[Jan 29 18:16:38] VERBOSE[3073] config.c: == Parsing '/etc/asterisk/logger.conf': [Jan 29 18:16:38] DEBUG[3073] config.c: Parsing /etc/asterisk/logger.conf [Jan 29 18:16:38] VERBOSE[3073] config.c: == Found [Jan 29 18:16:38] VERBOSE[3073] logger.c: Asterisk Queue Logger restarted [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.103:5062 ---> INVITE sip:cit02@sax3n03.local:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK694346591 From: ;tag=928937303 To: Call-ID: 668537057 CSeq: 20 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Subject: Phone call Content-Length: 423 v=0 o=cit03 123456 654321 IN IP4 192.168.12.103 s=A conversation c=IN IP4 192.168.12.103 t=0 0 m=audio 7080 RTP/AVP 112 111 110 3 0 8 101 a=rtpmap:112 speex/32000/1 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000/1 a=fmtp:111 vbr=vad a=rtpmap:110 speex/8000/1 a=fmtp:110 vbr=vad a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 0 [ 43]: INVITE sip:cit02@sax3n03.local:5062 SIP/2.0 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK694346591 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 2 [ 50]: From: ;tag=928937303 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 3 [ 34]: To: [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 4 [ 18]: Call-ID: 668537057 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 5 [ 15]: CSeq: 20 INVITE [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 6 [ 40]: Contact: [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 10 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 11 [ 19]: Subject: Phone call [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 12 [ 19]: Content-Length: 423 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 13 [ 0]: [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit03 123456 654321 IN IP4 192.168.12.103 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.103 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 5 [ 42]: m=audio 7080 RTP/AVP 112 111 110 3 0 8 101 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 6 [ 26]: a=rtpmap:112 speex/32000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 7 [ 17]: a=fmtp:112 vbr=on [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 8 [ 26]: a=rtpmap:111 speex/16000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 9 [ 18]: a=fmtp:111 vbr=vad [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 10 [ 25]: a=rtpmap:110 speex/8000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 11 [ 18]: a=fmtp:110 vbr=vad [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 12 [ 21]: a=rtpmap:3 GSM/8000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 13 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 14 [ 22]: a=rtpmap:8 PCMA/8000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 15 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-11 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Body 17 [ 10]: a=sendrecv [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: --- (13 headers 18 lines) --- [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: = Looking for Call ID: 668537057 (Checking From) --From tag 928937303 --To-tag [Jan 29 18:16:44] DEBUG[3062] acl.c: For destination '192.168.12.103', our source address is '192.168.12.103'. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.12.103:5060 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Allocating new SIP dialog for 668537057 - INVITE (No RTP) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 29 18:16:44] DEBUG[3062] netsock2.c: Splitting '192.168.12.103:5062' gives... [Jan 29 18:16:44] DEBUG[3062] netsock2.c: ...host '192.168.12.103' and port '5062'. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Sending to 192.168.12.103:5062 (no NAT) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Initializing initreq for method INVITE - callid 668537057 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Using INVITE request as basis request - 668537057 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found peer 'cit03' for 'cit03' from 192.168.12.103:5062 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9fbe350' [Jan 29 18:16:44] DEBUG[3062] res_rtp_asterisk.c: Allocated port 10882 for RTP instance '0x9fbe350' [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: RTP instance '0x9fbe350' is setup and ready to go [Jan 29 18:16:44] DEBUG[3062] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9fbe350' [Jan 29 18:16:44] VERBOSE[3062] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Setting NAT on RTP to Off [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing session-level SDP o=cit03 123456 654321 IN IP4 192.168.12.103... UNSUPPORTED. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jan 29 18:16:44] DEBUG[3062] netsock2.c: Splitting '192.168.12.103' gives... [Jan 29 18:16:44] DEBUG[3062] netsock2.c: ...host '192.168.12.103' and port '(null)'. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.12.103... OK. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 112 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 112 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 111 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 111 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 110 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 110 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 3 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 3 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 0 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 0 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 8 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 8 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found RTP audio format 101 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Setting payload 101 based on m type on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format speex for ID 112 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 speex/32000/1... OK. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=fmtp:112 vbr=on... UNSUPPORTED. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format speex for ID 111 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 speex/16000/1... OK. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 vbr=vad... UNSUPPORTED. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format speex for ID 110 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000/1... OK. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 vbr=vad... UNSUPPORTED. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format GSM for ID 3 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000/1... OK. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000/1... OK. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK. [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000/1... OK. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 0 on 0xb71a8e90 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 3 on 0xb71a8e90 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 8 on 0xb71a8e90 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 101 on 0xb71a8e90 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 110 on 0xb71a8e90 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 111 on 0xb71a8e90 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Incorporating payload 112 on 0xb71a8e90 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x20000021e (gsm|ulaw|alaw|speex|speex16|g726aal2)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 29 18:16:44] DEBUG[3062] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9fbe350' [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Peer audio RTP is at port 192.168.12.103:7080 [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 0 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 3 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 8 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 101 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 110 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 111 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] rtp_engine.c: Copying payload 112 from 0xb71a8e90 to 0x9fbe4fc [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Checking SIP call limits for device [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Updating call counter for incoming call [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: Looking for cit02 in sax3nlocal (domain sax3n03.local:5062) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: This channel will not be able to handle video. [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: build_route: Contact hop: [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: list_route: hop: [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: SIP/cit03-00000000: New call is still down.... Trying... [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK694346591;received=192.168.12.103;rport=5062 From: ;tag=928937303 To: Call-ID: 668537057 CSeq: 20 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:16:44] DEBUG[3054] devicestate.c: No provider found, checking channel drivers for SIP - cit03 [Jan 29 18:16:44] DEBUG[3054] chan_sip.c: Checking device state for peer cit03 [Jan 29 18:16:44] DEBUG[3054] devicestate.c: Changing state for SIP/cit03 - state 1 (Not in use) [Jan 29 18:16:44] DEBUG[3054] devicestate.c: device 'SIP/cit03' state '1' [Jan 29 18:16:44] DEBUG[3074] pbx.c: Result of 'EXTEN' is 'cit02' [Jan 29 18:16:44] DEBUG[3074] pbx.c: Launching 'NoOp' [Jan 29 18:16:44] VERBOSE[3074] pbx.c: -- Executing [cit02@sax3nlocal:1] NoOp("SIP/cit03-00000000", "Call to cit02") in new stack [Jan 29 18:16:44] DEBUG[3074] pbx.c: Result of 'EXTEN' is 'cit02' [Jan 29 18:16:44] DEBUG[3074] pbx.c: Launching 'Dial' [Jan 29 18:16:44] VERBOSE[3074] pbx.c: -- Executing [cit02@sax3nlocal:2] Dial("SIP/cit03-00000000", "SIP/cit02") in new stack [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Allocating new SIP dialog for 1b8f73cf6e2c9d226401c0046c63285b@192.168.12.103:0 - INVITE (No RTP) [Jan 29 18:16:44] DEBUG[3074] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9fb18f8' [Jan 29 18:16:44] DEBUG[3074] res_rtp_asterisk.c: Allocated port 16156 for RTP instance '0x9fb18f8' [Jan 29 18:16:44] DEBUG[3074] rtp_engine.c: RTP instance '0x9fb18f8' is setup and ready to go [Jan 29 18:16:44] DEBUG[3074] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9fb18f8' [Jan 29 18:16:44] VERBOSE[3074] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Setting NAT on RTP to Off [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 29 18:16:44] DEBUG[3074] acl.c: For destination '192.168.12.102', our source address is '192.168.12.103'. [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.12.103:5060 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: This channel will not be able to handle video. [Jan 29 18:16:44] DEBUG[3074] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 29 18:16:44] DEBUG[3074] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 29 18:16:44] DEBUG[3074] rtp_engine.c: Seeded SDP of 'SIP/cit02-00000001' with that of 'SIP/cit03-00000000' [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable DIALEDTIME. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable ANSWEREDTIME. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable DIALEDPEERNAME. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable DIALEDPEERNUMBER. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable DIALSTATUS. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable SIPCALLID. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable SIPDOMAIN. [Jan 29 18:16:44] DEBUG[3074] channel.c: Not copying variable SIPURI. [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Outgoing Call for [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Updating call counter for outgoing call [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 29 18:16:44] VERBOSE[3074] chan_sip.c: Audio is at 5060 [Jan 29 18:16:44] VERBOSE[3074] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 29 18:16:44] VERBOSE[3074] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: -- Done with adding codecs to SDP [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Initializing initreq for method INVITE - callid 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 0 [ 37]: INVITE sip:sax3n02.local:5062 SIP/2.0 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 3 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 4 [ 28]: To: [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 5 [ 40]: Contact: [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 6 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 8 [ 30]: User-Agent: Asterisk PBX 1.8.0 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 9 [ 35]: Date: Mon, 29 Jan 2007 18:16:44 GMT [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 29 18:16:44] VERBOSE[3074] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.12.102:5062: INVITE sip:sax3n02.local:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0 Date: Mon, 29 Jan 2007 18:16:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 238 v=0 o=root 1412403037 1412403037 IN IP4 192.168.12.103 s=Asterisk PBX 1.8.0 c=IN IP4 192.168.12.103 t=0 0 m=audio 16156 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #28 [Jan 29 18:16:44] DEBUG[3074] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:16:44] VERBOSE[3074] app_dial.c: -- Called cit02 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 3 [ 28]: To: [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 6 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 29 18:16:44] VERBOSE[3062] chan_sip.c: --- (8 headers 0 lines) --- [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: *** SIP TIMER: Cancelling retransmission #28 - INVITE (got response) [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Request 102: Found [Jan 29 18:16:44] DEBUG[3062] chan_sip.c: SIP response 100 to standard invite [Jan 29 18:16:45] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 101 Dialog Establishement Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 0 [ 33]: SIP/2.0 101 Dialog Establishement [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 6 [ 34]: Contact: [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 7 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 29 18:16:45] VERBOSE[3062] chan_sip.c: --- (9 headers 0 lines) --- [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Request 102: Found [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: SIP response 101 to standard invite [Jan 29 18:16:45] DEBUG[3071] app_queue.c: Device 'SIP/cit03' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:16:45] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> OPTIONS sip:192.168.12.103:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK1982414371 Route: "cit03" ;tag=as635524c7 From: ;tag=1438103931 To: "cit03" ;tag=as635524c7 Call-ID: 1346595893 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Expires: 120 Content-Length: 0 <-------------> [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 0 [ 39]: OPTIONS sip:192.168.12.103:5060 SIP/2.0 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK1982414371 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 2 [ 56]: Route: "cit03" ;tag=as635524c7 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 3 [ 45]: From: ;tag=1438103931 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 4 [ 53]: To: "cit03" ;tag=as635524c7 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 5 [ 19]: Call-ID: 1346595893 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 6 [ 16]: CSeq: 20 OPTIONS [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 7 [ 23]: Accept: application/sdp [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 9 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 10 [ 12]: Expires: 120 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 29 18:16:45] VERBOSE[3062] chan_sip.c: --- (12 headers 0 lines) --- [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: = Looking for Call ID: 1346595893 (Checking From) --From tag 1438103931 --To-tag as635524c7 [Jan 29 18:16:45] DEBUG[3062] acl.c: For destination '192.168.12.102', our source address is '192.168.12.103'. [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.12.103:5060 [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Allocating new SIP dialog for 1346595893 - OPTIONS (No RTP) [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jan 29 18:16:45] VERBOSE[3062] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.102:5062 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK1982414371;received=192.168.12.102 From: ;tag=1438103931 To: "cit03" ;tag=as635524c7 Call-ID: 1346595893 CSeq: 20 OPTIONS Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 29 18:16:45] DEBUG[3062] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:16:45] VERBOSE[3062] chan_sip.c: Scheduling destruction of SIP dialog '1346595893' in 32000 ms (Method: OPTIONS) [Jan 29 18:16:46] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 6 [ 34]: Contact: [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 7 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 29 18:16:46] VERBOSE[3062] chan_sip.c: --- (9 headers 0 lines) --- [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Request 102: Found [Jan 29 18:16:46] DEBUG[3062] chan_sip.c: SIP response 180 to standard invite [Jan 29 18:16:46] DEBUG[3054] devicestate.c: No provider found, checking channel drivers for SIP - cit02 [Jan 29 18:16:46] DEBUG[3054] chan_sip.c: Checking device state for peer cit02 [Jan 29 18:16:46] DEBUG[3054] devicestate.c: Changing state for SIP/cit02 - state 1 (Not in use) [Jan 29 18:16:46] DEBUG[3054] devicestate.c: device 'SIP/cit02' state '1' [Jan 29 18:16:46] VERBOSE[3074] app_dial.c: -- SIP/cit02-00000001 is ringing [Jan 29 18:16:46] DEBUG[3074] rtp_engine.c: Setting early bridge SDP of 'SIP/cit03-00000000' with that of 'SIP/cit02-00000001' [Jan 29 18:16:46] VERBOSE[3074] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK694346591;received=192.168.12.103;rport=5062 From: ;tag=928937303 To: ;tag=as71243d01 Call-ID: 668537057 CSeq: 20 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 29 18:16:46] DEBUG[3074] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:16:46] DEBUG[3071] app_queue.c: Device 'SIP/cit02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Acked pending invite 102 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Found [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: SIP response 200 to standard invite [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing session-level SDP o=cit02 123456 654321 IN IP4 192.168.12.102... UNSUPPORTED. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jan 29 18:16:48] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:16:48] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.12.102... OK. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Found RTP audio format 0 [Jan 29 18:16:48] DEBUG[3062] rtp_engine.c: Setting payload 0 based on m type on 0xb71a9470 [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Found RTP audio format 101 [Jan 29 18:16:48] DEBUG[3062] rtp_engine.c: Setting payload 101 based on m type on 0xb71a9470 [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000/1... OK. [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000/1... OK. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 29 18:16:48] DEBUG[3062] rtp_engine.c: Incorporating payload 0 on 0xb71a9470 [Jan 29 18:16:48] DEBUG[3062] rtp_engine.c: Incorporating payload 101 on 0xb71a9470 [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 29 18:16:48] DEBUG[3062] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9fb18f8' [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Peer audio RTP is at port 192.168.12.102:7080 [Jan 29 18:16:48] DEBUG[3062] rtp_engine.c: Copying payload 0 from 0xb71a9470 to 0x9fb1aa4 [Jan 29 18:16:48] DEBUG[3062] rtp_engine.c: Copying payload 101 from 0xb71a9470 to 0x9fb1aa4 [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: We have an owner, now see if we need to change this call [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Updating call counter for outgoing call [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: build_route: Contact hop: [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: list_route: hop: [Jan 29 18:16:48] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:16:48] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:16:48] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:16:48] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:16:48] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK6e411c60 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:16:48] VERBOSE[3074] app_dial.c: -- SIP/cit02-00000001 answered SIP/cit03-00000000 [Jan 29 18:16:48] DEBUG[3074] rtp_engine.c: Setting early bridge SDP of 'SIP/cit03-00000000' with that of 'SIP/cit02-00000001' [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: SIP answering channel: SIP/cit03-00000000 [Jan 29 18:16:48] DEBUG[3074] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: Setting framing from config on incoming call [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 29 18:16:48] VERBOSE[3074] chan_sip.c: Audio is at 5060 [Jan 29 18:16:48] VERBOSE[3074] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 29 18:16:48] VERBOSE[3074] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: -- Done with adding codecs to SDP [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 29 18:16:48] VERBOSE[3074] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK694346591;received=192.168.12.103;rport=5062 From: ;tag=928937303 To: ;tag=as71243d01 Call-ID: 668537057 CSeq: 20 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 427236093 427236093 IN IP4 192.168.12.103 s=Asterisk PBX 1.8.0 c=IN IP4 192.168.12.103 t=0 0 m=audio 10882 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #32 [Jan 29 18:16:48] DEBUG[3074] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:16:48] DEBUG[3074] features.c: bridge answer set, chan answer set [Jan 29 18:16:48] DEBUG[3074] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 29 18:16:48] DEBUG[3074] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 29 18:16:48] VERBOSE[3074] rtp_engine.c: -- Locally bridging SIP/cit03-00000000 and SIP/cit02-00000001 [Jan 29 18:16:48] DEBUG[3054] devicestate.c: No provider found, checking channel drivers for SIP - cit02 [Jan 29 18:16:48] DEBUG[3054] chan_sip.c: Checking device state for peer cit02 [Jan 29 18:16:48] DEBUG[3054] devicestate.c: Changing state for SIP/cit02 - state 1 (Not in use) [Jan 29 18:16:48] DEBUG[3054] devicestate.c: device 'SIP/cit02' state '1' [Jan 29 18:16:48] DEBUG[3054] devicestate.c: No provider found, checking channel drivers for SIP - cit03 [Jan 29 18:16:48] DEBUG[3054] chan_sip.c: Checking device state for peer cit03 [Jan 29 18:16:48] DEBUG[3054] devicestate.c: Changing state for SIP/cit03 - state 1 (Not in use) [Jan 29 18:16:48] DEBUG[3054] devicestate.c: device 'SIP/cit03' state '1' [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:48] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: INVITE [Jan 29 18:16:48] DEBUG[3071] app_queue.c: Device 'SIP/cit02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:16:48] DEBUG[3071] app_queue.c: Device 'SIP/cit03' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:16:49] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.103:5062 ---> ACK sip:cit02@192.168.12.103:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1260607958 From: ;tag=928937303 To: ;tag=as71243d01 Call-ID: 668537057 CSeq: 20 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 0 [ 41]: ACK sip:cit02@192.168.12.103:5060 SIP/2.0 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1260607958 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 2 [ 50]: From: ;tag=928937303 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 3 [ 49]: To: ;tag=as71243d01 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 4 [ 18]: Call-ID: 668537057 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 5 [ 12]: CSeq: 20 ACK [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 6 [ 40]: Contact: [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 29 18:16:49] VERBOSE[3062] chan_sip.c: --- (10 headers 0 lines) --- [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: = Looking for Call ID: 668537057 (Checking From) --From tag 928937303 --To-tag as71243d01 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #32 [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Stopping retransmission on '668537057' of Response 20: Match Found [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:49] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:50] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:16:50] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:50] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:16:50] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:16:50] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:16:50] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:16:50] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK6ab73940 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:50] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:51] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:51] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:52] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:16:52] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:52] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:16:52] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:16:52] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:16:52] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:16:52] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK1c73b0d1 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:52] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:53] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:53] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:53] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:16:54] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:16:54] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:54] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:55] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:16:55] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:16:55] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:16:55] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:16:55] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:16:55] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:16:55] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK2ac14615 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:55] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:56] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:56] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:57] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:57] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:58] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:58] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:16:58] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:16:59] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:16:59] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:16:59] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:00] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:17:00] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:00] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:00] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:17:00] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:17:00] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:17:00] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK2b6270fe Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:00] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:01] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:01] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:02] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:02] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:03] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:03] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:03] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:04] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:04] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:04] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:05] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:17:05] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:05] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:05] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:17:05] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:17:05] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:17:05] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK0e5f5e62 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:05] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:06] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:06] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:07] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:07] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:08] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:08] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:08] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:09] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:09] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:09] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:10] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:17:10] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:10] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:10] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:17:10] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:17:10] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:17:10] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK311f67f5 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:10] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:11] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:11] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:12] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:12] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:13] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:13] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:13] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:14] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:14] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:14] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:15] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:17:15] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:15] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:15] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:17:15] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:17:15] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:17:15] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK63f923f9 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:15] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:16] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:16] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:17] DEBUG[3062] chan_sip.c: Auto destroying SIP dialog '1346595893' [Jan 29 18:17:17] DEBUG[3062] chan_sip.c: Destroying SIP dialog 1346595893 [Jan 29 18:17:17] VERBOSE[3062] chan_sip.c: Really destroying SIP dialog '1346595893' Method: OPTIONS [Jan 29 18:17:17] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:17] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:18] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:18] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:18] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:19] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:19] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:19] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:20] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:17:20] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:20] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:20] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:17:20] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:17:20] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:17:20] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK76bd8cda Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:20] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:21] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:21] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:22] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:22] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:23] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:23] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:23] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:24] DEBUG[3074] res_rtp_asterisk.c: Got RTCP report of 144 bytes [Jan 29 18:17:24] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:24] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:25] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5a51dfb3 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as635524c7 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 3 [ 43]: To: ;tag=1224461129 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 6 [ 29]: Contact: [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 9 [ 19]: Content-Length: 218 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Header 10 [ 0]: [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:17:25] VERBOSE[3062] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking To) --From tag as635524c7 --To-tag 1224461129 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Stopping retransmission on '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' of Request 102: Match Not Found [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Strict routing enforced for session 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:25] VERBOSE[3062] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:25] DEBUG[3062] netsock2.c: Splitting '192.168.12.102' gives... [Jan 29 18:17:25] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '(null)'. [Jan 29 18:17:25] VERBOSE[3062] chan_sip.c: set_destination: set destination to 192.168.12.102:5060 [Jan 29 18:17:25] VERBOSE[3062] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5060: ACK sip:192.168.12.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK27f8ce93 Max-Forwards: 70 From: "cit03" ;tag=as635524c7 To: ;tag=1224461129 Contact: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5060 [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: INVITE [Jan 29 18:17:25] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> BYE sip:cit03@192.168.12.103:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK1439962954 From: ;tag=1224461129 To: "cit03" ;tag=as635524c7 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 2 BYE Contact: Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 0 [ 41]: BYE sip:cit03@192.168.12.103:5060 SIP/2.0 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK1439962954 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 2 [ 45]: From: ;tag=1224461129 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 3 [ 53]: To: "cit03" ;tag=as635524c7 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 4 [ 61]: Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 5 [ 11]: CSeq: 2 BYE [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 6 [ 41]: Contact: [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: --- (10 headers 0 lines) --- [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: = Looking for Call ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 (Checking From) --From tag 1224461129 --To-tag as635524c7 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Initializing initreq for method BYE - callid 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:26] DEBUG[3062] netsock2.c: Splitting '192.168.12.102:5062' gives... [Jan 29 18:17:26] DEBUG[3062] netsock2.c: ...host '192.168.12.102' and port '5062'. [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: Sending to 192.168.12.102:5062 (no NAT) [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Setting SIP_ALREADYGONE on dialog 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:26] DEBUG[3062] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9fb18f8' [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: Scheduling destruction of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' in 32000 ms (Method: BYE) [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Received bye, issuing owner hangup [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.102:5062;branch=z9hG4bK1439962954;received=192.168.12.102;rport=5062 From: ;tag=1224461129 To: "cit03" ;tag=as635524c7 Call-ID: 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 CSeq: 2 BYE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060' Method: BYE [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '668537057' Method: ACK [Jan 29 18:17:26] DEBUG[3074] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [Jan 29 18:17:26] DEBUG[3074] channel.c: Returning from native bridge, channels: SIP/cit03-00000000, SIP/cit02-00000001 [Jan 29 18:17:26] DEBUG[3074] pbx.c: Launching 'NoOp' [Jan 29 18:17:26] VERBOSE[3074] pbx.c: -- Executing [h@sax3nlocal:1] NoOp("SIP/cit03-00000000", "HANGUP!!!!") in new stack [Jan 29 18:17:26] DEBUG[3074] channel.c: Hanging up channel 'SIP/cit02-00000001' [Jan 29 18:17:26] DEBUG[3074] chan_sip.c: Hangup call SIP/cit02-00000001, SIP callid 48d016d22a0326e3111403d77c63fa9f@192.168.12.103:5060 [Jan 29 18:17:26] DEBUG[3074] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9fb18f8' [Jan 29 18:17:26] DEBUG[3074] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 29 18:17:26] DEBUG[3074] pbx.c: Spawn extension (sax3nlocal,cit02,2) exited non-zero on 'SIP/cit03-00000000' [Jan 29 18:17:26] VERBOSE[3074] pbx.c: == Spawn extension (sax3nlocal, cit02, 2) exited non-zero on 'SIP/cit03-00000000' [Jan 29 18:17:26] DEBUG[3074] channel.c: Soft-Hanging up channel 'SIP/cit03-00000000' [Jan 29 18:17:26] DEBUG[3074] channel.c: Hanging up channel 'SIP/cit03-00000000' [Jan 29 18:17:26] DEBUG[3074] chan_sip.c: Hangup call SIP/cit03-00000000, SIP callid 668537057 [Jan 29 18:17:26] DEBUG[3074] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9fbe350' [Jan 29 18:17:26] VERBOSE[3074] chan_sip.c: Scheduling destruction of SIP dialog '668537057' in 32000 ms (Method: ACK) [Jan 29 18:17:26] DEBUG[3074] chan_sip.c: Strict routing enforced for session 668537057 [Jan 29 18:17:26] VERBOSE[3074] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:17:26] DEBUG[3074] netsock2.c: Splitting '192.168.12.103:5062' gives... [Jan 29 18:17:26] DEBUG[3074] netsock2.c: ...host '192.168.12.103' and port '5062'. [Jan 29 18:17:26] VERBOSE[3074] chan_sip.c: set_destination: set destination to 192.168.12.103:5062 [Jan 29 18:17:26] VERBOSE[3074] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.12.103:5062: BYE sip:cit03@192.168.12.103:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK6caeaf10;rport Max-Forwards: 70 From: ;tag=as71243d01 To: ;tag=928937303 Call-ID: 668537057 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 29 18:17:26] DEBUG[3074] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Jan 29 18:17:26] DEBUG[3074] chan_sip.c: Trying to put 'BYE sip:cit' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:17:26] DEBUG[3054] devicestate.c: No provider found, checking channel drivers for SIP - cit02 [Jan 29 18:17:26] DEBUG[3054] chan_sip.c: Checking device state for peer cit02 [Jan 29 18:17:26] DEBUG[3054] devicestate.c: Changing state for SIP/cit02 - state 1 (Not in use) [Jan 29 18:17:26] DEBUG[3054] devicestate.c: device 'SIP/cit02' state '1' [Jan 29 18:17:26] DEBUG[3054] devicestate.c: No provider found, checking channel drivers for SIP - cit03 [Jan 29 18:17:26] DEBUG[3054] chan_sip.c: Checking device state for peer cit03 [Jan 29 18:17:26] DEBUG[3054] devicestate.c: Changing state for SIP/cit03 - state 1 (Not in use) [Jan 29 18:17:26] DEBUG[3054] devicestate.c: device 'SIP/cit03' state '1' [Jan 29 18:17:26] DEBUG[3071] app_queue.c: Device 'SIP/cit02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:17:26] DEBUG[3071] app_queue.c: Device 'SIP/cit03' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: <--- SIP read from UDP:192.168.12.103:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK6caeaf10;rport=5060 From: ;tag=as71243d01 To: ;tag=928937303 Call-ID: 668537057 CSeq: 102 BYE User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK6caeaf10;rport=5060 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 2 [ 51]: From: ;tag=as71243d01 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 3 [ 48]: To: ;tag=928937303 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 4 [ 18]: Call-ID: 668537057 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 6 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: --- (8 headers 0 lines) --- [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: = Looking for Call ID: 668537057 (Checking To) --From tag as71243d01 --To-tag 928937303 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Stopping retransmission on '668537057' of Request 102: Match Found [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Jan 29 18:17:26] DEBUG[3062] chan_sip.c: Destroying SIP dialog 668537057 [Jan 29 18:17:26] VERBOSE[3062] chan_sip.c: Really destroying SIP dialog '668537057' Method: ACK [Jan 29 18:17:26] DEBUG[3062] rtp_engine.c: Destroyed RTP instance '0x9fbe350'