[Feb 22 09:33:06] VERBOSE[19117] config.c: == Parsing '/etc/asterisk/logger.conf': [Feb 22 09:33:06] DEBUG[19117] config.c: Parsing /etc/asterisk/logger.conf [Feb 22 09:33:06] VERBOSE[19117] config.c: == Found [Feb 22 09:33:06] VERBOSE[19117] logger.c: Asterisk Queue Logger restarted [Feb 22 09:33:08] DEBUG[19144] chan_sip.c: Auto destroying SIP dialog '000e381b-9dbc0002-740fca6e-66ceb8de@10.10.0.122' [Feb 22 09:33:08] DEBUG[19144] chan_sip.c: Destroying SIP dialog 000e381b-9dbc0002-740fca6e-66ceb8de@10.10.0.122 [Feb 22 09:33:08] VERBOSE[19144] chan_sip.c: Really destroying SIP dialog '000e381b-9dbc0002-740fca6e-66ceb8de@10.10.0.122' Method: REGISTER [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.32.165:4343 ---> INVITE sip:01@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK242f7b44;rport Max-Forwards: 70 From: "069911160036" ;tag=as2e19f809 To: Contact: Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 22 Feb 2011 08:33:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 389 v=0 o=root 1226093537 1226093537 IN IP4 83.136.32.165 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.165 b=CT:128 t=0 0 m=audio 10636 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 10242 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=sendrecv <-------------> [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 0 [ 40]: INVITE sip:01@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK242f7b44;rport [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 3 [ 73]: From: "069911160036" ;tag=as2e19f809 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 4 [ 31]: To: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 5 [ 46]: Contact: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 6 [ 60]: Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.3 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 9 [ 35]: Date: Tue, 22 Feb 2011 08:33:12 GMT [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 13 [ 19]: Content-Length: 389 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 14 [ 0]: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 1 [ 49]: o=root 1226093537 1226093537 IN IP4 83.136.32.165 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.2.3 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 3 [ 22]: c=IN IP4 83.136.32.165 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 4 [ 8]: b=CT:128 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 5 [ 5]: t=0 0 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 6 [ 31]: m=audio 10636 RTP/AVP 8 3 0 101 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 13 [ 10]: a=sendrecv [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 14 [ 27]: m=video 10242 RTP/AVP 34 98 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 15 [ 22]: a=rtpmap:34 H263/90000 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 16 [ 27]: a=rtpmap:98 h263-1998/90000 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Body 17 [ 10]: a=sendrecv [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: --- (14 headers 18 lines) --- [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: = Looking for Call ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 (Checking From) --From tag as2e19f809 --To-tag [Feb 22 09:33:12] DEBUG[19144] acl.c: For destination '83.136.32.165', our source address is '83.136.32.138'. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 83.136.32.138:4343 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Allocating new SIP dialog for 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 - INVITE (No RTP) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 09:33:12] DEBUG[19144] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer" [Feb 22 09:33:12] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -replaces- [Feb 22 09:33:12] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: replaces [Feb 22 09:33:12] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -timer- [Feb 22 09:33:12] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: timer [Feb 22 09:33:12] DEBUG[19144] netsock2.c: Splitting '83.136.32.165:4343' gives... [Feb 22 09:33:12] DEBUG[19144] netsock2.c: ...host '83.136.32.165' and port '4343'. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Sending to 83.136.32.165:4343 (no NAT) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Initializing initreq for method INVITE - callid 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Using INVITE request as basis request - 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found peer 'trunk_to_nvst' for '069911160036' from 83.136.32.165:4343 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc974c10' [Feb 22 09:33:12] DEBUG[19144] res_rtp_asterisk.c: Allocated port 10034 for RTP instance '0xc974c10' [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: RTP instance '0xc974c10' is setup and ready to go [Feb 22 09:33:12] DEBUG[19144] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc974c10' [Feb 22 09:33:12] VERBOSE[19144] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Setting NAT on RTP to Off [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing session-level SDP o=root 1226093537 1226093537 IN IP4 83.136.32.165... UNSUPPORTED. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing session-level SDP s=Asterisk PBX 1.8.2.3... UNSUPPORTED. [Feb 22 09:33:12] DEBUG[19144] netsock2.c: Splitting '83.136.32.165' gives... [Feb 22 09:33:12] DEBUG[19144] netsock2.c: ...host '83.136.32.165' and port '(null)'. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing session-level SDP c=IN IP4 83.136.32.165... OK. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing session-level SDP b=CT:128... UNSUPPORTED. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found RTP audio format 8 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Setting payload 8 based on m type on 0xb3d523cc [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found RTP audio format 3 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Setting payload 3 based on m type on 0xb3d523cc [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found RTP audio format 0 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Setting payload 0 based on m type on 0xb3d523cc [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found RTP audio format 101 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Setting payload 101 based on m type on 0xb3d523cc [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found audio description format GSM for ID 3 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found audio description format PCMU for ID 0 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found RTP video format 34 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Setting payload 34 based on m type on 0xb3d5174c [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found RTP video format 98 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Setting payload 98 based on m type on 0xb3d5174c [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found video description format H263 for ID 34 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:34 H263/90000... OK. [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Found video description format h263-1998 for ID 98 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (video) SDP a=rtpmap:98 h263-1998/90000... OK. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Processing media-level (video) SDP a=sendrecv... OK. [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Incorporating payload 0 on 0xb3d523cc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Incorporating payload 3 on 0xb3d523cc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Incorporating payload 8 on 0xb3d523cc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Incorporating payload 101 on 0xb3d523cc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Incorporating payload 34 on 0xb3d5174c [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Incorporating payload 98 on 0xb3d5174c [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x8000e (gsm|ulaw|alaw|h263) [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 22 09:33:12] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc974c10' [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Peer audio RTP is at port 83.136.32.165:10636 [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Copying payload 0 from 0xb3d523cc to 0xc974dbc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Copying payload 3 from 0xb3d523cc to 0xc974dbc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Copying payload 8 from 0xb3d523cc to 0xc974dbc [Feb 22 09:33:12] DEBUG[19144] rtp_engine.c: Copying payload 101 from 0xb3d523cc to 0xc974dbc [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: We're settling with these formats: 0x8000e (gsm|ulaw|alaw|h263) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Checking SIP call limits for device [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Updating call counter for incoming call [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: Looking for 01 in from_nvst (domain 83.136.32.138:4343) [Feb 22 09:33:12] DEBUG[19144] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: *** Our native formats are 0x80004 (ulaw|h263) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: *** Joint capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Feb 22 09:33:12] DEBUG[19144] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: This channel will not be able to handle video. [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: build_route: Contact hop: [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: list_route: hop: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: SIP/trunk_to_nvst-00000000: New call is still down.... Trying... [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: <--- Transmitting (no NAT) to 83.136.32.165:4343 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK242f7b44;received=83.136.32.165;rport=4343 From: "069911160036" ;tag=as2e19f809 To: Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 83.136.32.165:4343 [Feb 22 09:33:12] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - trunk_to_nvst [Feb 22 09:33:12] DEBUG[19122] chan_sip.c: Checking device state for peer trunk_to_nvst [Feb 22 09:33:12] DEBUG[19122] devicestate.c: Changing state for SIP/trunk_to_nvst - state 1 (Not in use) [Feb 22 09:33:12] DEBUG[19122] devicestate.c: device 'SIP/trunk_to_nvst' state '1' [Feb 22 09:33:12] DEBUG[19156] app_queue.c: Device 'SIP/trunk_to_nvst' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:12] DEBUG[19158] pbx.c: Function result is '069911160036' [Feb 22 09:33:12] DEBUG[19158] pbx.c: Expression result is '0' [Feb 22 09:33:12] DEBUG[19158] pbx.c: Function result is '069911160036' [Feb 22 09:33:12] DEBUG[19158] pbx.c: Launching 'ExecIf' [Feb 22 09:33:12] VERBOSE[19158] pbx.c: -- Executing [01@from_nvst:1] ExecIf("SIP/trunk_to_nvst-00000000", "0?Set(CALLERID(num)=00911160036)") in new stack [Feb 22 09:33:12] DEBUG[19158] pbx.c: Result of 'EXTEN' is '01' [Feb 22 09:33:12] DEBUG[19158] pbx.c: Launching 'Goto' [Feb 22 09:33:12] VERBOSE[19158] pbx.c: -- Executing [01@from_nvst:2] Goto("SIP/trunk_to_nvst-00000000", "from_nvst_cliok,01,1") in new stack [Feb 22 09:33:12] VERBOSE[19158] pbx.c: -- Goto (from_nvst_cliok,01,1) [Feb 22 09:33:12] DEBUG[19158] pbx.c: Result of 'EXTEN' is '01' [Feb 22 09:33:12] DEBUG[19158] pbx.c: Launching 'Dial' [Feb 22 09:33:12] VERBOSE[19158] pbx.c: -- Executing [01@from_nvst_cliok:1] Dial("SIP/trunk_to_nvst-00000000", "SIP/dw01") in new stack [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Asked to create a SIP channel with formats: 0x80004 (ulaw|h263) [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Allocating new SIP dialog for 10fe8dac0cee2db76d1ec1e63169a6c4@83.136.32.138:4343 - INVITE (No RTP) [Feb 22 09:33:12] DEBUG[19158] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc970e78' [Feb 22 09:33:12] DEBUG[19158] res_rtp_asterisk.c: Allocated port 16754 for RTP instance '0xc970e78' [Feb 22 09:33:12] DEBUG[19158] rtp_engine.c: RTP instance '0xc970e78' is setup and ready to go [Feb 22 09:33:12] DEBUG[19158] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc970e78' [Feb 22 09:33:12] VERBOSE[19158] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Setting NAT on RTP to On [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Feb 22 09:33:12] DEBUG[19158] acl.c: For destination '83.136.33.3', our source address is '83.136.32.138'. [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 83.136.32.138:4343 [Feb 22 09:33:12] DEBUG[19158] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: *** Our native formats are 0x80004 (ulaw|h263) [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: *** Joint capabilities are 0x80004 (ulaw|h263) [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Feb 22 09:33:12] DEBUG[19158] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: *** Our preferred formats from the incoming channel are 0x80004 (ulaw|h263) [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: This channel will not be able to handle video. [Feb 22 09:33:12] DEBUG[19158] rtp_engine.c: Seeded SDP of 'SIP/dw01-00000001' with that of 'SIP/trunk_to_nvst-00000000' [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable DIALEDTIME. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable ANSWEREDTIME. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable DIALEDPEERNAME. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable DIALEDPEERNUMBER. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable DIALSTATUS. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable SIPCALLID. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable SIPDOMAIN. [Feb 22 09:33:12] DEBUG[19158] channel.c: Not copying variable SIPURI. [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Outgoing Call for dw01 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Updating call counter for outgoing call [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: This call needs video offers, but there's no video support enabled! [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: ** Our capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: ** Our prefcodec: 0x80004 (ulaw|h263) [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Audio is at 4343 [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: -- Done with adding codecs to SDP [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Done building SDP. Settling with this capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Initializing initreq for method INVITE - callid 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 0 [ 65]: INVITE sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 3 [ 73]: From: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 4 [ 56]: To: [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 5 [ 46]: Contact: [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 6 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.3 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 9 [ 35]: Date: Tue, 22 Feb 2011 08:33:12 GMT [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: Reliably Transmitting (NAT) to 83.136.33.3:5060: INVITE sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport Max-Forwards: 70 From: "069911160036" ;tag=as4ed96e0b To: Contact: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 22 Feb 2011 08:33:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 1255718788 1255718788 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 16754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39 [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:12] VERBOSE[19158] app_dial.c: -- Called dw01 [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport From: "069911160036" ;tag=as4ed96e0b To: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 Date: Tue, 22 Feb 2011 08:33:12 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 <-------------> [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 3 [ 56]: To: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 5 [ 35]: Date: Tue, 22 Feb 2011 08:33:12 GMT [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 7 [ 25]: Server: Cisco-CP7960G/8.0 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 8 [ 61]: Contact: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 9 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: --- (11 headers 0 lines) --- [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: = Looking for Call ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 (Checking To) --From tag as4ed96e0b --To-tag [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: *** SIP TIMER: Cancelling retransmission #39 - INVITE (got response) [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Request 102: Found [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: SIP response 100 to standard invite [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport From: "069911160036" ;tag=as4ed96e0b To: ;tag=000e381b9dbc03622601a219-18b3db81 Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 Date: Tue, 22 Feb 2011 08:33:12 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 3 [ 94]: To: ;tag=000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 5 [ 35]: Date: Tue, 22 Feb 2011 08:33:12 GMT [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 7 [ 25]: Server: Cisco-CP7960G/8.0 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 8 [ 61]: Contact: [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 9 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 10 [101]: Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Feb 22 09:33:12] VERBOSE[19144] chan_sip.c: --- (12 headers 0 lines) --- [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: = Looking for Call ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 (Checking To) --From tag as4ed96e0b --To-tag 000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Request 102: Found [Feb 22 09:33:12] DEBUG[19144] chan_sip.c: SIP response 180 to standard invite [Feb 22 09:33:12] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - dw01 [Feb 22 09:33:12] DEBUG[19122] chan_sip.c: Checking device state for peer dw01 [Feb 22 09:33:12] DEBUG[19122] devicestate.c: Changing state for SIP/dw01 - state 1 (Not in use) [Feb 22 09:33:12] DEBUG[19122] devicestate.c: device 'SIP/dw01' state '1' [Feb 22 09:33:12] DEBUG[19156] app_queue.c: Device 'SIP/dw01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:12] VERBOSE[19158] app_dial.c: -- SIP/dw01-00000001 is ringing [Feb 22 09:33:12] DEBUG[19158] rtp_engine.c: Setting early bridge SDP of 'SIP/trunk_to_nvst-00000000' with that of 'SIP/dw01-00000001' [Feb 22 09:33:12] VERBOSE[19158] chan_sip.c: <--- Transmitting (no NAT) to 83.136.32.165:4343 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK242f7b44;received=83.136.32.165;rport=4343 From: "069911160036" ;tag=as2e19f809 To: ;tag=as704c7829 Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:12] DEBUG[19158] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 83.136.32.165:4343 [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport From: "069911160036" ;tag=as4ed96e0b To: ;tag=000e381b9dbc03622601a219-18b3db81 Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 Date: Tue, 22 Feb 2011 08:33:16 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 203 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 18578 0 IN IP4 10.10.0.122 s=SIP Call t=0 0 m=audio 17990 RTP/AVP 8 101 c=IN IP4 10.10.0.122 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK548f279b;rport [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 3 [ 94]: To: ;tag=000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 5 [ 35]: Date: Tue, 22 Feb 2011 08:33:16 GMT [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 7 [ 25]: Server: Cisco-CP7960G/8.0 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 8 [ 61]: Contact: [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 9 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 10 [101]: Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 11 [ 35]: Supported: replaces,join,norefersub [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 12 [ 19]: Content-Length: 203 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 14 [ 46]: Content-Disposition: session;handling=optional [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 15 [ 0]: [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 1 [ 40]: o=Cisco-SIPUA 18578 0 IN IP4 10.10.0.122 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 3 [ 5]: t=0 0 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 4 [ 27]: m=audio 17990 RTP/AVP 8 101 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 5 [ 20]: c=IN IP4 10.10.0.122 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Body 9 [ 10]: a=sendrecv [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: --- (15 headers 10 lines) --- [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: = Looking for Call ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 (Checking To) --From tag as4ed96e0b --To-tag 000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Acked pending invite 102 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Stopping retransmission on '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' of Request 102: Match Found [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: SIP response 200 to standard invite [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing session-level SDP o=Cisco-SIPUA 18578 0 IN IP4 10.10.0.122... UNSUPPORTED. [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Found RTP audio format 8 [Feb 22 09:33:16] DEBUG[19144] rtp_engine.c: Setting payload 8 based on m type on 0xb3d5253c [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Found RTP audio format 101 [Feb 22 09:33:16] DEBUG[19144] rtp_engine.c: Setting payload 101 based on m type on 0xb3d5253c [Feb 22 09:33:16] DEBUG[19144] netsock2.c: Splitting '10.10.0.122' gives... [Feb 22 09:33:16] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '(null)'. [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.10.0.122... OK. [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Feb 22 09:33:16] DEBUG[19144] rtp_engine.c: Incorporating payload 8 on 0xb3d5253c [Feb 22 09:33:16] DEBUG[19144] rtp_engine.c: Incorporating payload 101 on 0xb3d5253c [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 22 09:33:16] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc970e78' [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Peer audio RTP is at port 10.10.0.122:17990 [Feb 22 09:33:16] DEBUG[19144] rtp_engine.c: Copying payload 8 from 0xb3d5253c to 0xc971024 [Feb 22 09:33:16] DEBUG[19144] rtp_engine.c: Copying payload 101 from 0xb3d5253c to 0xc971024 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: We have an owner, now see if we need to change this call [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Oooh, we need to change our audio formats since our peer supports only 0x8 (alaw) and not 0x80004 (ulaw|h263) [Feb 22 09:33:16] DEBUG[19144] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:16] DEBUG[19144] channel.c: Set channel SIP/dw01-00000001 to read format ulaw [Feb 22 09:33:16] DEBUG[19144] channel.c: Set channel SIP/dw01-00000001 to write format ulaw [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Updating call counter for outgoing call [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: build_route: Contact hop: [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: list_route: hop: [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Strict routing enforced for session 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: set_destination: Parsing for address/port to send to [Feb 22 09:33:16] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:16] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: set_destination: set destination to 10.10.0.122:5060 [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: Transmitting (NAT) to 83.136.33.3:5060: ACK sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK22b13b49;rport Max-Forwards: 70 From: "069911160036" ;tag=as4ed96e0b To: ;tag=000e381b9dbc03622601a219-18b3db81 Contact: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.3 Content-Length: 0 --- [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Trying to put 'ACK sip:dw0' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:16] VERBOSE[19158] app_dial.c: -- SIP/dw01-00000001 answered SIP/trunk_to_nvst-00000000 [Feb 22 09:33:16] DEBUG[19158] rtp_engine.c: Setting early bridge SDP of 'SIP/trunk_to_nvst-00000000' with that of 'SIP/dw01-00000001' [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: SIP answering channel: SIP/trunk_to_nvst-00000000 [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: Setting framing from config on incoming call [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: This call needs video offers, but there's no video support enabled! [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: True [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Feb 22 09:33:16] VERBOSE[19158] chan_sip.c: Audio is at 4343 [Feb 22 09:33:16] VERBOSE[19158] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Feb 22 09:33:16] VERBOSE[19158] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 22 09:33:16] VERBOSE[19158] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 22 09:33:16] VERBOSE[19158] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: -- Done with adding codecs to SDP [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Feb 22 09:33:16] VERBOSE[19158] chan_sip.c: <--- Reliably Transmitting (no NAT) to 83.136.32.165:4343 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK242f7b44;received=83.136.32.165;rport=4343 From: "069911160036" ;tag=as2e19f809 To: ;tag=as704c7829 Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 337 v=0 o=root 1665521051 1665521051 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 10034 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 0 RTP/AVP 34 98 <------------> [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #42 [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.136.32.165:4343 [Feb 22 09:33:16] DEBUG[19158] features.c: bridge answer set, chan answer set [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:16] VERBOSE[19158] rtp_engine.c: -- Locally bridging SIP/trunk_to_nvst-00000000 and SIP/dw01-00000001 [Feb 22 09:33:16] DEBUG[19158] rtp_engine.c: rtp-engine-local-bridge: Oooh, formats changed, backing out [Feb 22 09:33:16] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - dw01 [Feb 22 09:33:16] DEBUG[19122] chan_sip.c: Checking device state for peer dw01 [Feb 22 09:33:16] DEBUG[19122] devicestate.c: Changing state for SIP/dw01 - state 1 (Not in use) [Feb 22 09:33:16] DEBUG[19122] devicestate.c: device 'SIP/dw01' state '1' [Feb 22 09:33:16] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - trunk_to_nvst [Feb 22 09:33:16] DEBUG[19122] chan_sip.c: Checking device state for peer trunk_to_nvst [Feb 22 09:33:16] DEBUG[19122] devicestate.c: Changing state for SIP/trunk_to_nvst - state 1 (Not in use) [Feb 22 09:33:16] DEBUG[19122] devicestate.c: device 'SIP/trunk_to_nvst' state '1' [Feb 22 09:33:16] DEBUG[19156] app_queue.c: Device 'SIP/dw01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:16] DEBUG[19156] app_queue.c: Device 'SIP/trunk_to_nvst' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.32.165:4343 ---> ACK sip:01@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK7dc0b9e7;rport Max-Forwards: 70 From: "069911160036" ;tag=as2e19f809 To: ;tag=as704c7829 Contact: Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.3 Content-Length: 0 <-------------> [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 0 [ 37]: ACK sip:01@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.165:4343;branch=z9hG4bK7dc0b9e7;rport [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 3 [ 73]: From: "069911160036" ;tag=as2e19f809 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 4 [ 46]: To: ;tag=as704c7829 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 5 [ 46]: Contact: [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 6 [ 60]: Call-ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.3 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Feb 22 09:33:16] VERBOSE[19144] chan_sip.c: --- (10 headers 0 lines) --- [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: = Looking for Call ID: 55c826da25e893170126ffd0621e1699@83.136.32.165:4343 (Checking From) --From tag as2e19f809 --To-tag as704c7829 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 [Feb 22 09:33:16] DEBUG[19144] chan_sip.c: Stopping retransmission on '55c826da25e893170126ffd0621e1699@83.136.32.165:4343' of Response 102: Match Found [Feb 22 09:33:16] DEBUG[19158] chan_sip.c: Oooh, format changed to alaw [Feb 22 09:33:16] DEBUG[19158] channel.c: Set channel SIP/trunk_to_nvst-00000000 to read format ulaw [Feb 22 09:33:16] DEBUG[19158] channel.c: Set channel SIP/trunk_to_nvst-00000000 to write format ulaw [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xc970e78' [Feb 22 09:33:16] VERBOSE[19158] rtp_engine.c: -- Locally bridging SIP/trunk_to_nvst-00000000 and SIP/dw01-00000001 [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc970e78' [Feb 22 09:33:16] DEBUG[19158] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 83.136.33.3:17990 [Feb 22 09:33:17] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:17] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '55c826da25e893170126ffd0621e1699@83.136.32.165:4343' Method: ACK [Feb 22 09:33:18] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:18] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '55c826da25e893170126ffd0621e1699@83.136.32.165:4343' Method: ACK [Feb 22 09:33:19] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:19] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '55c826da25e893170126ffd0621e1699@83.136.32.165:4343' Method: ACK [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '55c826da25e893170126ffd0621e1699@83.136.32.165:4343' Method: ACK [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> INVITE sip:069911160036@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK27cd9704 From: ;tag=000e381b9dbc03622601a219-18b3db81 To: "069911160036" ;tag=as4ed96e0b Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:20 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 18578 1 IN IP4 10.10.0.122 s=SIP Call t=0 0 m=audio 17990 RTP/AVP 8 0 18 101 c=IN IP4 10.10.0.122 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 0 [ 50]: INVITE sip:069911160036@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK27cd9704 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 2 [ 96]: From: ;tag=000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 3 [ 71]: To: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:20 GMT [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 7 [ 16]: CSeq: 101 INVITE [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 9 [ 61]: Contact: [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 11 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 12 [101]: Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 13 [ 35]: Supported: replaces,join,norefersub [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 14 [ 19]: Content-Length: 274 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 16 [ 46]: Content-Disposition: session;handling=optional [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 17 [ 0]: [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 1 [ 40]: o=Cisco-SIPUA 18578 1 IN IP4 10.10.0.122 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 3 [ 5]: t=0 0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 4 [ 32]: m=audio 17990 RTP/AVP 8 0 18 101 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 5 [ 20]: c=IN IP4 10.10.0.122 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-15 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Body 12 [ 10]: a=sendonly [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: --- (17 headers 13 lines) --- [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: = Looking for Call ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 (Checking From) --From tag 000e381b9dbc03622601a219-18b3db81 --To-tag as4ed96e0b [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces,join,norefersub" [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -replaces- [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: replaces [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -join- [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: join [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -norefersub- [Feb 22 09:33:20] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: norefersub [Feb 22 09:33:20] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:20] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Sending to 83.136.33.3:5060 (NAT) [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Initializing initreq for method INVITE - callid 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing session-level SDP o=Cisco-SIPUA 18578 1 IN IP4 10.10.0.122... UNSUPPORTED. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found RTP audio format 8 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Setting payload 8 based on m type on 0xb3d523cc [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found RTP audio format 0 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Setting payload 0 based on m type on 0xb3d523cc [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found RTP audio format 18 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Setting payload 18 based on m type on 0xb3d523cc [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found RTP audio format 101 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Setting payload 101 based on m type on 0xb3d523cc [Feb 22 09:33:20] DEBUG[19144] netsock2.c: Splitting '10.10.0.122' gives... [Feb 22 09:33:20] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '(null)'. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.10.0.122... OK. [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found audio description format PCMU for ID 0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found audio description format G729 for ID 18 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Incorporating payload 0 on 0xb3d523cc [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Incorporating payload 8 on 0xb3d523cc [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Incorporating payload 18 on 0xb3d523cc [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Incorporating payload 101 on 0xb3d523cc [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 22 09:33:20] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc970e78' [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Peer audio RTP is at port 10.10.0.122:17990 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Copying payload 0 from 0xb3d523cc to 0xc971024 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Copying payload 8 from 0xb3d523cc to 0xc971024 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Copying payload 18 from 0xb3d523cc to 0xc971024 [Feb 22 09:33:20] DEBUG[19144] rtp_engine.c: Copying payload 101 from 0xb3d523cc to 0xc971024 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: We have an owner, now see if we need to change this call [Feb 22 09:33:20] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc970e78' [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Got a SIP re-invite for call 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: SIP/dw01-00000001: This call is UP.... [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: <--- Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK27cd9704;received=83.136.33.3;rport=5060 From: ;tag=000e381b9dbc03622601a219-18b3db81 To: "069911160036" ;tag=as4ed96e0b Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 CSeq: 101 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Setting framing from config on incoming call [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: ** Our prefcodec: 0x80004 (ulaw|h263) [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Audio is at 4343 [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: -- Done with adding codecs to SDP [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: <--- Reliably Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK27cd9704;received=83.136.33.3;rport=5060 From: ;tag=000e381b9dbc03622601a219-18b3db81 To: "069911160036" ;tag=as4ed96e0b Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 CSeq: 101 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 1255718788 1255718789 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 16754 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '55c826da25e893170126ffd0621e1699@83.136.32.165:4343' Method: ACK [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:20] VERBOSE[19158] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/trunk_to_nvst-00000000 [Feb 22 09:33:20] DEBUG[19158] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:20] DEBUG[19158] channel.c: Generator got voice, switching to phase locked mode [Feb 22 09:33:20] DEBUG[19158] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 22 09:33:20] DEBUG[19158] channel.c: Set channel SIP/trunk_to_nvst-00000000 to write format slin [Feb 22 09:33:20] DEBUG[19158] res_musiconhold.c: SIP/trunk_to_nvst-00000000 Opened file 0 '/var/lib/asterisk/moh/macroform-cold_day' [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> ACK sip:069911160036@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK383a904d From: ;tag=000e381b9dbc03622601a219-18b3db81 To: "069911160036" ;tag=as4ed96e0b Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:21 GMT CSeq: 101 ACK User-Agent: Cisco-CP7960G/8.0 Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 0 [ 47]: ACK sip:069911160036@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK383a904d [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 2 [ 96]: From: ;tag=000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 3 [ 71]: To: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:21 GMT [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 7 [ 13]: CSeq: 101 ACK [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 9 [101]: Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Feb 22 09:33:20] VERBOSE[19144] chan_sip.c: --- (11 headers 0 lines) --- [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: = Looking for Call ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 (Checking From) --From tag 000e381b9dbc03622601a219-18b3db81 --To-tag as4ed96e0b [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 [Feb 22 09:33:20] DEBUG[19144] chan_sip.c: Stopping retransmission on '7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343' of Response 101: Match Found [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:20] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:21] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:22] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:23] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> INVITE sip:36@83.136.32.138 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK034545f5 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:24 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 11602 0 IN IP4 10.10.0.122 s=SIP Call t=0 0 m=audio 19036 RTP/AVP 8 0 18 101 c=IN IP4 10.10.0.122 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 0 [ 35]: INVITE sip:36@83.136.32.138 SIP/2.0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK034545f5 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 3 [ 26]: To: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 4 [ 56]: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:24 GMT [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 7 [ 16]: CSeq: 101 INVITE [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 9 [ 61]: Contact: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 10 [ 12]: Expires: 180 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 12 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 13 [102]: Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 14 [ 35]: Supported: replaces,join,norefersub [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 15 [ 19]: Content-Length: 274 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 17 [ 46]: Content-Disposition: session;handling=optional [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 18 [ 0]: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 1 [ 40]: o=Cisco-SIPUA 11602 0 IN IP4 10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 3 [ 5]: t=0 0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 4 [ 32]: m=audio 19036 RTP/AVP 8 0 18 101 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 5 [ 20]: c=IN IP4 10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-15 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 12 [ 10]: a=sendrecv [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: --- (18 headers 13 lines) --- [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: = Looking for Call ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (Checking From) --From tag 000e381b9dbc0363690d0ff3-3b445336 --To-tag [Feb 22 09:33:24] DEBUG[19144] acl.c: For destination '83.136.33.3', our source address is '83.136.32.138'. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 83.136.32.138:4343 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Allocating new SIP dialog for 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 - INVITE (No RTP) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces,join,norefersub" [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -replaces- [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: replaces [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -join- [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: join [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Found SIP option: -norefersub- [Feb 22 09:33:24] DEBUG[19144] sip/reqresp_parser.c: Matched SIP option: norefersub [Feb 22 09:33:24] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:24] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Sending to 83.136.33.3:5060 (no NAT) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Initializing initreq for method INVITE - callid 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Using INVITE request as basis request - 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found peer 'dw01' for 'dw01' from 83.136.33.3:5060 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: <--- Reliably Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK034545f5;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as38cfa202 Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 101 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d78701a" Content-Length: 0 <------------> [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #46 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Scheduling destruction of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' in 11200 ms (Method: INVITE) [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> ACK sip:36@83.136.32.138 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK034545f5 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as38cfa202 Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 Date: Tue, 22 Feb 2011 08:33:24 GMT CSeq: 101 ACK Content-Length: 0 <-------------> [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 0 [ 32]: ACK sip:36@83.136.32.138 SIP/2.0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK034545f5 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 3 [ 41]: To: ;tag=as38cfa202 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 4 [ 56]: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 5 [ 35]: Date: Tue, 22 Feb 2011 08:33:24 GMT [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 6 [ 13]: CSeq: 101 ACK [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: --- (8 headers 0 lines) --- [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: = Looking for Call ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (Checking From) --From tag 000e381b9dbc0363690d0ff3-3b445336 --To-tag as38cfa202 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #46 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Stopping retransmission on '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' of Response 101: Match Found [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> INVITE sip:36@83.136.32.138 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK5219d0d5 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:24 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 11602 0 IN IP4 10.10.0.122 s=SIP Call t=0 0 m=audio 19036 RTP/AVP 8 0 18 101 c=IN IP4 10.10.0.122 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 0 [ 35]: INVITE sip:36@83.136.32.138 SIP/2.0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK5219d0d5 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 3 [ 26]: To: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 4 [ 56]: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:24 GMT [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 9 [ 61]: Contact: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 11 [ 12]: Expires: 180 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 12 [ 23]: Accept: application/sdp [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 13 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 14 [102]: Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 15 [ 35]: Supported: replaces,join,norefersub [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 16 [ 19]: Content-Length: 274 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 17 [ 29]: Content-Type: application/sdp [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 18 [ 46]: Content-Disposition: session;handling=optional [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 19 [ 0]: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 1 [ 40]: o=Cisco-SIPUA 11602 0 IN IP4 10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 3 [ 5]: t=0 0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 4 [ 32]: m=audio 19036 RTP/AVP 8 0 18 101 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 5 [ 20]: c=IN IP4 10.10.0.122 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-15 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Body 12 [ 10]: a=sendrecv [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: --- (19 headers 13 lines) --- [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: = Looking for Call ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (Checking From) --From tag 000e381b9dbc0363690d0ff3-3b445336 --To-tag [Feb 22 09:33:24] DEBUG[19144] netsock2.c: Splitting '83.136.32.138' gives... [Feb 22 09:33:24] DEBUG[19144] netsock2.c: ...host '83.136.32.138' and port '(null)'. [Feb 22 09:33:24] DEBUG[19144] netsock2.c: Splitting '83.136.32.138' gives... [Feb 22 09:33:24] DEBUG[19144] netsock2.c: ...host '83.136.32.138' and port '(null)'. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 09:33:24] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:24] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Sending to 83.136.33.3:5060 (NAT) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Initializing initreq for method INVITE - callid 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Using INVITE request as basis request - 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found peer 'dw01' for 'dw01' from 83.136.33.3:5060 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc99c478' [Feb 22 09:33:24] DEBUG[19144] res_rtp_asterisk.c: Allocated port 17388 for RTP instance '0xc99c478' [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: RTP instance '0xc99c478' is setup and ready to go [Feb 22 09:33:24] DEBUG[19144] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc99c478' [Feb 22 09:33:24] VERBOSE[19144] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Setting NAT on RTP to On [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing session-level SDP o=Cisco-SIPUA 11602 0 IN IP4 10.10.0.122... UNSUPPORTED. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found RTP audio format 8 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Setting payload 8 based on m type on 0xb3d523cc [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found RTP audio format 0 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Setting payload 0 based on m type on 0xb3d523cc [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found RTP audio format 18 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Setting payload 18 based on m type on 0xb3d523cc [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found RTP audio format 101 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Setting payload 101 based on m type on 0xb3d523cc [Feb 22 09:33:24] DEBUG[19144] netsock2.c: Splitting '10.10.0.122' gives... [Feb 22 09:33:24] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '(null)'. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.10.0.122... OK. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found audio description format PCMU for ID 0 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found audio description format G729 for ID 18 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Incorporating payload 0 on 0xb3d523cc [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Incorporating payload 8 on 0xb3d523cc [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Incorporating payload 18 on 0xb3d523cc [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Incorporating payload 101 on 0xb3d523cc [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 22 09:33:24] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc99c478' [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Peer audio RTP is at port 10.10.0.122:19036 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Copying payload 0 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Copying payload 8 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Copying payload 18 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:24] DEBUG[19144] rtp_engine.c: Copying payload 101 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Checking SIP call limits for device dw01 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Updating call counter for incoming call [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: Looking for 36 in from_sip_phone (domain 83.136.32.138) [Feb 22 09:33:24] DEBUG[19144] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Feb 22 09:33:24] DEBUG[19144] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: This channel will not be able to handle video. [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: build_route: Contact hop: [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: list_route: hop: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: SIP/dw01-00000002: New call is still down.... Trying... [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: <--- Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK5219d0d5;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:24] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - dw01 [Feb 22 09:33:24] DEBUG[19122] chan_sip.c: Checking device state for peer dw01 [Feb 22 09:33:24] DEBUG[19122] devicestate.c: Changing state for SIP/dw01 - state 1 (Not in use) [Feb 22 09:33:24] DEBUG[19122] devicestate.c: device 'SIP/dw01' state '1' [Feb 22 09:33:24] DEBUG[19156] app_queue.c: Device 'SIP/dw01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:24] DEBUG[19159] pbx.c: Result of 'EXTEN' is '36' [Feb 22 09:33:24] DEBUG[19159] pbx.c: Launching 'Dial' [Feb 22 09:33:24] VERBOSE[19159] pbx.c: -- Executing [36@from_sip_phone:1] Dial("SIP/dw01-00000002", "SIP/trunk_to_nvst/+431505641636") in new stack [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Allocating new SIP dialog for 6dd62b0b55da51d776507b3a2456d12f@83.136.32.138:4343 - INVITE (No RTP) [Feb 22 09:33:24] DEBUG[19159] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc991a58' [Feb 22 09:33:24] DEBUG[19159] res_rtp_asterisk.c: Allocated port 14962 for RTP instance '0xc991a58' [Feb 22 09:33:24] DEBUG[19159] rtp_engine.c: RTP instance '0xc991a58' is setup and ready to go [Feb 22 09:33:24] DEBUG[19159] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc991a58' [Feb 22 09:33:24] VERBOSE[19159] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Setting NAT on RTP to Off [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Feb 22 09:33:24] DEBUG[19159] acl.c: For destination '83.136.32.165', our source address is '83.136.32.138'. [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 83.136.32.138:4343 [Feb 22 09:33:24] DEBUG[19159] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Feb 22 09:33:24] DEBUG[19159] frame.c: Could not find preferred codec - Going for the best codec [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: This channel will not be able to handle video. [Feb 22 09:33:24] DEBUG[19159] rtp_engine.c: Seeded SDP of 'SIP/trunk_to_nvst-00000003' with that of 'SIP/dw01-00000002' [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable DIALEDTIME. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable ANSWEREDTIME. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable DIALEDPEERNAME. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable DIALEDPEERNUMBER. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable DIALSTATUS. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable SIPCALLID. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable SIPDOMAIN. [Feb 22 09:33:24] DEBUG[19159] channel.c: Not copying variable SIPURI. [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Outgoing Call for +431505641636 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Updating call counter for outgoing call [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: This call needs video offers, but there's no video support enabled! [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: ** Our capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Audio is at 4343 [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: -- Done with adding codecs to SDP [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Done building SDP. Settling with this capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Initializing initreq for method INVITE - callid 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 0 [ 51]: INVITE sip:+431505641636@83.136.32.165:4343 SIP/2.0 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 3 [ 61]: From: "Cisco 7960" ;tag=as61c0c219 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 4 [ 42]: To: [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 5 [ 36]: Contact: [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 6 [ 60]: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.3 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 9 [ 35]: Date: Tue, 22 Feb 2011 08:33:24 GMT [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Feb 22 09:33:24] VERBOSE[19159] chan_sip.c: Reliably Transmitting (no NAT) to 83.136.32.165:4343: INVITE sip:+431505641636@83.136.32.165:4343 SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c Max-Forwards: 70 From: "Cisco 7960" ;tag=as61c0c219 To: Contact: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 22 Feb 2011 08:33:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 1517434290 1517434290 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 14962 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #49 [Feb 22 09:33:24] DEBUG[19159] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 83.136.32.165:4343 [Feb 22 09:33:24] VERBOSE[19159] app_dial.c: -- Called trunk_to_nvst/+431505641636 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.32.165:4343 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 From: "Cisco 7960" ;tag=as61c0c219 To: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 2 [ 61]: From: "Cisco 7960" ;tag=as61c0c219 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 3 [ 42]: To: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.2.3 [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 9 [ 47]: Contact: [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Feb 22 09:33:24] VERBOSE[19144] chan_sip.c: --- (11 headers 0 lines) --- [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: = Looking for Call ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 (Checking To) --From tag as61c0c219 --To-tag [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: *** SIP TIMER: Cancelling retransmission #49 - INVITE (got response) [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Request 102: Found [Feb 22 09:33:24] DEBUG[19144] chan_sip.c: SIP response 100 to standard invite [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:24] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.32.165:4343 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 From: "Cisco 7960" ;tag=as61c0c219 To: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 2 [ 61]: From: "Cisco 7960" ;tag=as61c0c219 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 3 [ 42]: To: [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.2.3 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 9 [ 47]: Contact: [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Feb 22 09:33:25] VERBOSE[19144] chan_sip.c: --- (11 headers 0 lines) --- [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: = Looking for Call ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 (Checking To) --From tag as61c0c219 --To-tag [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Request 102: Found [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: SIP response 100 to standard invite [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.32.165:4343 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 From: "Cisco 7960" ;tag=as61c0c219 To: ;tag=as3ae34377 Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 2 [ 61]: From: "Cisco 7960" ;tag=as61c0c219 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 3 [ 57]: To: ;tag=as3ae34377 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.2.3 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 9 [ 47]: Contact: [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Feb 22 09:33:25] VERBOSE[19144] chan_sip.c: --- (11 headers 0 lines) --- [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: = Looking for Call ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 (Checking To) --From tag as61c0c219 --To-tag as3ae34377 [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Request 102: Found [Feb 22 09:33:25] DEBUG[19144] chan_sip.c: SIP response 180 to standard invite [Feb 22 09:33:25] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - trunk_to_nvst [Feb 22 09:33:25] DEBUG[19122] chan_sip.c: Checking device state for peer trunk_to_nvst [Feb 22 09:33:25] DEBUG[19122] devicestate.c: Changing state for SIP/trunk_to_nvst - state 1 (Not in use) [Feb 22 09:33:25] DEBUG[19122] devicestate.c: device 'SIP/trunk_to_nvst' state '1' [Feb 22 09:33:25] DEBUG[19156] app_queue.c: Device 'SIP/trunk_to_nvst' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:25] VERBOSE[19159] app_dial.c: -- SIP/trunk_to_nvst-00000003 is ringing [Feb 22 09:33:25] DEBUG[19159] rtp_engine.c: Setting early bridge SDP of 'SIP/dw01-00000002' with that of 'SIP/trunk_to_nvst-00000003' [Feb 22 09:33:25] VERBOSE[19159] chan_sip.c: <--- Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK5219d0d5;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:25] DEBUG[19159] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:25] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:26] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.32.165:4343 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 From: "Cisco 7960" ;tag=as61c0c219 To: ;tag=as3ae34377 Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 867906520 867906520 IN IP4 83.136.32.165 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.165 t=0 0 m=audio 11822 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK3ae8d30c;received=83.136.32.138;rport=4343 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 2 [ 61]: From: "Cisco 7960" ;tag=as61c0c219 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 3 [ 57]: To: ;tag=as3ae34377 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.2.3 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 9 [ 47]: Contact: [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 11 [ 19]: Content-Length: 283 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 12 [ 0]: [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 1 [ 47]: o=root 867906520 867906520 IN IP4 83.136.32.165 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.2.3 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 3 [ 22]: c=IN IP4 83.136.32.165 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 4 [ 5]: t=0 0 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 5 [ 31]: m=audio 11822 RTP/AVP 3 0 8 101 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 6 [ 19]: a=rtpmap:3 GSM/8000 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Body 12 [ 10]: a=sendrecv [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: --- (12 headers 13 lines) --- [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: = Looking for Call ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 (Checking To) --From tag as61c0c219 --To-tag as3ae34377 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Acked pending invite 102 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Stopping retransmission on '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' of Request 102: Match Found [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: SIP response 200 to standard invite [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing session-level SDP o=root 867906520 867906520 IN IP4 83.136.32.165... UNSUPPORTED. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing session-level SDP s=Asterisk PBX 1.8.2.3... UNSUPPORTED. [Feb 22 09:33:27] DEBUG[19144] netsock2.c: Splitting '83.136.32.165' gives... [Feb 22 09:33:27] DEBUG[19144] netsock2.c: ...host '83.136.32.165' and port '(null)'. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing session-level SDP c=IN IP4 83.136.32.165... OK. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found RTP audio format 3 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Setting payload 3 based on m type on 0xb3d5253c [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found RTP audio format 0 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Setting payload 0 based on m type on 0xb3d5253c [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found RTP audio format 8 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Setting payload 8 based on m type on 0xb3d5253c [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found RTP audio format 101 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Setting payload 101 based on m type on 0xb3d5253c [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found audio description format GSM for ID 3 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found audio description format PCMU for ID 0 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Incorporating payload 0 on 0xb3d5253c [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Incorporating payload 3 on 0xb3d5253c [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Incorporating payload 8 on 0xb3d5253c [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Incorporating payload 101 on 0xb3d5253c [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 22 09:33:27] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc991a58' [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Peer audio RTP is at port 83.136.32.165:11822 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Copying payload 0 from 0xb3d5253c to 0xc991c04 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Copying payload 3 from 0xb3d5253c to 0xc991c04 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Copying payload 8 from 0xb3d5253c to 0xc991c04 [Feb 22 09:33:27] DEBUG[19144] rtp_engine.c: Copying payload 101 from 0xb3d5253c to 0xc991c04 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: We have an owner, now see if we need to change this call [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Updating call counter for outgoing call [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: build_route: Contact hop: [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: list_route: hop: [Feb 22 09:33:27] DEBUG[19144] netsock2.c: Splitting '83.136.32.165:4343' gives... [Feb 22 09:33:27] DEBUG[19144] netsock2.c: ...host '83.136.32.165' and port '4343'. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Strict routing enforced for session 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: set_destination: Parsing for address/port to send to [Feb 22 09:33:27] DEBUG[19144] netsock2.c: Splitting '83.136.32.165:4343' gives... [Feb 22 09:33:27] DEBUG[19144] netsock2.c: ...host '83.136.32.165' and port '4343'. [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: set_destination: set destination to 83.136.32.165:4343 [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Transmitting (no NAT) to 83.136.32.165:4343: ACK sip:+431505641636@83.136.32.165:4343 SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK18f9255b Max-Forwards: 70 From: "Cisco 7960" ;tag=as61c0c219 To: ;tag=as3ae34377 Contact: Call-ID: 1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.3 Content-Length: 0 --- [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Trying to put 'ACK sip:+43' onto UDP socket destined for 83.136.32.165:4343 [Feb 22 09:33:27] VERBOSE[19159] app_dial.c: -- SIP/trunk_to_nvst-00000003 answered SIP/dw01-00000002 [Feb 22 09:33:27] DEBUG[19159] rtp_engine.c: Setting early bridge SDP of 'SIP/dw01-00000002' with that of 'SIP/trunk_to_nvst-00000003' [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: SIP answering channel: SIP/dw01-00000002 [Feb 22 09:33:27] DEBUG[19159] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: Setting framing from config on incoming call [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Feb 22 09:33:27] VERBOSE[19159] chan_sip.c: Audio is at 4343 [Feb 22 09:33:27] VERBOSE[19159] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 22 09:33:27] VERBOSE[19159] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 22 09:33:27] VERBOSE[19159] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: -- Done with adding codecs to SDP [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 09:33:27] VERBOSE[19159] chan_sip.c: <--- Reliably Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK5219d0d5;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 518012286 518012286 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 17388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #52 [Feb 22 09:33:27] DEBUG[19159] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:27] DEBUG[19159] features.c: bridge answer set, chan answer set [Feb 22 09:33:27] DEBUG[19159] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:27] DEBUG[19159] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:27] VERBOSE[19159] rtp_engine.c: -- Locally bridging SIP/dw01-00000002 and SIP/trunk_to_nvst-00000003 [Feb 22 09:33:27] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - trunk_to_nvst [Feb 22 09:33:27] DEBUG[19122] chan_sip.c: Checking device state for peer trunk_to_nvst [Feb 22 09:33:27] DEBUG[19122] devicestate.c: Changing state for SIP/trunk_to_nvst - state 1 (Not in use) [Feb 22 09:33:27] DEBUG[19122] devicestate.c: device 'SIP/trunk_to_nvst' state '1' [Feb 22 09:33:27] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - dw01 [Feb 22 09:33:27] DEBUG[19122] chan_sip.c: Checking device state for peer dw01 [Feb 22 09:33:27] DEBUG[19122] devicestate.c: Changing state for SIP/dw01 - state 1 (Not in use) [Feb 22 09:33:27] DEBUG[19122] devicestate.c: device 'SIP/dw01' state '1' [Feb 22 09:33:27] DEBUG[19156] app_queue.c: Device 'SIP/trunk_to_nvst' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:27] DEBUG[19156] app_queue.c: Device 'SIP/dw01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: INVITE [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19159] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc99c478' [Feb 22 09:33:27] DEBUG[19159] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 83.136.33.3:19036 [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: SIP TIMER: Rescheduling retransmission #52 (1) SIP/2.0 - 1 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 350 ms (t1 175 ms (Retrans id #52)) [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: Retransmitting #1 (NAT) to 83.136.33.3:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK5219d0d5;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 518012286 518012286 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 17388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: INVITE [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> ACK sip:36@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK3060cbcd From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:27 GMT CSeq: 102 ACK User-Agent: Cisco-CP7960G/8.0 Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 0 [ 37]: ACK sip:36@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK3060cbcd [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 3 [ 41]: To: ;tag=as52d3fe7b [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 4 [ 56]: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:27 GMT [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 10 [102]: Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Feb 22 09:33:27] VERBOSE[19144] chan_sip.c: --- (12 headers 0 lines) --- [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: = Looking for Call ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (Checking From) --From tag 000e381b9dbc0363690d0ff3-3b445336 --To-tag as52d3fe7b [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #52 [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Stopping retransmission on '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' of Response 102: Match Found [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:27] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:27] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:28] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:28] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:29] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:29] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:30] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:30] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Auto destroying SIP dialog 'MjlkNWMyZmQ3M2NjYTBlYmU0OTY5NDZjYjE5NTcxMWY.' [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Destroying SIP dialog MjlkNWMyZmQ3M2NjYTBlYmU0OTY5NDZjYjE5NTcxMWY. [Feb 22 09:33:31] VERBOSE[19144] chan_sip.c: Really destroying SIP dialog 'MjlkNWMyZmQ3M2NjYTBlYmU0OTY5NDZjYjE5NTcxMWY.' Method: REGISTER [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Auto destroying SIP dialog 'YjdkNGRjZmIyNzFlM2E5NTE4OWY4YmNiYjFlNWEwMGM.' [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Destroying SIP dialog YjdkNGRjZmIyNzFlM2E5NTE4OWY4YmNiYjFlNWEwMGM. [Feb 22 09:33:31] VERBOSE[19144] chan_sip.c: Really destroying SIP dialog 'YjdkNGRjZmIyNzFlM2E5NTE4OWY4YmNiYjFlNWEwMGM.' Method: REGISTER [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:31] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:31] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19159] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:32] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:32] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:33] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:33] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:34] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:34] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:35] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:35] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Allocating new SIP dialog for 39e03571399771503e5fd3d53682346e@83.136.32.138:4343 - OPTIONS (No RTP) [Feb 22 09:33:36] DEBUG[19144] acl.c: For destination '83.136.33.3', our source address is '83.136.32.138'. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 83.136.32.138:4343 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Initializing initreq for method OPTIONS - callid 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 0 [ 66]: OPTIONS sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK13a4c10b;rport [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 3 [ 65]: From: "asterisk" ;tag=as5665df30 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 4 [ 56]: To: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 5 [ 42]: Contact: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 6 [ 60]: Call-ID: 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.3 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 9 [ 35]: Date: Tue, 22 Feb 2011 08:33:36 GMT [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Reliably Transmitting (NAT) to 83.136.33.3:5060: OPTIONS sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK13a4c10b;rport Max-Forwards: 70 From: "asterisk" ;tag=as5665df30 To: Contact: Call-ID: 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 22 Feb 2011 08:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #53 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK13a4c10b;rport From: "asterisk" ;tag=as5665df30 To: ;tag=000e381b9dbc03644827434c-1bc8228c Call-ID: 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 Date: Tue, 22 Feb 2011 08:33:36 GMT CSeq: 102 OPTIONS Server: Cisco-CP7960G/8.0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 236 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 16040 0 IN IP4 10.10.0.122 s=SIP Call t=0 0 m=audio 0 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK13a4c10b;rport [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 2 [ 65]: From: "asterisk" ;tag=as5665df30 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 3 [ 94]: To: ;tag=000e381b9dbc03644827434c-1bc8228c [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 5 [ 35]: Date: Tue, 22 Feb 2011 08:33:36 GMT [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 7 [ 25]: Server: Cisco-CP7960G/8.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 8 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 9 [ 61]: Accept: application/sdp,multipart/mixed,multipart/alternative [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 10 [ 25]: Accept-Encoding: identity [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 11 [ 19]: Accept-Language: en [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 12 [ 35]: Supported: replaces,join,norefersub [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 13 [ 19]: Content-Length: 236 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 15 [ 46]: Content-Disposition: session;handling=optional [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 16 [ 0]: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 1 [ 40]: o=Cisco-SIPUA 16040 0 IN IP4 10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 3 [ 5]: t=0 0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 4 [ 28]: m=audio 0 RTP/AVP 8 0 18 101 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 5 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 8 [ 19]: a=fmtp:18 annexb=no [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: --- (16 headers 11 lines) --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: = Looking for Call ID: 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 (Checking To) --From tag as5665df30 --To-tag 000e381b9dbc03644827434c-1bc8228c [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #53 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Stopping retransmission on '1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343' of Request 102: Match Found [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: ACK [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Destroying SIP dialog 1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Really destroying SIP dialog '1e7c00616be5d7da5f90a61456d42073@83.136.32.138:4343' Method: OPTIONS [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> INVITE sip:36@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK73f740ac From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:36 GMT CSeq: 103 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 11602 1 IN IP4 10.10.0.122 s=SIP Call t=0 0 m=audio 19036 RTP/AVP 8 0 18 101 c=IN IP4 10.10.0.122 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 0 [ 40]: INVITE sip:36@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK73f740ac [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 3 [ 41]: To: ;tag=as52d3fe7b [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 4 [ 56]: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:36 GMT [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 9 [ 61]: Contact: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 11 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 12 [102]: Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 13 [ 35]: Supported: replaces,join,norefersub [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 15 [ 19]: Content-Length: 274 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 17 [ 46]: Content-Disposition: session;handling=optional [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 18 [ 0]: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 0 [ 3]: v=0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 1 [ 40]: o=Cisco-SIPUA 11602 1 IN IP4 10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 2 [ 10]: s=SIP Call [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 3 [ 5]: t=0 0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 4 [ 32]: m=audio 19036 RTP/AVP 8 0 18 101 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 5 [ 20]: c=IN IP4 10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-15 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Body 12 [ 10]: a=sendonly [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: --- (18 headers 13 lines) --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: = Looking for Call ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (Checking From) --From tag 000e381b9dbc0363690d0ff3-3b445336 --To-tag as52d3fe7b [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Feb 22 09:33:36] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:36] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Sending to 83.136.33.3:5060 (NAT) [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Initializing initreq for method INVITE - callid 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing session-level SDP o=Cisco-SIPUA 11602 1 IN IP4 10.10.0.122... UNSUPPORTED. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found RTP audio format 8 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Setting payload 8 based on m type on 0xb3d523cc [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found RTP audio format 0 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Setting payload 0 based on m type on 0xb3d523cc [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found RTP audio format 18 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Setting payload 18 based on m type on 0xb3d523cc [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found RTP audio format 101 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Setting payload 101 based on m type on 0xb3d523cc [Feb 22 09:33:36] DEBUG[19144] netsock2.c: Splitting '10.10.0.122' gives... [Feb 22 09:33:36] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '(null)'. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.10.0.122... OK. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found audio description format PCMU for ID 0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found audio description format G729 for ID 18 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Incorporating payload 0 on 0xb3d523cc [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Incorporating payload 8 on 0xb3d523cc [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Incorporating payload 18 on 0xb3d523cc [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Incorporating payload 101 on 0xb3d523cc [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 22 09:33:36] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc99c478' [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Peer audio RTP is at port 10.10.0.122:19036 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Copying payload 0 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Copying payload 8 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Copying payload 18 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:36] DEBUG[19144] rtp_engine.c: Copying payload 101 from 0xb3d523cc to 0xc99c624 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: We have an owner, now see if we need to change this call [Feb 22 09:33:36] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc99c478' [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Got a SIP re-invite for call 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP/dw01-00000002: This call is UP.... [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK73f740ac;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 103 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Setting framing from config on incoming call [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Audio is at 4343 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: -- Done with adding codecs to SDP [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- Reliably Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK73f740ac;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 103 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 518012286 518012287 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 17388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #56 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1079cfb5375d081a1cf480464d5528a7@83.136.32.138:4343' Method: INVITE [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' Method: INVITE [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:36] VERBOSE[19159] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/trunk_to_nvst-00000003 [Feb 22 09:33:36] DEBUG[19159] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:36] DEBUG[19159] chan_sip.c: Oooh, format changed to alaw [Feb 22 09:33:36] DEBUG[19159] channel.c: Set channel SIP/trunk_to_nvst-00000003 to read format ulaw [Feb 22 09:33:36] DEBUG[19159] channel.c: Set channel SIP/trunk_to_nvst-00000003 to write format ulaw [Feb 22 09:33:36] DEBUG[19159] channel.c: Generator got voice, switching to phase locked mode [Feb 22 09:33:36] DEBUG[19159] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 22 09:33:36] DEBUG[19159] channel.c: Set channel SIP/trunk_to_nvst-00000003 to write format slin [Feb 22 09:33:36] DEBUG[19159] res_musiconhold.c: SIP/trunk_to_nvst-00000003 Opened file 0 '/var/lib/asterisk/moh/macroform-cold_day' [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19159] rtp_engine.c: rtp-engine-local-bridge: Oooh, formats changed, backing out [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP TIMER: Rescheduling retransmission #56 (1) SIP/2.0 - 1 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 350 ms (t1 175 ms (Retrans id #56)) [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Retransmitting #1 (NAT) to 83.136.33.3:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK73f740ac;received=83.136.33.3;rport=5060 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 103 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 518012286 518012287 IN IP4 83.136.32.138 s=Asterisk PBX 1.8.2.3 c=IN IP4 83.136.32.138 t=0 0 m=audio 17388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> ACK sip:36@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK3ad41a32 From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 To: ;tag=as52d3fe7b Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:36 GMT CSeq: 103 ACK User-Agent: Cisco-CP7960G/8.0 Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 0 [ 37]: ACK sip:36@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK3ad41a32 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 2 [ 73]: From: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 3 [ 41]: To: ;tag=as52d3fe7b [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 4 [ 56]: Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:36 GMT [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 10 [102]: Remote-Party-ID: "01" ;party=calling;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: --- (12 headers 0 lines) --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: = Looking for Call ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (Checking From) --From tag 000e381b9dbc0363690d0ff3-3b445336 --To-tag as52d3fe7b [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #56 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Stopping retransmission on '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' of Response 103: Match Found [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] DEBUG[19158] res_rtp_asterisk.c: No remote address on RTP instance '0xc970e78' so dropping frame [Feb 22 09:33:36] DEBUG[19159] res_rtp_asterisk.c: No remote address on RTP instance '0xc99c478' so dropping frame [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- SIP read from UDP:83.136.33.3:5060 ---> REFER sip:069911160036@83.136.32.138:4343 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK7daf3b4a From: ;tag=000e381b9dbc03622601a219-18b3db81 To: "069911160036" ;tag=as4ed96e0b Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 Max-Forwards: 70 Date: Tue, 22 Feb 2011 08:33:37 GMT CSeq: 102 REFER User-Agent: Cisco-CP7960G/8.0 Contact: Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes Refer-To: Referred-By: Content-Length: 0 <-------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 0 [ 49]: REFER sip:069911160036@83.136.32.138:4343 SIP/2.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 1 [ 56]: Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK7daf3b4a [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 2 [ 96]: From: ;tag=000e381b9dbc03622601a219-18b3db81 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 3 [ 71]: To: "069911160036" ;tag=as4ed96e0b [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 4 [ 60]: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 6 [ 35]: Date: Tue, 22 Feb 2011 08:33:37 GMT [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 7 [ 15]: CSeq: 102 REFER [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 8 [ 29]: User-Agent: Cisco-CP7960G/8.0 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 9 [ 61]: Contact: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 10 [101]: Remote-Party-ID: "01" ;party=called;id-type=subscriber;privacy=off;screen=yes [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 11 [160]: Refer-To: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 12 [ 37]: Referred-By: [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: --- (14 headers 0 lines) --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: = Looking for Call ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 (Checking From) --From tag 000e381b9dbc03622601a219-18b3db81 --To-tag as4ed96e0b [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Call 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 got a SIP call transfer from caller: (REFER)! [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 F-tag: 000e381b9dbc0363690d0ff3-3b445336 T-tag: as52d3fe7b [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: SIP transfer to extension 36@from_sip_phone by dw01@83.136.32.138 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP attended transfer: Transferer channel SIP/dw01-00000001, transferee channel SIP/trunk_to_nvst-00000000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/trunk_to_nvst-00000000' [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Looking for callid 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 (fromtag 000e381b9dbc0363690d0ff3-3b445336 totag as52d3fe7b) [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Matched INCOMING call - their tag is 000e381b9dbc0363690d0ff3-3b445336 Our tag is as52d3fe7b [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: <--- Transmitting (NAT) to 83.136.33.3:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.10.0.122:5060;branch=z9hG4bK7daf3b4a;received=83.136.33.3;rport=5060 From: ;tag=000e381b9dbc03622601a219-18b3db81 To: "069911160036" ;tag=as4ed96e0b Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 CSeq: 102 REFER Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP attended transfer: trying to bridge SIP/dw01-00000002 and SIP/trunk_to_nvst-00000000 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Sip transfer:-------------------- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: -- Transferer to PBX channel: SIP/dw01-00000001 State Up [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: -- Transferer to PBX second channel (target): SIP/dw01-00000002 State Up [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: -- Bridged call to transferee: SIP/trunk_to_nvst-00000000 State Up [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: -- Bridged call to transfer target: SIP/trunk_to_nvst-00000003 State Up [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: -- END Sip transfer:-------------------- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP transfer: Four channels to handle [Feb 22 09:33:36] VERBOSE[19144] res_musiconhold.c: -- Stopped music on hold on SIP/trunk_to_nvst-00000000 [Feb 22 09:33:36] DEBUG[19144] channel.c: Set channel SIP/trunk_to_nvst-00000000 to write format ulaw [Feb 22 09:33:36] DEBUG[19144] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 22 09:33:36] VERBOSE[19144] res_musiconhold.c: -- Stopped music on hold on SIP/trunk_to_nvst-00000003 [Feb 22 09:33:36] DEBUG[19144] channel.c: Set channel SIP/trunk_to_nvst-00000003 to write format ulaw [Feb 22 09:33:36] DEBUG[19144] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP transfer: trying to masquerade SIP/trunk_to_nvst-00000000 into SIP/dw01-00000002 [Feb 22 09:33:36] DEBUG[19144] channel.c: Planning to masquerade channel SIP/trunk_to_nvst-00000000 into the structure of SIP/dw01-00000002 [Feb 22 09:33:36] DEBUG[19144] channel.c: Done planning to masquerade channel SIP/trunk_to_nvst-00000000 into the structure of SIP/dw01-00000002 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Strict routing enforced for session 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: set_destination: Parsing for address/port to send to [Feb 22 09:33:36] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:36] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: set_destination: set destination to 10.10.0.122:5060 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Reliably Transmitting (NAT) to 83.136.33.3:5060: NOTIFY sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK5e802ec3;rport Max-Forwards: 70 From: "069911160036" ;tag=as4ed96e0b To: ;tag=000e381b9dbc03622601a219-18b3db81 Contact: Call-ID: 7adfd7a61a1558a82ba61cf14c7f4582@83.136.32.138:4343 CSeq: 103 NOTIFY User-Agent: Asterisk PBX 1.8.2.3 Event: refer;id=102 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19144] channel.c: Actually Masquerading SIP/trunk_to_nvst-00000000(6) into the structure of SIP/dw01-00000002(6) [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP Fixup: New owner for dialogue 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122: SIP/trunk_to_nvst-00000000 (Old parent: SIP/trunk_to_nvst-00000000) [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Hangup call SIP/trunk_to_nvst-00000000, SIP callid 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: update_call_counter(dw01) - decrement call limit counter on hangup [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Updating call counter for incoming call [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Call from peer 'dw01' removed from call limit 0 [Feb 22 09:33:36] DEBUG[19122] devicestate.c: No provider found, checking channel drivers for SIP - dw01 [Feb 22 09:33:36] DEBUG[19122] chan_sip.c: Checking device state for peer dw01 [Feb 22 09:33:36] DEBUG[19122] devicestate.c: Changing state for SIP/dw01 - state 1 (Not in use) [Feb 22 09:33:36] DEBUG[19122] devicestate.c: device 'SIP/dw01' state '1' [Feb 22 09:33:36] DEBUG[19156] app_queue.c: Device 'SIP/dw01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 22 09:33:36] DEBUG[19144] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc99c478' [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Scheduling destruction of SIP dialog '000e381b-9dbc0014-4662655e-70f00130@10.10.0.122' in 11200 ms (Method: ACK) [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Strict routing enforced for session 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: set_destination: Parsing for address/port to send to [Feb 22 09:33:36] DEBUG[19144] netsock2.c: Splitting '10.10.0.122:5060' gives... [Feb 22 09:33:36] DEBUG[19144] netsock2.c: ...host '10.10.0.122' and port '5060'. [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: set_destination: set destination to 10.10.0.122:5060 [Feb 22 09:33:36] VERBOSE[19144] chan_sip.c: Reliably Transmitting (NAT) to 83.136.33.3:5060: BYE sip:dw01@10.10.0.122:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 83.136.32.138:4343;branch=z9hG4bK0de2ec57;rport Max-Forwards: 70 From: ;tag=as52d3fe7b To: "01" ;tag=000e381b9dbc0363690d0ff3-3b445336 Call-ID: 000e381b-9dbc0014-4662655e-70f00130@10.10.0.122 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.2.3 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #60 [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: Trying to put 'BYE sip:dw0' onto UDP socket destined for 83.136.33.3:5060 [Feb 22 09:33:36] DEBUG[19144] channel.c: Set channel SIP/trunk_to_nvst-00000000 to write format ulaw [Feb 22 09:33:36] DEBUG[19144] channel.c: Set channel SIP/trunk_to_nvst-00000000 to read format ulaw [Feb 22 09:33:36] DEBUG[19144] channel.c: Putting channel SIP/trunk_to_nvst-00000000 in ulaw/ulaw formats [Feb 22 09:33:36] DEBUG[19144] chan_sip.c: SIP Fixup: New owner for dialogue 55c826da25e893170126ffd0621e1699@83.136.32.165:4343: SIP/trunk_to_nvst-00000000 (Old parent: SIP/dw01-00000002) [Feb 22 09:33:36] DEBUG[19144] channel.c: Released clone lock on 'SIP/dw01-00000002' [Feb 22 09:33:36] DEBUG[19144] channel.c: Done Masquerading SIP/trunk_to_nvst-00000000 (6) [Feb 22 09:33:36] DEBUG[19144] res_rtp_asterisk.c: Changing ssrc from 1837469404 to 1811483981 due to a source change [Feb 22 09:33:36] DEBUG[19144] res_rtp_asterisk.c: Changing ssrc from 807773904 to 580035119 due to a source change [Feb 22 09:33:36] DEBUG[19144] res_rtp_asterisk.c: Setting the marker bit due to a source update [Feb 22 09:33:36] VERBOSE[19159] rtp_engine.c: -- Locally bridging SIP/trunk_to_nvst-00000000 and SIP/trunk_to_nvst-00000003 [Feb 22 09:33:36] DEBUG[19158] rtp_engine.c: rtp-engine-local-bridge: Oooh, formats changed, backing out