:287 do_state_change: Changing state for SIP/12347 - state 1 (Not in use) [Nov 18 10:29:07] DEBUG[19209]: app_queue.c:676 handle_statechange: Device 'SIP/12347' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. -- SIP/12347-00000008 is ringing <--- Transmitting (no NAT) to 172.16.102.104:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0;received=172.16.102.104 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29923 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0;received=172.16.102.104 (110) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: CSeq: 29923 INVITE (18) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Supported: replaces (19) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Contact: (36) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 0 (17) [Nov 18 10:29:07] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: (0) [Nov 18 10:29:08] DEBUG[19261]: rtp.c:924 ast_rtcp_read: Got RTCP report of 132 bytes (none)*CLI> <--- SIP read from 192.168.222.79:42400 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport=5060 Contact: To: ;tag=ea5fed10 From: "x";tag=as3fc6596f Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003l stamp 30936 Content-Length: 241 v=0 o=- 2 2 IN IP4 192.168.222.79 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.222.79 t=0 0 m=audio 39872 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:7CBC4178DFB14B8FBE80147F9C175828 <-------------> [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5a3624a5;rport=5060 (71) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Contact: (68) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: From: "x";tag=as3fc6596f (51) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 INVITE (16) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Content-Type: application/sdp (29) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: User-Agent: eyeBeam release 1003l stamp 30936 (45) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 241 (19) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: (0) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=- 2 2 IN IP4 192.168.222.79 (29) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=CounterPath eyeBeam 1.5 (25) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.79 (23) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 39872 RTP/AVP 8 101 (27) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-15 (15) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) --- (11 headers 10 lines) --- [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:2239 __sip_ack: Acked pending invite 102 [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' of Request 102: Match Found [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:12840 handle_response_invite: SIP response 200 to standard invite [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP o=- 2 2 IN IP4 192.168.222.79... UNSUPPORTED. [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED. [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP c=IN IP4 192.168.222.79... OK. [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. Found RTP audio format 8 Found RTP audio format 101 [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. Found audio description format telephone-event for ID 101 [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5582 process_sdp: T38 state changed to 0 on channel SIP/12347-00000008 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.222.79:39872 [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5710 process_sdp: We're settling with these formats: 0x8 (alaw) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5717 process_sdp: We have an owner, now see if we need to change this call [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:3389 update_call_counter: Updating call counter for outgoing call [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:8906 build_route: build_route: Contact hop: list_route: hop: [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:6452 reqprep: Strict routing enforced for session 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.222.79, port 42400 Transmitting (no NAT) to 192.168.222.79:42400: ACK sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK1c7b9225;rport From: "x" ;tag=as3fc6596f To: ;tag=ea5fed10 Contact: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: ACK sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 (69) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK1c7b9225;rport (66) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=as3fc6596f (52) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 ACK (13) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Content-Length: 0 (17) [Nov 18 10:29:08] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: (0) [Nov 18 10:29:08] DEBUG[19261]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/12347 -- SIP/12347-00000008 answered SIP/14237-00000007 [Nov 18 10:29:08] DEBUG[19261]: pbx.c:1861 pbx_extension_helper: Launching 'Set' -- Executing [s@macro-dumpchan:1] Set("SIP/12347-00000008", "TRANSFER_CONTEXT=transfers") in new stack [Nov 18 10:29:08] DEBUG[19261]: app_macro.c:379 _macro_exec: Executed application: Set [Nov 18 10:29:08] DEBUG[19261]: pbx.c:1861 pbx_extension_helper: Launching 'DumpChan' -- Executing [s@macro-dumpchan:2] DumpChan("SIP/12347-00000008", "") in new stack Dumping Info For Channel: SIP/12347-00000008: ================================================================================ Info: Name= SIP/12347-00000008 Type= SIP UniqueID= 1258536547.8 CallerID= 12347 CallerIDName= (N/A) DNIDDigits= (N/A) RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x8 (alaw) WriteFormat= 0x8 (alaw) ReadFormat= 0x8 (alaw) 1stFileDescriptor= 27 Framesin= 3 Framesout= 0 TimetoHangup= 0 ElapsedTime= 0h0m1s Context= macro-dumpchan Extension= s Priority= 2 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: MACRO_DEPTH=1 TRANSFER_CONTEXT=transfers MACRO_PRIORITY=1 MACRO_CONTEXT=testtransfer MACRO_EXTEN= DIALEDPEERNUMBER=12347 SIPCALLID=7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 ================================================================================ [Nov 18 10:29:08] DEBUG[19261]: app_macro.c:379 _macro_exec: Executed application: Dumpchan [Nov 18 10:29:08] DEBUG[19261]: app_dial.c:1737 dial_exec_full: Macro exited with status 0 [Nov 18 10:29:08] DEBUG[19188]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 12347 [Nov 18 10:29:08] DEBUG[19188]: chan_sip.c:17217 sip_devicestate: Checking device state for peer 12347 [Nov 18 10:29:08] DEBUG[19188]: devicestate.c:287 do_state_change: Changing state for SIP/12347 - state 1 (Not in use) [Nov 18 10:29:08] DEBUG[19209]: app_queue.c:676 handle_statechange: Device 'SIP/12347' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 18 10:29:08] DEBUG[19261]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/14237 [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:3867 sip_answer: SIP answering channel: SIP/14237-00000007 [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:7228 transmit_response_with_sdp: Setting framing from config on incoming call [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:6943 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:6944 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.222.160 port 10922 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:7061 add_sdp: -- Done with adding codecs to SDP [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:7170 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) <--- Reliably Transmitting (no NAT) to 172.16.102.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0;received=172.16.102.104 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29923 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 19181 19181 IN IP4 192.168.222.160 s=session c=IN IP4 192.168.222.160 t=0 0 m=audio 10922 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9a1413db5955effd8.26c2ff0638331d6a0;received=172.16.102.104 (110) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: CSeq: 29923 INVITE (18) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Supported: replaces (19) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Contact: (36) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Content-Type: application/sdp (29) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: Content-Length: 246 (19) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: (0) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19181 IN IP4 192.168.222.160 (41) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.160 (24) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 10922 RTP/AVP 8 101 (27) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 -- Native bridging SIP/14237-00000007 and SIP/12347-00000008 [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:19068 sip_set_rtp_peer: Deferring reinvite on SIP '2a8af3779cf82b2c' - It's audio will be redirected to IP 192.168.222.79 [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:19063 sip_set_rtp_peer: Sending reinvite on SIP '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' - It's audio soon redirected to IP 172.16.102.104 [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:6452 reqprep: Strict routing enforced for session 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.222.79, port 42400 [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:6943 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:6944 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 192.168.222.160 port 19664 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:7061 add_sdp: -- Done with adding codecs to SDP [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:7170 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:1707 initialize_initreq: Initializing already initialized SIP dialog 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (presumably reinvite) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 (72) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK588b85f6;rport (66) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=as3fc6596f (52) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 INVITE (16) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Supported: replaces (19) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 243 (19) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19182 IN IP4 172.16.102.104 (40) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 172.16.102.104 (23) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 3000 RTP/AVP 8 101 (26) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.222.79:42400: INVITE sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK588b85f6;rport From: "x" ;tag=as3fc6596f To: ;tag=ea5fed10 Contact: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 243 v=0 o=root 19181 19182 IN IP4 172.16.102.104 s=session c=IN IP4 172.16.102.104 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 (72) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK588b85f6;rport (66) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=as3fc6596f (52) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 INVITE (16) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Supported: replaces (19) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 243 (19) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19182 IN IP4 172.16.102.104 (40) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 172.16.102.104 (23) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 3000 RTP/AVP 8 101 (26) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) [Nov 18 10:29:08] DEBUG[19261]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 18 10:29:08] DEBUG[19188]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 14237 [Nov 18 10:29:08] DEBUG[19188]: chan_sip.c:17217 sip_devicestate: Checking device state for peer 14237 [Nov 18 10:29:08] DEBUG[19188]: devicestate.c:287 do_state_change: Changing state for SIP/14237 - state 1 (Not in use) [Nov 18 10:29:08] DEBUG[19209]: app_queue.c:676 handle_statechange: Device 'SIP/14237' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. (none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> ACK sip:12347@192.168.222.160 SIP/2.0 Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK2d9859a16d63dac7c.e0a443b189e5aed4b Max-Forwards: 70 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29923 ACK User-Agent: Aastra 53i/2.5.2.1010 Content-Length: 0 <-------------> [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: ACK sip:12347@192.168.222.160 SIP/2.0 (37) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK2d9859a16d63dac7c.e0a443b189e5aed4b (86) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Max-Forwards: 70 (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 29923 ACK (15) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Aastra 53i/2.5.2.1010 (33) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Content-Length: 0 (17) --- (9 headers 0 lines) --- [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:16443 handle_request: **** Received ACK (6) - Command in SIP ACK [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2247 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #67 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '2a8af3779cf82b2c' of Response 29923: Match Found [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12802 check_pendings: Sending pending reinvite on '2a8af3779cf82b2c' [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:6452 reqprep: Strict routing enforced for session 2a8af3779cf82b2c set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.102.104, port 5060 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:6943 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:6944 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.222.160 port 10922 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:7061 add_sdp: -- Done with adding codecs to SDP [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:7170 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:1707 initialize_initreq: Initializing already initialized SIP dialog 2a8af3779cf82b2c (presumably reinvite) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (58) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK57bf6d9e;rport (66) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 INVITE (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Supported: replaces (19) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 244 (19) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=root 19181 19182 IN IP4 192.168.222.79 (40) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.79 (23) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 39872 RTP/AVP 8 101 (27) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 172.16.102.104:5060: INVITE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK57bf6d9e;rport From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Contact: Call-ID: 2a8af3779cf82b2c CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 244 v=0 o=root 19181 19182 IN IP4 192.168.222.79 s=session c=IN IP4 192.168.222.79 t=0 0 m=audio 39872 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (58) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK57bf6d9e;rport (66) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 INVITE (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Supported: replaces (19) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 244 (19) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=root 19181 19182 IN IP4 192.168.222.79 (40) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.79 (23) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 39872 RTP/AVP 8 101 (27) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 18 10:29:09] DEBUG[19261]: rtp.c:2898 ast_rtp_write: Ooh, format changed from unknown to alaw [Nov 18 10:29:09] DEBUG[19261]: rtp.c:2915 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 (none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK57bf6d9e;rport=5060;received=192.168.222.160 From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Call-ID: 2a8af3779cf82b2c CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "x" ;+sip.instance="" Server: Aastra 53i/2.5.2.1010 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 237 v=0 o=MxSIP 0 1 IN IP4 172.16.102.104 s=SIP Call c=IN IP4 172.16.102.104 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK57bf6d9e;rport=5060;received=192.168.222.160 (96) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 102 INVITE (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (87) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow-Events: talk, hold, conference, LocalModeStatus (53) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Contact: "x" ;+sip.instance="" (122) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Server: Aastra 53i/2.5.2.1010 (29) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Supported: gruu, path, timer, replaces (38) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: Content-Type: application/sdp (29) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: Content-Length: 237 (19) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 13: (0) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=MxSIP 0 1 IN IP4 172.16.102.104 (33) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=SIP Call (10) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 172.16.102.104 (23) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 3000 RTP/AVP 8 101 (26) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-15 (15) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) --- (13 headers 12 lines) --- [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2239 __sip_ack: Acked pending invite 102 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2247 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #69 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '2a8af3779cf82b2c' of Request 102: Match Found [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12838 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2a8af3779cf82b2c [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP o=MxSIP 0 1 IN IP4 172.16.102.104... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP c=IN IP4 172.16.102.104... OK. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. Found audio description format telephone-event for ID 101 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5582 process_sdp: T38 state changed to 0 on channel SIP/14237-00000007 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.102.104:3000 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5710 process_sdp: We're settling with these formats: 0x8 (alaw) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5717 process_sdp: We have an owner, now see if we need to change this call [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:3389 update_call_counter: Updating call counter for incoming call [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12970 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12975 handle_response_invite: T38 state changed to 0 on channel SIP [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12978 handle_response_invite: T38 state changed to 0 on channel SIP/14237-00000007 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:6452 reqprep: Strict routing enforced for session 2a8af3779cf82b2c set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.102.104, port 5060 Transmitting (no NAT) to 172.16.102.104:5060: ACK sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK377c3034;rport From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Contact: Call-ID: 2a8af3779cf82b2c CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: ACK sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (55) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK377c3034;rport (66) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 102 ACK (13) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Content-Length: 0 (17) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: (0) (none)*CLI> <--- SIP read from 192.168.222.79:42400 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK588b85f6;rport=5060 Contact: To: ;tag=ea5fed10 From: "x";tag=as3fc6596f Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003l stamp 30936 Content-Length: 241 v=0 o=- 2 2 IN IP4 192.168.222.79 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.222.79 t=0 0 m=audio 39872 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:7CBC4178DFB14B8FBE80147F9C175828 <-------------> [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK588b85f6;rport=5060 (71) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Contact: (68) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: From: "x";tag=as3fc6596f (51) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 INVITE (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Content-Type: application/sdp (29) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: User-Agent: eyeBeam release 1003l stamp 30936 (45) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 241 (19) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: (0) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=- 2 2 IN IP4 192.168.222.79 (29) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=CounterPath eyeBeam 1.5 (25) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.79 (23) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 39872 RTP/AVP 8 101 (27) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-15 (15) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) --- (11 headers 10 lines) --- [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2239 __sip_ack: Acked pending invite 103 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2247 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #68 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' of Request 103: Match Found [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12838 handle_response_invite: SIP response 200 to RE-invite on outgoing call 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP o=- 2 2 IN IP4 192.168.222.79... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP c=IN IP4 192.168.222.79... OK. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. Found RTP audio format 8 Found RTP audio format 101 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. Found audio description format telephone-event for ID 101 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5582 process_sdp: T38 state changed to 0 on channel SIP/12347-00000008 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.222.79:39872 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5710 process_sdp: We're settling with these formats: 0x8 (alaw) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5717 process_sdp: We have an owner, now see if we need to change this call [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:3389 update_call_counter: Updating call counter for outgoing call [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12970 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12975 handle_response_invite: T38 state changed to 0 on channel SIP [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:12978 handle_response_invite: T38 state changed to 0 on channel SIP/12347-00000008 [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:6452 reqprep: Strict routing enforced for session 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.222.79, port 42400 Transmitting (no NAT) to 192.168.222.79:42400: ACK sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK616f3767;rport From: "x" ;tag=as3fc6596f To: ;tag=ea5fed10 Contact: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: ACK sip:12347@192.168.222.79:42400;rinstance=a308bf3782f9d39b SIP/2.0 (69) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK616f3767;rport (66) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=as3fc6596f (52) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: ;tag=ea5fed10 (76) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 ACK (13) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Content-Length: 0 (17) [Nov 18 10:29:09] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: (0) (none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> INVITE sip:12347@192.168.222.160 SIP/2.0 Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK8f55f71c768774a26.f8daac06646b77770 Max-Forwards: 70 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29924 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "x" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 53i/2.5.2.1010 Content-Type: application/sdp Content-Length: 263 v=0 o=MxSIP 0 2 IN IP4 172.16.102.104 s=SIP Call c=IN IP4 172.16.102.104 t=0 0 m=audio 3000 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendonly <-------------> [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:12347@192.168.222.160 SIP/2.0 (40) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK8f55f71c768774a26.f8daac06646b77770 (86) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Max-Forwards: 70 (16) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 29924 INVITE (18) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (87) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Allow-Events: talk, hold, conference, LocalModeStatus (53) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Contact: "x" ;+sip.instance="" (122) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Supported: gruu, path, timer, 100rel, replaces (46) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: User-Agent: Aastra 53i/2.5.2.1010 (33) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 263 (19) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=MxSIP 0 2 IN IP4 172.16.102.104 (33) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=SIP Call (10) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 172.16.102.104 (23) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 3000 RTP/AVP 18 8 101 (29) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-15 (15) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) --- (14 headers 13 lines) --- [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:16443 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 172.16.102.104 : 5060 (no NAT) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP o=MxSIP 0 2 IN IP4 172.16.102.104... UNSUPPORTED. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP c=IN IP4 172.16.102.104... OK. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. Found audio description format PCMA for ID 8 [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. Found audio description format telephone-event for ID 101 [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5582 process_sdp: T38 state changed to 0 on channel SIP/14237-00000007 Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.102.104:3000 [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5710 process_sdp: We're settling with these formats: 0x8 (alaw) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5717 process_sdp: We have an owner, now see if we need to change this call [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:15142 handle_request_invite: Got a SIP re-invite for call 2a8af3779cf82b2c [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:15256 handle_request_invite: SIP/14237-00000007: This call is UP.... (none)*CLI> <--- Transmitting (no NAT) to 172.16.102.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK8f55f71c768774a26.f8daac06646b77770;received=172.16.102.104 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29924 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 100 Trying (18) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK8f55f71c768774a26.f8daac06646b77770;received=172.16.102.104 (110) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 29924 INVITE (18) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Supported: replaces (19) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Contact: (36) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 0 (17) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: (0) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:7228 transmit_response_with_sdp: Setting framing from config on incoming call [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:6943 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:6944 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.222.160 port 10922 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:7061 add_sdp: -- Done with adding codecs to SDP [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:7170 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) (none)*CLI> <--- Reliably Transmitting (no NAT) to 172.16.102.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK8f55f71c768774a26.f8daac06646b77770;received=172.16.102.104 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29924 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: ontent-Type: application/sdp Content-Length: 244 v=0 o=root 19181 19183 IN IP4 192.168.222.79 s=session c=IN IP4 192.168.222.79 t=0 0 m=audio 39872 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK8f55f71c768774a26.f8daac06646b77770;received=172.16.102.104 (110) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 29924 INVITE (18) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Supported: replaces (19) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Contact: (36) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Type: application/sdp (29) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: Content-Length: 244 (19) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: (0) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=root 19181 19183 IN IP4 192.168.222.79 (40) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.79 (23) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 39872 RTP/AVP 8 101 (27) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) [Nov 18 10:29:12] DEBUG[19213]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 (none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> ACK sip:12347@192.168.222.160 SIP/2.0 Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9028dcd50d88c3b97.b99573d6a39a87d79 Max-Forwards: 70 From: "x" ;tag=fded0efa50 To: "12347" ;tag=as24c56fcc Call-ID: 2a8af3779cf82b2c CSeq: 29924 ACK User-Agent: Aastra 53i/2.5.2.1010 Content-Length: 0 <-------------> [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: ACK sip:12347@192.168.222.160 SIP/2.0 (37) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.102.104:5060;branch=z9hG4bK9028dcd50d88c3b97.b99573d6a39a87d79 (86) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Max-Forwards: 70 (16) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: From: "x" ;tag=fded0efa50 (52) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: To: "12347" ;tag=as24c56fcc (54) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 29924 ACK (15) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Aastra 53i/2.5.2.1010 (33) [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Content-Length: 0 (17) --- (9 headers 0 lines) --- [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:16443 handle_request: **** Received ACK (6) - Command in SIP ACK [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:2247 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #72 [Nov 18 10:29:13] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '2a8af3779cf82b2c' of Response 29924: Match Found (none)*CLI> <--- SIP read from 192.168.222.79:42400 ---> BYE sip:14237@192.168.222.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.222.79:42400;branch=z9hG4bK-d87543-ca48a819ac600f0b-1--d87543-;rport Max-Forwards: 70 Contact: To: "x";tag=as3fc6596f From: ;tag=ea5fed10 Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 2 BYE User-Agent: eyeBeam release 1003l stamp 30936 eason: SIP;description="User Hung Up" Content-Length: 0 <-------------> [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: BYE sip:14237@192.168.222.160 SIP/2.0 (37) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.79:42400;branch=z9hG4bK-d87543-ca48a819ac600f0b-1--d87543-;rport (93) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: Max-Forwards: 70 (16) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: Contact: (68) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: To: "x";tag=as3fc6596f (49) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: From: ;tag=ea5fed10 (78) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: CSeq: 2 BYE (11) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: User-Agent: eyeBeam release 1003l stamp 30936 (45) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Reason: SIP;description="User Hung Up" (38) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 0 (17) --- (11 headers 0 lines) --- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:16443 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.222.79 : 42400 (no NAT) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:1719 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:15978 handle_request_bye: Received bye, issuing owner hangup (none)*CLI> <--- Transmitting (no NAT) to 192.168.222.79:42400 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.222.79:42400;branch=z9hG4bK-d87543-ca48a819ac600f0b-1--d87543-;received=192.168.222.79;rport=42400 From: ;tag=ea5fed10 To: "x";tag=as3fc6596f Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Nov 18 10:29:20] DEBUG[19261]: rtp.c:3038 bridge_native_loop: Oooh, 'SIP/12347-00000008' changed end address to 0.0.0.0:0 (format 8) [Nov 18 10:29:20] DEBUG[19261]: rtp.c:3040 bridge_native_loop: Oooh, 'SIP/12347-00000008' changed end vaddress to 0.0.0.0:0 (format 8) [Nov 18 10:29:20] DEBUG[19261]: rtp.c:3042 bridge_native_loop: Oooh, 'SIP/12347-00000008' was 192.168.222.79:39872/(format 8) [Nov 18 10:29:20] DEBUG[19261]: rtp.c:3044 bridge_native_loop: Oooh, 'SIP/12347-00000008' was 0.0.0.0:0/(format 8) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:19063 sip_set_rtp_peer: Sending reinvite on SIP '2a8af3779cf82b2c' - It's audio soon redirected to IP 192.168.222.160 [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:6452 reqprep: Strict routing enforced for session 2a8af3779cf82b2c set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.102.104, port 5060 [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:6943 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:6944 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.222.160 port 10922 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:7061 add_sdp: -- Done with adding codecs to SDP [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:7170 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:1707 initialize_initreq: Initializing already initialized SIP dialog 2a8af3779cf82b2c (presumably reinvite) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (58) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK0acadf95;rport (66) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 INVITE (16) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Supported: replaces (19) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 246 (19) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19184 IN IP4 192.168.222.160 (41) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.160 (24) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 10922 RTP/AVP 8 101 (27) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 172.16.102.104:5060: INVITE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK0acadf95;rport From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Contact: Call-ID: 2a8af3779cf82b2c CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 246 v=0 o=root 19181 19184 IN IP4 192.168.222.160 s=session c=IN IP4 192.168.222.160 t=0 0 m=audio 10922 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 0: INVITE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (58) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK0acadf95;rport (66) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 INVITE (16) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 10: Supported: replaces (19) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 12: Content-Type: application/sdp (29) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 13: Content-Length: 246 (19) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5053 parse_request: Header 14: (0) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: o=root 19181 19184 IN IP4 192.168.222.160 (41) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: s=session (9) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: c=IN IP4 192.168.222.160 (24) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: m=audio 10922 RTP/AVP 8 101 (27) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-16 (15) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:5089 parse_request: Line: a=sendrecv (10) [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.79:42400;branch=z9hG4bK-d87543-ca48a819ac600f0b-1--d87543-;received=192.168.222.79;rport=42400 (123) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: ;tag=ea5fed10 (78) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x";tag=as3fc6596f (49) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160 (57) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 2 BYE (11) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Supported: replaces (19) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Content-Length: 0 (17) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: (0) [Nov 18 10:29:20] DEBUG[19261]: rtp.c:3093 bridge_native_loop: Oooh, got a hangup [Nov 18 10:29:20] DEBUG[19261]: channel.c:4562 ast_channel_bridge: Returning from native bridge, channels: SIP/14237-00000007, SIP/12347-00000008 [Nov 18 10:29:20] DEBUG[19261]: channel.c:1564 ast_hangup: Hanging up channel 'SIP/12347-00000008' [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:3706 sip_hangup: Hangup call SIP/12347-00000008, SIP callid 7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160) [Nov 18 10:29:20] DEBUG[19261]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/12347 [Nov 18 10:29:20] DEBUG[19261]: rtp.c:1573 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Nov 18 10:29:20] DEBUG[19261]: app_dial.c:1901 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Nov 18 10:29:20] DEBUG[19261]: pbx.c:2411 __ast_pbx_run: Spawn extension (testtransfer,12347,3) exited non-zero on 'SIP/14237-00000007' == Spawn extension (testtransfer, 12347, 3) exited non-zero on 'SIP/14237-00000007' [Nov 18 10:29:20] DEBUG[19261]: channel.c:1461 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/14237-00000007' [Nov 18 10:29:20] DEBUG[19261]: channel.c:1564 ast_hangup: Hanging up channel 'SIP/14237-00000007' [Nov 18 10:29:20] DEBUG[19261]: chan_sip.c:3706 sip_hangup: Hangup call SIP/14237-00000007, SIP callid 2a8af3779cf82b2c) Scheduling destruction of SIP dialog '2a8af3779cf82b2c' in 32000 ms (Method: ACK) [Nov 18 10:29:20] DEBUG[19261]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/14237 [Nov 18 10:29:20] DEBUG[19188]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 12347 [Nov 18 10:29:20] DEBUG[19188]: chan_sip.c:17217 sip_devicestate: Checking device state for peer 12347 [Nov 18 10:29:20] DEBUG[19188]: devicestate.c:287 do_state_change: Changing state for SIP/12347 - state 1 (Not in use) [Nov 18 10:29:20] DEBUG[19188]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 14237 [Nov 18 10:29:20] DEBUG[19188]: chan_sip.c:17217 sip_devicestate: Checking device state for peer 14237 [Nov 18 10:29:20] DEBUG[19188]: devicestate.c:287 do_state_change: Changing state for SIP/14237 - state 1 (Not in use) [Nov 18 10:29:20] DEBUG[19209]: app_queue.c:676 handle_statechange: Device 'SIP/12347' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 18 10:29:20] DEBUG[19209]: app_queue.c:676 handle_statechange: Device 'SIP/14237' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. (none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK0acadf95;rport=5060;received=192.168.222.160 From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Call-ID: 2a8af3779cf82b2c CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "x" ;+sip.instance="" Server: Aastra 53i/2.5.2.1010 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 237 v=0 o=MxSIP 0 3 IN IP4 172.16.102.104 s=SIP Call c=IN IP4 172.16.102.104 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendonly <-------------> [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK0acadf95;rport=5060;received=192.168.222.160 (96) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 103 INVITE (16) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (87) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Allow-Events: talk, hold, conference, LocalModeStatus (53) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Contact: "x" ;+sip.instance="" (122) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Server: Aastra 53i/2.5.2.1010 (29) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Supported: gruu, path, timer, replaces (38) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: Content-Type: application/sdp (29) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 12: Content-Length: 237 (19) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 13: (0) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: v=0 (3) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: o=MxSIP 0 3 IN IP4 172.16.102.104 (33) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: s=SIP Call (10) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: c=IN IP4 172.16.102.104 (23) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: t=0 0 (5) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: m=audio 3000 RTP/AVP 8 101 (26) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=silenceSupp:off - - - - (25) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=fmtp:101 0-15 (15) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5089 parse_request: Line: a=ptime:20 (10) --- (13 headers 12 lines) --- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:2239 __sip_ack: Acked pending invite 103 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:2247 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #73 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '2a8af3779cf82b2c' of Request 103: Match Found [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:12840 handle_response_invite: SIP response 200 to standard invite [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP o=MxSIP 0 3 IN IP4 172.16.102.104... UNSUPPORTED. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP c=IN IP4 172.16.102.104... OK. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5419 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. Found audio description format telephone-event for ID 101 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5557 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5582 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.102.104:3000 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5710 process_sdp: We're settling with these formats: 0x8 (alaw) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:3389 update_call_counter: Updating call counter for incoming call [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:8844 build_route: build_route: Retaining previous route: [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:6452 reqprep: Strict routing enforced for session 2a8af3779cf82b2c set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.102.104, port 5060 Transmitting (no NAT) to 172.16.102.104:5060: ACK sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK56e0b7fa;rport From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Contact: Call-ID: 2a8af3779cf82b2c CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: ACK sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (55) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK56e0b7fa;rport (66) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Contact: (36) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: CSeq: 103 ACK (13) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: Max-Forwards: 70 (16) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: Content-Length: 0 (17) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: (0) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:6452 reqprep: Strict routing enforced for session 2a8af3779cf82b2c set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.102.104, port 5060 Reliably Transmitting (no NAT) to 172.16.102.104:5060: BYE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5082a199;rport From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Call-ID: 2a8af3779cf82b2c CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: BYE sip:14237@172.16.102.104:5060;transport=udp SIP/2.0 (55) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5082a199;rport (66) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 104 BYE (13) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Max-Forwards: 70 (16) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 8: X-Asterisk-HangupCause: Normal Clearing (39) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 9: X-Asterisk-HangupCauseCode: 16 (30) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 10: Content-Length: 0 (17) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 11: (0) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:2114 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 Scheduling destruction of SIP dialog '2a8af3779cf82b2c' in 32000 ms (Method: ACK) Really destroying SIP dialog '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' Method: BYE [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11868 sip_dump_history: ---------- SIP HISTORY for '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11872 sip_dump_history: * SIP Call [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 001. NewChan Channel SIP/12347-00000008 - from 7040c7d832ae16de26c7decb02a5e [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 002. TxReqRel INVITE / 102 INVITE - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 004. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 005. TxReq ACK / 102 ACK - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 006. ReInv Re-invite sent [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 007. TxReqRel INVITE / 103 INVITE - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 008. Rx SIP/2.0 / 103 INVITE / 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 009. TxReq ACK / 103 ACK - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 010. Rx BYE / 2 BYE / sip:14237@192.168.222.160 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 011. RTCPaudio Quality:ssrc=757698489;themssrc=0;lp=1;rxjitter=0.000000;rxcoun [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 012. TxResp SIP/2.0 / 2 BYE - 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 013. Hangup Cause Normal Clearing [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11878 sip_dump_history: ---------- END SIP HISTORY for '7040c7d832ae16de26c7decb02a5e6fa@192.168.222.160' (none)*CLI> <--- SIP read from 172.16.102.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5082a199;rport=5060;received=192.168.222.160 From: "12347" ;tag=as24c56fcc To: "x" ;tag=fded0efa50 Call-ID: 2a8af3779cf82b2c CSeq: 104 BYE Server: Aastra 53i/2.5.2.1010 Content-Length: 0 <-------------> [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 0: SIP/2.0 200 OK (14) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK5082a199;rport=5060;received=192.168.222.160 (96) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 2: From: "12347" ;tag=as24c56fcc (56) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 3: To: "x" ;tag=fded0efa50 (50) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 4: Call-ID: 2a8af3779cf82b2c (25) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 5: CSeq: 104 BYE (13) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 6: Server: Aastra 53i/2.5.2.1010 (29) [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:5053 parse_request: Header 7: Content-Length: 0 (17) --- (8 headers 0 lines) --- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:2247 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #75 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:2271 __sip_ack: Stopping retransmission on '2a8af3779cf82b2c' of Request 104: Match Found Really destroying SIP dialog '2a8af3779cf82b2c' Method: ACK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11868 sip_dump_history: ---------- SIP HISTORY for '2a8af3779cf82b2c' [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11872 sip_dump_history: * SIP Call [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 001. Rx INVITE / 29923 INVITE / sip:12347@192.168.222.160 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 002. NewChan Channel SIP/14237-00000007 - from 2a8af3779cf82b2c [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 003. TxResp SIP/2.0 / 29923 INVITE - 100 Trying [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 004. TxResp SIP/2.0 / 29923 INVITE - 180 Ringing [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 005. TxRespRel SIP/2.0 / 29923 INVITE - 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 006. Rx ACK / 29923 ACK / sip:12347@192.168.222.160 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 007. ReInv Re-invite sent [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 008. TxReqRel INVITE / 102 INVITE - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 009. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 010. TxReq ACK / 102 ACK - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 011. Rx INVITE / 29924 INVITE / sip:12347@192.168.222.160 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 012. ReInv Re-invite received [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 013. TxResp SIP/2.0 / 29924 INVITE - 100 Trying [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 014. TxRespRel SIP/2.0 / 29924 INVITE - 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 015. Rx ACK / 29924 ACK / sip:12347@192.168.222.160 [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 016. ReInv Re-invite sent [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 017. TxReqRel INVITE / 103 INVITE - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 018. Hangup Cause Normal Clearing [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 019. SchedDestroy 32000 ms [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 020. CancelDestroy [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 021. Rx SIP/2.0 / 103 INVITE / 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 022. TxReq ACK / 103 ACK - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 023. TxReqRel BYE / 104 BYE - -UNKNOWN- [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 024. SchedDestroy 32000 ms [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11875 sip_dump_history: 025. Rx SIP/2.0 / 104 BYE / 200 OK [Nov 18 10:29:20] DEBUG[19213]: chan_sip.c:11878 sip_dump_history: ---------- END SIP HISTORY for '2a8af3779cf82b2c' (none)*CLI>