[Nov 11 10:46:20] Asterisk 11.0.1 built by root @ localhost on a ppc64 running Linux on 2012-11-10 10:45:58 UTC [Nov 11 10:46:20] DEBUG[5657] config.c: Parsing /etc/asterisk/logger.conf [Nov 11 10:46:20] VERBOSE[5657] config.c: == Parsing '/etc/asterisk/logger.conf': Found [Nov 11 10:46:20] VERBOSE[5657] logger.c: Asterisk Queue Logger restarted [Nov 11 10:47:50] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> INVITE sip:950@sip.server.domain SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-aeb31d72 From: 808 ;tag=93197be22671e3f2o0 To: Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 101 INVITE Max-Forwards: 70 Contact: 808 Expires: 240 User-Agent: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62114797 62114797 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16412 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 0 [ 38]: INVITE sip:950@sip.server.domain SIP/2.0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-aeb31d72 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 2 [ 58]: From: 808 ;tag=93197be22671e3f2o0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 3 [ 29]: To: [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 4 [ 39]: Call-ID: 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 5 [ 16]: CSeq: 101 INVITE [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 7 [ 40]: Contact: 808 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 8 [ 12]: Expires: 240 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 10 [ 19]: Content-Length: 257 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 12 [ 29]: Supported: x-sipura, replaces [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 14 [ 0]: [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62114797 62114797 IN IP4 192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16412 RTP/AVP 8 100 101 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:47:50] VERBOSE[28487] chan_sip.c: --- (14 headers 13 lines) --- [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: = Looking for Call ID: 26f48ea2-7db3f032@192.168.0.11 (Checking From) --From tag 93197be22671e3f2o0 --To-tag [Nov 11 10:47:50] DEBUG[28487] logger.c: CALL_ID [C-0000003a] created by thread. [Nov 11 10:47:50] DEBUG[28487] acl.c: For destination '83.251.x.x', our source address is '192.168.24.17'. [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Target address 83.251.x.x:5060 is not local, substituting externaddr [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.224.x.x:5060 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Allocating new SIP dialog for 26f48ea2-7db3f032@192.168.0.11 - INVITE (No RTP) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 11 10:47:50] DEBUG[28487][C-0000003a] sip/reqresp_parser.c: Begin: parsing SIP "Supported: x-sipura, replaces" [Nov 11 10:47:50] DEBUG[28487][C-0000003a] sip/reqresp_parser.c: Found SIP option: -x-sipura- [Nov 11 10:47:50] DEBUG[28487][C-0000003a] sip/reqresp_parser.c: Found private SIP option, not supported: x-sipura [Nov 11 10:47:50] DEBUG[28487][C-0000003a] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 11 10:47:50] DEBUG[28487][C-0000003a] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: NAT detected for 192.168.0.11:5060 / 83.251.x.x:5060 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Sending to 83.251.x.x:5060 (NAT) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Initializing initreq for method INVITE - callid 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Using INVITE request as basis request - 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting 'sip.server.domain' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host 'sip.server.domain' and port ''. [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found peer '808' for '808' from 83.251.x.x:5060 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: <--- Reliably Transmitting (NAT) to 83.251.x.x:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-aeb31d72;received=83.251.x.x;rport=5060 From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as57a6b9c7 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 101 INVITE Server: SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1165f2e1" Content-Length: 0 <------------> [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39449 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Scheduling destruction of SIP dialog '26f48ea2-7db3f032@192.168.0.11' in 32000 ms (Method: INVITE) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:47:50] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> ACK sip:950@sip.server.domain SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-aeb31d72 From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as57a6b9c7 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 101 ACK Max-Forwards: 70 Contact: 808 User-Agent: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 0 [ 35]: ACK sip:950@sip.server.domain SIP/2.0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-aeb31d72 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 2 [ 58]: From: 808 ;tag=93197be22671e3f2o0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 3 [ 44]: To: ;tag=as57a6b9c7 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 4 [ 39]: Call-ID: 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 5 [ 13]: CSeq: 101 ACK [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 7 [ 40]: Contact: 808 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 8 [ 35]: User-Agent: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 11 10:47:50] VERBOSE[28487] chan_sip.c: --- (10 headers 0 lines) --- [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: = Looking for Call ID: 26f48ea2-7db3f032@192.168.0.11 (Checking From) --From tag 93197be22671e3f2o0 --To-tag as57a6b9c7 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39449 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Stopping retransmission on '26f48ea2-7db3f032@192.168.0.11' of Response 101: Match Found [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:47:50] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> INVITE sip:950@sip.server.domain SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-bce052ea From: 808 ;tag=93197be22671e3f2o0 To: Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="808",realm="asterisk",nonce="1165f2e1",uri="sip:950@sip.server.domain",algorithm=MD5,response="***removed***" Contact: 808 Expires: 240 User-Agent: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62114797 62114797 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16412 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 0 [ 38]: INVITE sip:950@sip.server.domain SIP/2.0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-bce052ea [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 2 [ 58]: From: 808 ;tag=93197be22671e3f2o0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 3 [ 29]: To: [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 4 [ 39]: Call-ID: 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 7 [158]: Authorization: Digest username="808",realm="asterisk",nonce="1165f2e1",uri="sip:950@sip.server.domain",algorithm=MD5,response="***removed***" [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 8 [ 40]: Contact: 808 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 9 [ 12]: Expires: 240 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 10 [ 35]: User-Agent: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 11 [ 19]: Content-Length: 257 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 12 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 13 [ 29]: Supported: x-sipura, replaces [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 15 [ 0]: [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62114797 62114797 IN IP4 192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16412 RTP/AVP 8 100 101 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:47:50] VERBOSE[28487] chan_sip.c: --- (15 headers 13 lines) --- [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: = Looking for Call ID: 26f48ea2-7db3f032@192.168.0.11 (Checking From) --From tag 93197be22671e3f2o0 --To-tag [Nov 11 10:47:50] DEBUG[28487] netsock2.c: Splitting 'sip.server.domain' into... [Nov 11 10:47:50] DEBUG[28487] netsock2.c: ...host 'sip.server.domain' and port ''. [Nov 11 10:47:50] DEBUG[28487] netsock2.c: Splitting 'sip.server.domain' into... [Nov 11 10:47:50] DEBUG[28487] netsock2.c: ...host 'sip.server.domain' and port ''. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: NAT detected for 192.168.0.11:5060 / 83.251.x.x:5060 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Sending to 83.251.x.x:5060 (NAT) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Initializing initreq for method INVITE - callid 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Using INVITE request as basis request - 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting 'sip.server.domain' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host 'sip.server.domain' and port ''. [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found peer '808' for '808' from 83.251.x.x:5060 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x109d9d2c' [Nov 11 10:47:50] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Allocated port 16572 for RTP instance '0x109d9d2c' [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: RTP instance '0x109d9d2c' is setup and ready to go [Nov 11 10:47:50] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x109d9d2c' [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Setting NAT on RTP to On [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP o=- 62114797 62114797 IN IP4 192.168.0.11... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.0.11' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.0.11' and port ''. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.11... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found RTP audio format 8 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Setting payload 8 based on m type on 0xf73c4128 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found RTP audio format 100 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Setting payload 100 based on m type on 0xf73c4128 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found RTP audio format 101 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Setting payload 101 based on m type on 0xf73c4128 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Unsetting payload 100 on 0xf73c4128 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found unknown media description format NSE for ID 100 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 NSE/8000... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 192-193... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x109d9d2c' [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Peer audio RTP is at port 192.168.0.11:16412 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Copying payload 8 from 0xf73c4128 to 0x109d9ed8 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Copying payload 101 from 0xf73c4128 to 0x109d9ed8 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x109d9d2c' [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: We're settling with these formats: (alaw) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Checking SIP call limits for device 808 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Updating call counter for incoming call [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting 'sip.server.domain' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host 'sip.server.domain' and port ''. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting 'sip.server.domain' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host 'sip.server.domain' and port ''. [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: Looking for 950 in localnumbers (domain sip.server.domain) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: *** Our native formats are (alaw) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: *** Joint capabilities are (alaw) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: This channel will not be able to handle video. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: build_route: Contact hop: 808 [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: list_route: hop: [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: SIP/808-0000004f: New call is still down.... Trying... [Nov 11 10:47:50] VERBOSE[28487][C-0000003a] chan_sip.c: <--- Transmitting (NAT) to 83.251.x.x:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-bce052ea;received=83.251.x.x;rport=5060 From: 808 ;tag=93197be22671e3f2o0 To: Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 INVITE Server: SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:47:50] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 808 [Nov 11 10:47:50] DEBUG[28478] chan_sip.c: Checking device state for peer 808 [Nov 11 10:47:50] DEBUG[28478] devicestate.c: Changing state for SIP/808 - state 1 (Not in use) [Nov 11 10:47:50] DEBUG[28478] devicestate.c: device 'SIP/808' state '1' [Nov 11 10:47:50] DEBUG[28514] app_queue.c: Device 'SIP/808' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Launching 'Macro' [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] pbx.c: -- Executing [950@localnumbers:1] Macro("SIP/808-0000004f", "growl-local") in new stack [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '808' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is 'Remote User' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Expression result is '0' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '808' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '808' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is 'Remote User' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Expression result is '0' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '808' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is 'Remote User' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '808' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '"Remote User <808>' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '"Remote User <808>' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Launching 'Set' [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] pbx.c: -- Executing [s@macro-growl-local:1] Set("SIP/808-0000004f", "CALLER="Remote User <808>"") in new stack [Nov 11 10:47:50] DEBUG[5673][C-0000003a] app_macro.c: Executed application: Set [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Result of 'EPOCH' is '1352627270' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is '10:47:50' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Result of 'CALLER' is '"Remote User <808>"' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=91, consumed=1, args.allowed=[0-9][A-Z][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=48, c2=57 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=93, consumed=1, args.allowed=][A-Z][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=91, consumed=1, args.allowed=[A-Z][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=65, c2=90 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=93, consumed=1, args.allowed=][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=91, consumed=1, args.allowed=[a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=97, c2=122 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=93, consumed=1, args.allowed=][åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=91, consumed=1, args.allowed=[åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=195, consumed=1, args.allowed=åäöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=165, consumed=1, args.allowed=¥Ã¤Ã¶Ã…ÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=195, consumed=1, args.allowed=äöÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=164, consumed=1, args.allowed=¤Ã¶Ã…ÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=195, consumed=1, args.allowed=öÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=182, consumed=1, args.allowed=¶Ã…ÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=195, consumed=1, args.allowed=ÅÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=133, consumed=1, args.allowed=…ÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=195, consumed=1, args.allowed=ÄÖ][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=132, consumed=1, args.allowed=„Ö][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=195, consumed=1, args.allowed=Ö][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=150, consumed=1, args.allowed=–][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=93, consumed=1, args.allowed=][+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=91, consumed=1, args.allowed=[+-><()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=43, c2=62 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=60, consumed=1, args.allowed=<()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=40, consumed=1, args.allowed=()][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=41, consumed=1, args.allowed=)][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=93, consumed=1, args.allowed=][ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=91, consumed=1, args.allowed=[ ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=32, consumed=1, args.allowed= ] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: c1=93, consumed=1, args.allowed=] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] func_strings.c: Allowed: ()+,-./0123456789:;<=>ABCDEFGHIJKLMNOPQRSTUVWXYZ[]abcdefghijklmnopqrstuvwxyz„…–¤¥¶Ã [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Function result is 'Remote User <808>' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Launching 'System' [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] pbx.c: -- Executing [s@macro-growl-local:2] System("SIP/808-0000004f", "growl 192.168.24.27 ******** "Inkommande samtal" "[10:47:50] Remote User <808>"") in new stack [Nov 11 10:47:50] DEBUG[5673][C-0000003a] app_macro.c: Executed application: System [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Result of 'EXTEN' is '950' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] pbx.c: Launching 'Dial' [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] pbx.c: -- Executing [950@localnumbers:2] Dial("SIP/808-0000004f", "SIP/950,240,gtwW") in new stack [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Allocating new SIP dialog for 6a6d199b283d44de0bad08bf5acba298@sip.server.domain - INVITE (No RTP) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x10b8eff4' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Allocated port 16186 for RTP instance '0x10b8eff4' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] rtp_engine.c: RTP instance '0x10b8eff4' is setup and ready to go [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x10b8eff4' [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Setting NAT on RTP to On [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 11 10:47:50] DEBUG[5673][C-0000003a] acl.c: For destination '192.168.24.3', our source address is '192.168.24.17'. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.24.17:5060 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: *** Our native formats are (alaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: *** Joint capabilities are (alaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: This channel will not be able to handle video. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel_internal_api.c: Channel Call ID changing from [C-0000003a] to [C-0000003a] [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable DIALEDTIME. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable ANSWEREDTIME. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable DIALEDPEERNAME. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable DIALSTATUS. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable MACRO_DEPTH. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable SYSTEMSTATUS. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable CALLER. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable SIPCALLID. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable SIPDOMAIN. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] channel.c: Not copying variable SIPURI. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Outgoing Call for 9950 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Updating call counter for outgoing call [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: This call needs video offers, but there's no video support enabled! [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: ** Our prefcodec: (alaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: -- Done with adding codecs to SDP [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Initializing initreq for method INVITE - callid 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 0 [ 36]: INVITE sip:9950@192.168.24.3 SIP/2.0 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.24.17:5060;branch=z9hG4bK624858b7;rport [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 3 [ 61]: From: "Remote User" ;tag=as28ae984b [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 4 [ 27]: To: [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 5 [ 37]: Contact: [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 6 [ 52]: Call-ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 8 [ 30]: User-Agent: SIP Server [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 9 [ 35]: Date: Sun, 11 Nov 2012 09:47:50 GMT [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39452 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.24.3:5060 [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] app_dial.c: -- Called SIP/950 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.24.17:5060;branch=z9hG4bK624858b7;rport [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 2 [ 61]: From: "Remote User" ;tag=as28ae984b [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 3 [ 45]: To: ;tag=31B0D756-11F6 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:47:50 GMT [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 5 [ 52]: Call-ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 8 [ 29]: Allow-Events: telephone-event [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain (Checking To) --From tag as28ae984b --To-tag 31B0D756-11F6 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: *** SIP TIMER: Cancelling retransmission #39452 - INVITE (got response) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0d260e6537ad6958395efdac5f73a2d9@sip.server.domain' Request 102: Found [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: SIP response 100 to standard invite [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 0 [ 28]: SIP/2.0 183 Session Progress [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.24.17:5060;branch=z9hG4bK624858b7;rport [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 2 [ 61]: From: "Remote User" ;tag=as28ae984b [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 3 [ 45]: To: ;tag=31B0D756-11F6 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:47:50 GMT [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 5 [ 52]: Call-ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 8 [ 29]: Allow-Events: telephone-event [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 9 [ 37]: Contact: [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 10 [ 46]: Content-Disposition: session;handling=required [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 12 [ 19]: Content-Length: 191 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Header 13 [ 0]: [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 9372 5082 IN IP4 192.168.24.3 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.24.3 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 5 [ 23]: m=audio 16958 RTP/AVP 8 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 6 [ 21]: c=IN IP4 192.168.24.3 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Nov 11 10:47:50] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain (Checking To) --From tag as28ae984b --To-tag 31B0D756-11F6 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0d260e6537ad6958395efdac5f73a2d9@sip.server.domain' Request 102: Found [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: SIP response 183 to standard invite [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: build_route: Contact hop: [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP o=CiscoSystemsSIP-GW-UserAgent 9372 5082 IN IP4 192.168.24.3... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.24.3' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.24.3' and port ''. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.24.3... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Setting payload 8 based on m type on 0xf73c3dc8 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.24.3' into... [Nov 11 10:47:50] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.24.3' and port ''. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.24.3... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 11 10:47:50] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10b8eff4' [Nov 11 10:47:50] DEBUG[28487][C-0000003a] rtp_engine.c: Copying payload 8 from 0xf73c3dc8 to 0x10b8f1a0 [Nov 11 10:47:50] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x10b8eff4' [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: We're settling with these formats: (alaw) [Nov 11 10:47:50] DEBUG[28487][C-0000003a] chan_sip.c: We have an owner, now see if we need to change this call [Nov 11 10:47:50] DEBUG[28487][C-0000003a] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:47:50] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] app_dial.c: -- SIP/950-00000050 is making progress passing it to SIP/808-0000004f [Nov 11 10:47:50] DEBUG[5673][C-0000003a] rtp_engine.c: Setting early bridge SDP of 'SIP/808-0000004f' with that of 'SIP/950-00000050' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Setting framing from config on incoming call [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: ** Our prefcodec: (nothing) [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] chan_sip.c: Audio is at 16572 [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] chan_sip.c: Adding codec 100004 (alaw) to SDP [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: -- Done with adding codecs to SDP [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Nov 11 10:47:50] VERBOSE[5673][C-0000003a] chan_sip.c: <--- Transmitting (NAT) to 83.251.x.x:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-bce052ea;received=83.251.x.x;rport=5060 From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as4a4be531 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 INVITE Server: SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1808833947 1808833947 IN IP4 85.224.x.x s=Asterisk PBX 11.0.1 c=IN IP4 85.224.x.x t=0 0 m=audio 16572 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Nov 11 10:47:50] DEBUG[5673][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 4, seq = 3506 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- start learning mode pass with addr = 192.168.24.3:16958 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- probation = 4, seq = 2770 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 3, seq = 3507 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- start learning mode pass with addr = 192.168.24.3:16958 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- probation = 3, seq = 2771 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 2, seq = 3508 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- start learning mode pass with addr = 192.168.24.3:16958 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- probation = 2, seq = 2772 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 1, seq = 3509 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x109d9d2c' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x10b8eff4' [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- start learning mode pass with addr = 192.168.24.3:16958 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- probation = 1, seq = 2773 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x10a6edb8 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.24.3:16958 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 4, seq = 3510 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 3, seq = 3511 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 2, seq = 3512 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- start learning mode pass with addr = 83.251.x.x:16412 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- probation = 1, seq = 3513 [Nov 11 10:47:50] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: 0x109d5390 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 83.251.x.x:16412 [Nov 11 10:47:55] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.24.17:5060;branch=z9hG4bK624858b7;rport [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 2 [ 61]: From: "Remote User" ;tag=as28ae984b [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 3 [ 45]: To: ;tag=31B0D756-11F6 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:47:50 GMT [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 5 [ 52]: Call-ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 6 [ 33]: Server: Cisco-SIPGateway/IOS-12.x [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 8 [ 86]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 9 [ 29]: Allow-Events: telephone-event [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 10 [ 37]: Contact: [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 12 [ 19]: Content-Length: 191 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 13 [ 0]: [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 1 [ 60]: o=CiscoSystemsSIP-GW-UserAgent 9372 5082 IN IP4 192.168.24.3 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.24.3 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 5 [ 23]: m=audio 16958 RTP/AVP 8 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 6 [ 21]: c=IN IP4 192.168.24.3 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Body 8 [ 10]: a=ptime:20 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain (Checking To) --From tag as28ae984b --To-tag 31B0D756-11F6 [Nov 11 10:47:56] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Acked pending invite 102 [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Stopping retransmission on '0d260e6537ad6958395efdac5f73a2d9@sip.server.domain' of Request 102: Match Found [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: SIP response 200 to standard invite [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Call 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain responded to our reinvite without changing SDP version; ignoring SDP. [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Updating call counter for outgoing call [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: build_route: Contact hop: [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Strict routing enforced for session 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:47:56] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.24.3:5060' into... [Nov 11 10:47:56] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.24.3' and port '5060'. [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Trying to put 'ACK sip:995' onto UDP socket destined for 192.168.24.3:5060 [Nov 11 10:47:56] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:47:56] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 950 [Nov 11 10:47:56] DEBUG[28478] chan_sip.c: Checking device state for peer 950 [Nov 11 10:47:56] DEBUG[28478] devicestate.c: Changing state for SIP/950 - state 1 (Not in use) [Nov 11 10:47:56] DEBUG[28478] devicestate.c: device 'SIP/950' state '1' [Nov 11 10:47:56] DEBUG[28514] app_queue.c: Device 'SIP/950' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:47:56] VERBOSE[5673][C-0000003a] app_dial.c: -- SIP/950-00000050 answered SIP/808-0000004f [Nov 11 10:47:56] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 808 [Nov 11 10:47:56] DEBUG[28478] chan_sip.c: Checking device state for peer 808 [Nov 11 10:47:56] DEBUG[28478] devicestate.c: Changing state for SIP/808 - state 1 (Not in use) [Nov 11 10:47:56] DEBUG[28478] devicestate.c: device 'SIP/808' state '1' [Nov 11 10:47:56] DEBUG[28514] app_queue.c: Device 'SIP/808' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: SIP answering channel: SIP/808-0000004f [Nov 11 10:47:56] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: Setting framing from config on incoming call [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: ** Our prefcodec: (nothing) [Nov 11 10:47:56] VERBOSE[5673][C-0000003a] chan_sip.c: Audio is at 16572 [Nov 11 10:47:56] VERBOSE[5673][C-0000003a] chan_sip.c: Adding codec 100004 (alaw) to SDP [Nov 11 10:47:56] VERBOSE[5673][C-0000003a] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: -- Done with adding codecs to SDP [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Nov 11 10:47:56] VERBOSE[5673][C-0000003a] chan_sip.c: <--- Reliably Transmitting (NAT) to 83.251.x.x:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-bce052ea;received=83.251.x.x;rport=5060 From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as4a4be531 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 INVITE Server: SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 1808833947 1808833948 IN IP4 85.224.x.x s=Asterisk PBX 11.0.1 c=IN IP4 85.224.x.x t=0 0 m=audio 16572 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39460 [Nov 11 10:47:56] DEBUG[5673][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:47:56] DEBUG[5673][C-0000003a] features.c: bridge answer set, chan answer set [Nov 11 10:47:56] DEBUG[5673][C-0000003a] features.c: Removing dialed interfaces datastore on SIP/950-00000050 since we're bridging [Nov 11 10:47:56] DEBUG[5673][C-0000003a] channel.c: setting peeraccount to 08 for SIP/950-00000050 from data on channel SIP/808-0000004f [Nov 11 10:47:56] DEBUG[5673][C-0000003a] channel.c: setting peeraccount to 08 for SIP/808-0000004f from data on channel SIP/950-00000050 [Nov 11 10:47:56] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:47:56] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:47:56] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> ACK sip:950@85.224.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-d15dbac5 From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as4a4be531 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="808",realm="asterisk",nonce="1165f2e1",uri="sip:950@sip.server.domain",algorithm=MD5,response="***removed***" Contact: 808 User-Agent: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 0 [ 38]: ACK sip:950@85.224.x.x:5060 SIP/2.0 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-d15dbac5 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 2 [ 58]: From: 808 ;tag=93197be22671e3f2o0 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 3 [ 44]: To: ;tag=as4a4be531 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 4 [ 39]: Call-ID: 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 5 [ 13]: CSeq: 102 ACK [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 7 [158]: Authorization: Digest username="808",realm="asterisk",nonce="1165f2e1",uri="sip:950@sip.server.domain",algorithm=MD5,response="***removed***" [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 8 [ 40]: Contact: 808 [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 11 10:47:56] VERBOSE[28487] chan_sip.c: --- (11 headers 0 lines) --- [Nov 11 10:47:56] DEBUG[28487] chan_sip.c: = Looking for Call ID: 26f48ea2-7db3f032@192.168.0.11 (Checking From) --From tag 93197be22671e3f2o0 --To-tag as4a4be531 [Nov 11 10:47:56] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39460 [Nov 11 10:47:56] DEBUG[28487][C-0000003a] chan_sip.c: Stopping retransmission on '26f48ea2-7db3f032@192.168.0.11' of Response 102: Match Found [Nov 11 10:47:56] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 0 [ 38]: BYE sip:808@192.168.24.17:5060 SIP/2.0 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 1 [ 34]: Via: SIP/2.0/UDP 192.168.24.3:5060 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 2 [ 47]: From: ;tag=31B0D756-11F6 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 3 [ 59]: To: "Remote User" ;tag=as28ae984b [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:47:56 GMT [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 5 [ 52]: Call-ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 6 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 7 [ 15]: Max-Forwards: 6 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 8 [ 21]: Timestamp: 1352627313 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 9 [ 13]: CSeq: 101 BYE [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain (Checking From) --From tag 31B0D756-11F6 --To-tag as28ae984b [Nov 11 10:48:33] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Initializing initreq for method BYE - callid 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:48:33] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.24.3:5060' into... [Nov 11 10:48:33] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.24.3' and port '5060'. [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: NAT detected for 192.168.24.3:5060 / 192.168.24.3:58663 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:48:33] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10b8eff4' [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Received bye, issuing owner hangup [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.24.3:58663 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:48:33] DEBUG[5673][C-0000003a] channel.c: Didn't get a frame from channel: SIP/950-00000050 [Nov 11 10:48:33] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:48:33] DEBUG[5673][C-0000003a] channel.c: Bridge stops bridging channels SIP/808-0000004f and SIP/950-00000050 [Nov 11 10:48:33] DEBUG[5673][C-0000003a] cdr_mysql.c: Inserting a CDR record. [Nov 11 10:48:33] DEBUG[5673][C-0000003a] cdr_mysql.c: SQL command as follows: INSERT INTO calls (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`accountcode`) VALUES ('2012-11-11 10:47:50','\"Remote User\" <808>','808','950','localnumbers','SIP/808-0000004f','SIP/950-00000050','Dial','SIP/950,240,gtwW','43','37','ANSWERED','3','08') [Nov 11 10:48:33] DEBUG[5673][C-0000003a] channel.c: Hanging up channel 'SIP/950-00000050' [Nov 11 10:48:33] DEBUG[5673][C-0000003a] chan_sip.c: Hangup call SIP/950-00000050, SIP callid 0d260e6537ad6958395efdac5f73a2d9@sip.server.domain [Nov 11 10:48:33] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10b8eff4' [Nov 11 10:48:33] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 950 [Nov 11 10:48:33] DEBUG[28478] chan_sip.c: Checking device state for peer 950 [Nov 11 10:48:33] DEBUG[28478] devicestate.c: Changing state for SIP/950 - state 1 (Not in use) [Nov 11 10:48:33] DEBUG[28478] devicestate.c: device 'SIP/950' state '1' [Nov 11 10:48:33] DEBUG[28514] app_queue.c: Device 'SIP/950' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:48:33] DEBUG[5673][C-0000003a] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 11 10:48:33] DEBUG[5673][C-0000003a] pbx.c: Launching 'Hangup' [Nov 11 10:48:33] VERBOSE[5673][C-0000003a] pbx.c: -- Executing [950@localnumbers:3] Hangup("SIP/808-0000004f", "") in new stack [Nov 11 10:48:33] DEBUG[5673][C-0000003a] channel.c: Soft-Hanging up channel 'SIP/808-0000004f' [Nov 11 10:48:33] DEBUG[5673][C-0000003a] pbx.c: Spawn extension (localnumbers,950,3) exited non-zero on 'SIP/808-0000004f' [Nov 11 10:48:33] VERBOSE[5673][C-0000003a] pbx.c: == Spawn extension (localnumbers, 950, 3) exited non-zero on 'SIP/808-0000004f' [Nov 11 10:48:33] DEBUG[5673][C-0000003a] channel.c: Soft-Hanging up channel 'SIP/808-0000004f' [Nov 11 10:48:33] DEBUG[5673][C-0000003a] channel.c: Hanging up channel 'SIP/808-0000004f' [Nov 11 10:48:33] DEBUG[5673][C-0000003a] chan_sip.c: Hangup call SIP/808-0000004f, SIP callid 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:48:33] DEBUG[5673][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x109d9d2c' [Nov 11 10:48:33] VERBOSE[5673][C-0000003a] chan_sip.c: Scheduling destruction of SIP dialog '26f48ea2-7db3f032@192.168.0.11' in 32000 ms (Method: ACK) [Nov 11 10:48:33] DEBUG[5673][C-0000003a] chan_sip.c: Strict routing enforced for session 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:48:33] VERBOSE[5673][C-0000003a] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 11 10:48:33] DEBUG[5673][C-0000003a] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:48:33] DEBUG[5673][C-0000003a] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:48:33] VERBOSE[5673][C-0000003a] chan_sip.c: set_destination: set destination to 192.168.0.11:5060 [Nov 11 10:48:33] VERBOSE[5673][C-0000003a] chan_sip.c: Reliably Transmitting (NAT) to 83.251.x.x:5060: BYE sip:808@192.168.0.11:5060 SIP/2.0 Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK4a3fcd0e;rport Max-Forwards: 70 From: ;tag=as4a4be531 To: 808 ;tag=93197be22671e3f2o0 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 BYE User-Agent: SIP Server Proxy-Authorization: Digest username="808", realm="asterisk", algorithm=MD5, uri="sip:sip.server.domain", nonce="1165f2e1", response="***removed***" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Nov 11 10:48:33] DEBUG[5673][C-0000003a] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39481 [Nov 11 10:48:33] DEBUG[5673][C-0000003a] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:48:33] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 808 [Nov 11 10:48:33] DEBUG[28478] chan_sip.c: Checking device state for peer 808 [Nov 11 10:48:33] DEBUG[28478] devicestate.c: Changing state for SIP/808 - state 1 (Not in use) [Nov 11 10:48:33] DEBUG[28478] devicestate.c: device 'SIP/808' state '1' [Nov 11 10:48:33] DEBUG[28514] app_queue.c: Device 'SIP/808' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:48:33] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> BYE sip:950@85.224.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-acf28b5b From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as4a4be531 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 103 BYE Max-Forwards: 70 Authorization: Digest username="808",realm="asterisk",nonce="1165f2e1",uri="sip:950@85.224.x.x:5060",algorithm=MD5,response="***removed***" User-Agent: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 0 [ 38]: BYE sip:950@85.224.x.x:5060 SIP/2.0 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-acf28b5b [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 2 [ 58]: From: 808 ;tag=93197be22671e3f2o0 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 3 [ 44]: To: ;tag=as4a4be531 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 4 [ 39]: Call-ID: 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 7 [161]: Authorization: Digest username="808",realm="asterisk",nonce="1165f2e1",uri="sip:950@85.224.x.x:5060",algorithm=MD5,response="***removed***" [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 8 [ 35]: User-Agent: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 11 10:48:33] VERBOSE[28487] chan_sip.c: --- (10 headers 0 lines) --- [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: = Looking for Call ID: 26f48ea2-7db3f032@192.168.0.11 (Checking From) --From tag 93197be22671e3f2o0 --To-tag as4a4be531 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39481 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Stopping retransmission on '26f48ea2-7db3f032@192.168.0.11' of Request 102: Match Found [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Initializing initreq for method BYE - callid 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:48:33] DEBUG[28487][C-0000003a] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: NAT detected for 192.168.0.11:5060 / 83.251.x.x:5060 [Nov 11 10:48:33] VERBOSE[28487][C-0000003a] chan_sip.c: Sending to 83.251.x.x:5060 (NAT) [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Setting SIP_ALREADYGONE on dialog 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x109d9d2c' [Nov 11 10:48:33] VERBOSE[28487][C-0000003a] chan_sip.c: Scheduling destruction of SIP dialog '26f48ea2-7db3f032@192.168.0.11' in 32000 ms (Method: BYE) [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Received bye, no owner, selfdestruct soon. [Nov 11 10:48:33] VERBOSE[28487][C-0000003a] chan_sip.c: <--- Transmitting (NAT) to 83.251.x.x:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-acf28b5b;received=83.251.x.x;rport=5060 From: 808 ;tag=93197be22671e3f2o0 To: ;tag=as4a4be531 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 103 BYE Server: SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:48:33] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: 808 ;tag=93197be22671e3f2o0 From: ;tag=as4a4be531 Call-ID: 26f48ea2-7db3f032@192.168.0.11 CSeq: 102 BYE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK4a3fcd0e Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 1 [ 56]: To: 808 ;tag=93197be22671e3f2o0 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 2 [ 46]: From: ;tag=as4a4be531 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 3 [ 39]: Call-ID: 26f48ea2-7db3f032@192.168.0.11 [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 4 [ 13]: CSeq: 102 BYE [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK4a3fcd0e [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:48:33] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:48:33] DEBUG[28487] chan_sip.c: = Looking for Call ID: 26f48ea2-7db3f032@192.168.0.11 (Checking To) --From tag as4a4be531 --To-tag 93197be22671e3f2o0 [Nov 11 10:48:33] DEBUG[28487][C-0000003a] logger.c: CALL_ID [C-0000003a] bound to thread. [Nov 11 10:48:33] DEBUG[28487][C-0000003a] chan_sip.c: Stopping retransmission on '26f48ea2-7db3f032@192.168.0.11' of Request 102: Match Not Found [Nov 11 10:48:33] DEBUG[28487][C-0000003a] logger.c: Call_ID [C-0000003a] being removed from thread. [Nov 11 10:48:36] VERBOSE[5660] asterisk.c: -- Remote UNIX connection disconnected [Nov 11 10:48:36] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> NOTIFY sip:sip.server.domain SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-f86f04ce From: 808 ;tag=9ee991e75ab11929o0 To: Call-ID: 8b8e8f5f-3f77a11@192.168.0.11 CSeq: 5030 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 0 [ 34]: NOTIFY sip:sip.server.domain SIP/2.0 [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-f86f04ce [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 2 [ 58]: From: 808 ;tag=9ee991e75ab11929o0 [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 3 [ 25]: To: [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 4 [ 38]: Call-ID: 8b8e8f5f-3f77a11@192.168.0.11 [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 5 [ 17]: CSeq: 5030 NOTIFY [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Event: keep-alive [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 8 [ 35]: User-Agent: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 11 10:48:36] VERBOSE[28487] chan_sip.c: --- (10 headers 0 lines) --- [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: = Looking for Call ID: 8b8e8f5f-3f77a11@192.168.0.11 (Checking From) --From tag 9ee991e75ab11929o0 --To-tag [Nov 11 10:48:36] DEBUG[28487] acl.c: For destination '83.251.x.x', our source address is '192.168.24.17'. [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Target address 83.251.x.x:5060 is not local, substituting externaddr [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.224.x.x:5060 [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Allocating new SIP dialog for 8b8e8f5f-3f77a11@192.168.0.11 - NOTIFY (No RTP) [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Got NOTIFY Event: keep-alive [Nov 11 10:48:36] VERBOSE[28487] chan_sip.c: <--- Transmitting (NAT) to 83.251.x.x:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-f86f04ce;received=83.251.x.x;rport=5060 From: 808 ;tag=9ee991e75ab11929o0 To: ;tag=as645ff7a8 Call-ID: 8b8e8f5f-3f77a11@192.168.0.11 CSeq: 5030 NOTIFY Server: SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 11 10:48:36] DEBUG[28487] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:48:36] VERBOSE[28487] chan_sip.c: Scheduling destruction of SIP dialog '8b8e8f5f-3f77a11@192.168.0.11' in 32000 ms (Method: NOTIFY) [Nov 11 10:49:16] VERBOSE[5685] asterisk.c: -- Remote UNIX connection disconnected